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Versions: (RFC 2326) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 34 35 36 37 38 39 40 RFC 7826

Internet Engineering Task Force                                MMUSIC WG
Internet Draft                                            H. Schulzrinne
draft-ietf-mmusic-rfc2326bis-14.txt                          Columbia U.
Dec 22, 2006                                                      A. Rao
Expires: June, 2007                                                Cisco
                                                             R. Lanphier
                                                            RealNetworks
                                                       Magnus Westerlund
                                                                Ericsson
                                                           A. Narasimhan
                                                                Overture


                Real Time Streaming Protocol 2.0 (RTSP)

STATUS OF THIS MEMO

   By submitting this Internet-Draft, each author represents that any
   applicable patent or other IPR claims of which he or she is aware
   have been or will be disclosed, and any of which he or she becomes
   aware will be disclosed, in accordance with Section 6 of BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
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   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
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   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   The list of current Internet-Drafts can be accessed at
   http://www.ietf.org/ietf/1id-abstracts.txt

   The list of Internet-Draft Shadow Directories can be accessed at
   http://www.ietf.org/shadow.html

Abstract

   This memorandum defines RTSP version 2.0 which is a revision of the
   Proposed Standard RTSP version 1.0 which is defined in RFC 2326.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over the delivery of data with real-time
   properties. RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video. Sources of data can include both live data feeds and stored
   clips. This protocol is intended to control multiple data delivery



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   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).
















































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                           Table of Contents



   1          Introduction ........................................    9
   1.1        RTSP Specification Update ...........................    9
   1.2        Purpose .............................................    9
   1.3        Notational Conventions ..............................   11
   1.3.1      RFC Editor Consideration ............................   11
   1.4        Terminology .........................................   11
   1.5        Protocol Properties .................................   15
   1.6        Extending RTSP ......................................   16
   1.7        Overall Operation ...................................   17
   1.8        RTSP States .........................................   18
   1.9        Relationship with Other Protocols ...................   19
   2          RTSP Use Cases ......................................   20
   2.1        On-demand Playback of Stored Content ................   20
   2.2        Unicast distribution of Live Content ................   21
   2.3        On-demand Playback using Multicast ..................   22
   2.4        Inviting an RTSP server into a conference ...........   22
   2.5        Live Content using Multicast ........................   23
   3          Protocol Parameters .................................   24
   3.1        RTSP Version ........................................   24
   3.2        RTSP IRI and URI ....................................   24
   3.3        Session Identifiers .................................   26
   3.4        SMPTE Relative Timestamps ...........................   26
   3.5        Normal Play Time ....................................   27
   3.6        Absolute Time .......................................   27
   3.7        Feature-tags ........................................   28
   3.8        Entity Tags .........................................   28
   4          RTSP Message ........................................   28
   4.1        Message Types .......................................   29
   4.2        Message Headers .....................................   29
   4.3        Message Body ........................................   29
   4.4        Message Length ......................................   29
   5          General Header Fields ...............................   30
   6          Request .............................................   30
   6.1        Request Line ........................................   31
   6.2        Request Header Fields ...............................   32
   7          Response ............................................   32
   7.1        Status-Line .........................................   33
   7.1.1      Status Code and Reason Phrase .......................   33
   7.2        Response Header Fields ..............................   34
   8          Entity ..............................................   35
   8.1        Entity Header Fields ................................   37



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   8.2        Entity Body .........................................   37
   9          Connections .........................................   37
   9.1        Reliability and Acknowledgements ....................   38
   9.2        Using Connections ...................................   39
   9.3        Closing Connections .................................   40
   9.4        Timing Out Connections and RTSP Messages ............   41
   9.5        Use of IPv6 .........................................   41
   10         Capability Handling .................................   41
   11         Method Definitions ..................................   43
   11.1       OPTIONS .............................................   44
   11.2       DESCRIBE ............................................   45
   11.3       SETUP ...............................................   47
   11.3.1     Changing Transport Parameters .......................   50
   11.4       PLAY ................................................   50
   11.5       PAUSE ...............................................   56
   11.6       TEARDOWN ............................................   58
   11.7       GET_PARAMETER .......................................   59
   11.8       SET_PARAMETER .......................................   59
   11.9       REDIRECT ............................................   61
   12         Embedded (Interleaved) Binary Data ..................   63
   13         Status Code Definitions .............................   65
   13.1       Success 1xx .........................................   65
   13.1.1     100 Continue ........................................   65
   13.2       Success 2xx .........................................   65
   13.3       Redirection 3xx .....................................   65
   13.3.1     300 Multiple Choices ................................   66
   13.3.2     301 Moved Permanently ...............................   66
   13.3.3     302 Found ...........................................   66
   13.3.4     303 See Other .......................................   67
   13.3.5     304 Not Modified ....................................   67
   13.3.6     305 Use Proxy .......................................   67
   13.4       Client Error 4xx ....................................   67
   13.4.1     400 Bad Request .....................................   68
   13.4.2     405 Method Not Allowed ..............................   68
   13.4.3     451 Parameter Not Understood ........................   68
   13.4.4     452 reserved ........................................   68
   13.4.5     453 Not Enough Bandwidth ............................   68
   13.4.6     454 Session Not Found ...............................   68
   13.4.7     455 Method Not Valid in This State ..................   68
   13.4.8     456 Header Field Not Valid for Resource .............   68
   13.4.9     457 Invalid Range ...................................   69
   13.4.10    458 Parameter Is Read-Only ..........................   69
   13.4.11    459 Aggregate Operation Not Allowed .................   69
   13.4.12    460 Only Aggregate Operation Allowed ................   69
   13.4.13    461 Unsupported Transport ...........................   69
   13.4.14    462 Destination Unreachable .........................   69
   13.4.15    463 Destination Prohibited ..........................   69
   13.4.16    464 Data Transport Not Ready Yet ....................   70



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   13.4.17    470 Connection Authorization Required ...............   70
   13.4.18    471 Connection Credentials not accepted .............   70
   13.5       Server Error 5xx ....................................   70
   13.5.1     551 Option not supported ............................   70
   14         Header Field Definitions ............................   70
   14.1       Accept ..............................................   74
   14.2       Accept-Credentials ..................................   75
   14.3       Accept-Encoding .....................................   75
   14.4       Accept-Language .....................................   77
   14.5       Accept-Ranges .......................................   77
   14.6       Allow ...............................................   78
   14.7       Authorization .......................................   78
   14.8       Bandwidth ...........................................   78
   14.9       Blocksize ...........................................   78
   14.10      Cache-Control .......................................   79
   14.11      Connection ..........................................   81
   14.12      Connection-Credentials ..............................   81
   14.13      Content-Base ........................................   82
   14.14      Content-Encoding ....................................   82
   14.15      Content-Language ....................................   82
   14.16      Content-Length ......................................   82
   14.17      Content-Location ....................................   82
   14.18      Content-Type ........................................   82
   14.19      CSeq ................................................   83
   14.20      Date ................................................   83
   14.21      ETag ................................................   83
   14.22      Expires .............................................   84
   14.23      From ................................................   85
   14.24      If-Match ............................................   85
   14.25      If-Modified-Since ...................................   85
   14.26      If-None-Match .......................................   86
   14.27      Last-Modified .......................................   86
   14.28      Location ............................................   86
   14.29      Proxy-Authenticate ..................................   86
   14.30      Proxy-Authorization .................................   86
   14.31      Proxy-Require .......................................   86
   14.32      Proxy-Supported .....................................   87
   14.33      Public ..............................................   87
   14.34      Range ...............................................   88
   14.35      Referer .............................................   90
   14.36      Retry-After .........................................   90
   14.37      Require .............................................   90
   14.38      RTP-Info ............................................   91
   14.39      Scale ...............................................   93
   14.40      Speed ...............................................   94
   14.41      Server ..............................................   95
   14.42      Session .............................................   95
   14.43      Supported ...........................................   96



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   14.44      Timestamp ...........................................   97
   14.45      Transport ...........................................   97
   14.46      Unsupported .........................................  103
   14.47      User-Agent ..........................................  104
   14.48      Vary ................................................  104
   14.49      Via .................................................  104
   14.50      WWW-Authenticate ....................................  104
   15         Proxies .............................................  104
   16         Caching .............................................  105
   17         Examples ............................................  106
   17.1       Media on Demand (Unicast) ...........................  106
   17.2       Media on Demand (Unicast) ...........................  109
   17.3       Single Stream Container Files .......................  112
   17.4       Live Media Presentation Using Multicast .............  114
   17.5       Capability Negotiation ..............................  115
   18         Security Framework ..................................  116
   18.1       RTSP and HTTP Authentication ........................  116
   18.2       RTSP over TLS .......................................  116
   18.3       Security and Proxies ................................  117
   18.3.1     Accept-Credentials ..................................  118
   18.3.2     User approved TLS procedure .........................  119
   19         Syntax ..............................................  121
   19.1       Base Syntax .........................................  121
   19.2       RTSP Protocol Definition ............................  123
   19.2.1     Generic Protocol elements ...........................  123
   19.2.2     Message Syntax ......................................  125
   19.2.3     Header Syntax .......................................  129
   19.3       SDP extension Syntax ................................  135
   20         Security Considerations .............................  135
   20.1       Remote denial of Service Attack .....................  137
   21         IANA Considerations .................................  138
   21.1       Feature-tags ........................................  139
   21.1.1     Description .........................................  139
   21.1.2     Registering New Feature-tags with IANA ..............  139
   21.1.3     Registered entries ..................................  139
   21.2       RTSP Methods ........................................  140
   21.2.1     Description .........................................  140
   21.2.2     Registering New Methods with IANA ...................  140
   21.2.3     Registered Entries ..................................  140
   21.3       RTSP Status Codes ...................................  141
   21.3.1     Description .........................................  141
   21.3.2     Registering New Status Codes with IANA ..............  141
   21.3.3     Registered Entries ..................................  141
   21.4       RTSP Headers ........................................  141
   21.4.1     Description .........................................  141
   21.4.2     Registering New Headers with IANA ...................  141
   21.4.3     Registered entries ..................................  142
   21.5       Transport Header registries .........................  142



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   21.5.1     Transport Protocol Specification ....................  142
   21.5.2     Transport modes .....................................  144
   21.5.3     Transport Parameters ................................  144
   21.6       Cache Directive Extensions ..........................  145
   21.7       Accept-Credentials ..................................  145
   21.7.1     Accept-Credentials policies .........................  145
   21.7.2     Accept-Credentials hash algorithms ..................  146
   21.8       Range header formats ................................  146
   21.9       URI Schemes .........................................  147
   21.9.1     The rtsp URI Scheme .................................  147
   21.9.2     The rtsps URI Scheme ................................  148
   21.9.3     The rtspu URI Scheme ................................  149
   21.10      SDP attributes ......................................  149
   A          RTSP Protocol State Machine .........................  150
   A.1        States ..............................................  151
   A.2        State variables .....................................  151
   A.3        Abbreviations .......................................  151
   A.4        State Tables ........................................  151
   B          Media Transport Alternatives ........................  155
   B.1        RTP .................................................  155
   B.1.1      AVP .................................................  155
   B.1.2      AVP/UDP .............................................  155
   B.1.3      AVPF/UDP ............................................  156
   B.1.4      SAVP/UDP ............................................  157
   B.1.5      SAVPF/UDP ...........................................  157
   B.2        RTP over TCP ........................................  157
   B.2.1      Interleaved RTP over TCP ............................  157
   B.2.2      RTP over independent TCP ............................  158
   B.2.3      Handling NPT Jumps in the RTP Media Layer ...........  161
   B.2.4      Handling RTP Timestamps after PAUSE .................  163
   B.2.5      RTSP / RTP Integration ..............................  166
   B.2.6      Scaling with RTP ....................................  166
   B.2.7      Maintaining NPT synchronization with RTP
   timestamps .....................................................  166
   B.2.8      Continuous Audio ....................................  166
   B.2.9      Multiple Sources in an RTP Session ..................  166
   B.2.10     Usage of SSRCs and the RTCP BYE Message During an
   RTSP Session ...................................................  166
   B.3        Future Additions ....................................  167
   C          Use of SDP for RTSP Session Descriptions ............  167
   C.1        Definitions .........................................  168
   C.1.1      Control URI .........................................  168
   C.1.2      Media Streams .......................................  169
   C.1.3      Payload Type(s) .....................................  170
   C.1.4      Format-Specific Parameters ..........................  170
   C.1.5      Directionality of media stream ......................  170
   C.1.6      Range of Presentation ...............................  171
   C.1.7      Time of Availability ................................  172



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   C.1.8      Connection Information ..............................  172
   C.1.9      Entity Tag ..........................................  172
   C.2        Aggregate Control Not Available .....................  173
   C.3        Aggregate Control Available .........................  174
   C.4        RTSP external SDP delivery ..........................  175
   D          Minimal RTSP implementation .........................  175
   D.1        Minimal Core Implementation .........................  175
   D.2        Recommended Core Implementation .....................  176
   D.3        The Basic Playback Feature Support ..................  176
   D.3.1      Client ..............................................  176
   D.3.2      Server ..............................................  177
   D.3.3      Proxy ...............................................  177
   D.4        Secure Transport ....................................  178
   E          Requirements for Unreliable Transport of RTSP
   messages .......................................................  178
   F          Backwards Compatibility Considerations ..............  179
   F.1        Play Request in Play mode ...........................  179
   F.2        Using Persistent Connections ........................  180
   G          Open Issues .........................................  180
   H          Changes .............................................  181
     H.1        Changes needing to be updated .....................  187
   I          Author Addresses ....................................  188
   J          Contributors ........................................  188
   K          Acknowledgements ....................................  189
   L          Normative References ................................  189
   M          Informative References ..............................  191

























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1 Introduction

1.1 RTSP Specification Update

   This memorandum specifies RTSP 2.0 which is an update of RTSP 1.0, a
   proposed standard defined in RFC 2326 [25]. The goal of this version
   is to correct the many flaws that have been identified in RTSP 1.0
   since its publication. The corrections are such that backwards
   compatibility was impossible. Thus a new version was deemed the most
   appropriate solution to get a more functional protocol. There are no
   plans to revise RTSP 1.0. Appendix H catalogs the changes of this
   version in relation to RTSP 1.0.

   RTSP 2.0 has reduced functionality compared to RTSP 1.0 and aims at
   specifying the RTSP core, functionality and rules for extensions, and
   basic interaction with the media delivery protocol RTP.

   Any other functionality would be need to be published as extension
   documents. This specification provides rules for such extensions and
   defines registries to avoid naming collisions.

1.2 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls one
   or several time-synchronized streams of continuous media such as
   audio and video. Put simply, RTSP acts as a "network remote control"
   for multimedia servers.

   There is no notion of an RTSP connection in the protocol. Instead, an
   RTSP server maintains a session labeled by an identifier to associate
   groups of media streams and their states. An RTSP session is not tied
   to a transport-level connection such as a TCP connection. During a
   session, a client may open and close multiple reliable transport
   connections to the server to issue RTSP requests for that session.

   This memorandum describes the use of RTSP over a reliable connection
   based transport level protocol such as TCP. RTSP may be implemented
   over an unreliable connectionless transport protocol such as UDP.
   While nothing in RTSP precludes this, additional definition of this
   problem area needs to be handled as an extension to the core
   specification.


        The mechanisms of RTSP's operation over UDP were left out
        of this spec. because they were poorly defined in RFC 2326
        [25] and the tradeoff in size and complexity of this
        memorandum for a small gain in a limited problem space was
        not deemed justifiable.



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   The set of streams to be controlled in an RTSP session is defined by
   a presentation description. This memorandum does not define a format
   for the presentation description. However appendix C describes how
   SDP [1] is used for this purpose. The streams controlled by RTSP may
   use RTP [2] for their data transport, but the operation of RTSP does
   not depend on the transport mechanism used to carry continuous media.
   RTSP is intentionally similar in syntax and operation to HTTP/1.1 [3]
   so that extension mechanisms to HTTP can in most cases also be
   applied to RTSP. However, RTSP differs in a number of important
   aspects from HTTP:

        o RTSP introduces a number of new methods and has a different
          protocol identifier.

        o RTSP has the notion of a session built into the protocol.

        o An RTSP server needs to maintain state in almost all cases, as
          opposed to the stateless nature of HTTP.

        o Both an RTSP server and client can issue requests.

        o Data is usually carried out-of-band by a different protocol.
          Session descriptions returned in a DESCRIBE response (see
          Section 11.2) and interleaving of RTP with RTSP over TCP are
          exceptions to this rule (see Section 12).

        o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
          8859-1, consistent with HTML internationalization efforts
          [26].

        o The Request-URI always contains the absolute URI. Because of
          backward compatibility with a historical blunder, HTTP/1.1 [3]
          carries only the absolute path in the request and puts the
          host name in a separate header field.


             This makes "virtual hosting" easier, where a single
             host with one IP address hosts several document trees.

   The protocol supports the following operations:

        Retrieval of media from media server: The client can either
             request a presentation description via RTSP DESCRIBE, HTTP
             or some other method. If the presentation is being
             multicast, the presentation description contains the
             multicast addresses and ports to be used for the continuous
             media. If the presentation is to be sent only to the client
             via unicast, the client provides the destination.



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        Invitation of a media server to a conference: A media server can
             be "invited" to join an existing conference to play back
             media into the presentation. This mode is useful, for
             example, in distributed teaching applications. Several
             parties in the conference may take turns "pushing the
             remote control buttons".  Note: This functionality will
             require RTSP external application level functionality.

   RTSP requests may be handled by proxies, tunnels and caches as in
   HTTP/1.1 [3].

1.3 Notational Conventions

   Since many of the definitions and syntax are identical to HTTP/1.1,
   this specification only points to the section where they are defined
   rather than copying it. For brevity, [HX.Y] is to be taken to refer
   to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [3]).

   All the mechanisms specified in this document are described in both
   prose and the Augmented Backus-Naur form (ABNF) described in detail
   in RFC 4234 [4].

   Indented and smaller-type paragraphs are used to provide informative
   background and motivation. This is intended to give readers who were
   not involved with the formulation of the specification an
   understanding of why things are the way they are in RTSP.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [5].

   The word, "unspecified" is used to indicate functionality or features
   that are not defined in this specification. Such functionality cannot
   be used in a standardized manner without further definition in an
   extension specification to RTSP.

1.3.1 RFC Editor Consideration

   Please replace RFC XXXX with the RFC number this specification
   recieves.

   Please replace RFC YYYY with the RFC number that SAVPF [6] receives.

1.4 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [3]. Terms not
   listed here are defined as in HTTP/1.1.




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        Aggregate control: The concept of controlling multiple streams
             using a single timeline, generally maintained by the
             server. A client, for example, uses aggregate control when
             it issues a single play or pause message to simultaneously
             control both the audio and video in a movie. A session
             which is under aggregate control is referred to as an
             aggregated session.

        Aggregate control URI: The URI used in an RTSP request to refer
             to and control an aggregated session. It normally, but not
             always, corresponds to the presentation URI specified in
             the session description. See Section 11.3 for more
             information.

        Conference: a multiparty, multimedia presentation, where "multi"
             implies greater than or equal to one.

        Client: The client requests media service from the media server.

        Connection: A transport layer virtual circuit established
             between two programs for the purpose of communication.

        Container file: A file which may contain multiple media streams
             which often constitutes a presentation when played
             together. The concept of a container file is not embedded
             in the protocol.  However, RTSP servers may offer aggregate
             control on the media streams within these files.

        Continuous media: Data where there is a timing relationship
             between source and sink; that is, the sink needs to
             reproduce the timing relationship that existed at the
             source. The most common examples of continuous media are
             audio and motion video.  Continuous media can be real-time
             (interactive or conversational), where there is a "tight"
             timing relationship between source and sink, or streaming
             (playback), where the relationship is less strict.

        Entity: The information transferred as the payload of a request
             or response. An entity consists of meta-information in the
             form of entity-header fields and content in the form of an
             entity-body, as described in Section 8.

        Feature-tag: A tag representing a certain set of functionality,
             i.e. a feature.

        IRI: Internationalized Resource Identifier, is the same as an
             URI, with the exception that it allows characters from the
             whole Universal Character Set (Unicode/ISO 10646), rather



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             than the US-ASCII only. See RFC 3987 [7] for more
             information.

        Live: Normally used to describe a presentation or session with
             media coming from an ongoing event. This generally results
             in the session having an unbound or only loosely defined
             duration, and sometimes no seek operations are possible.

        Media initialization: Datatype/codec specific initialization.
             This includes such things as clock rates, color tables,
             etc. Any transport-independent information which is
             required by a client for playback of a media stream occurs
             in the media initialization phase of stream setup.

        Media parameter: Parameter specific to a media type that may be
             changed before or during stream playback.

        Media server: The server providing playback services for one or
             more media streams. Different media streams within a
             presentation may originate from different media servers.  A
             media server may reside on the same host or on a different
             host from which the presentation is invoked.

        Media server indirection: Redirection of a media client to a
             different media server.

        (Media) stream: A single media instance, e.g., an audio stream
             or a video stream as well as a single whiteboard or shared
             application group. When using RTP, a stream consists of all
             RTP and RTCP packets created by a source within an RTP
             session.

        Message: The basic unit of RTSP communication, consisting of a
             structured sequence of octets matching the syntax defined
             in Section 19 and transmitted over a connection or a
             connectionless transport.

        Non-Aggregated Control: Control of a single media stream.  This
             is only possible in RTSP sessions with a single media.

        Participant: Member of a conference. A participant may be a
             machine, e.g., a playback server.

        Presentation: A set of one or more streams presented to the
             client as a complete media feed and described by a
             presentation description as defined below. Presentations
             with more than one media stream are often handled in RTSP
             under aggregate control.



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        Presentation description: A presentation description contains
             information about one or more media streams within a
             presentation, such as the set of encodings, network
             addresses and information about the content. Other IETF
             protocols such as SDP (RFC 4566 [1]) use the term "session"
             for a presentation. The presentation description may take
             several different formats, including but not limited to the
             session description protocol format, SDP.

        Response: An RTSP response. If an HTTP response is meant, that
             is indicated explicitly.

        Request: An RTSP request. If an HTTP request is meant, that is
             indicated explicitly.

        Request-URI: The URI used in a request to indicate the resource
             on which the request is to be performed.

        RTSP agent: Refers to either an RTSP client, an RTSP server, or
             an RTSP Proxy. In this specification, there are many
             capabilities that are common to these three entities such
             as the capability to send requests or receive responses.
             This term will be used when describing functionality that
             is applicable to all three of these entities.

        RTSP session: A stateful abstraction upon which the main control
             methods of RTSP operate. An RTSP session is a server
             entity; it is created, maintained and destroyed by the
             server. It is established by an RTSP server upon the
             completion of a successful SETUP request (when a 200 OK
             response is sent) and is labelled with a session identifier
             at that time. The session exists until timed out by the
             server or explicitly removed by a TEARDOWN request. An RTSP
             session is a stateful entity; an RTSP server maintains an
             explicit session state machine (see Appendix  A) where most
             state transitions are triggered by client requests. The
             existence of a session implies the existence of state about
             the session's media streams and their respective transport
             mechanisms. A given session can have one or more media
             streams associated with it.  An RTSP server uses the
             session to aggregate control over multiple media streams.

        Transport initialization: The negotiation of transport
             information (e.g., port numbers, transport protocols)
             between the client and the server.

        URI: Universal Resource Identifier, see RFC 3986 [8]. The URIs
             used in RTSP are generally URLs as they give a location for



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             the resource. As URLs are a subset of URIs, they will be
             referred to as URIs to cover also the cases when an RTSP
             URI would not be an URL.

        URL: Universal Resource Locator, is an URI which identifies the
             resource through its primary access mechanism, rather than
             identifying the resource by name or by some other
             attribute(s) of that resource.

1.5 Protocol Properties

   RTSP has the following properties:

        Extendable: New methods and parameters can be easily added to
             RTSP.

        Easy to parse: RTSP can be parsed by standard HTTP or MIME
             parsers.

        Secure: RTSP re-uses web security mechanisms, either at the
             transport level (TLS, RFC 4346 [9]) or within the protocol
             itself. All HTTP authentication mechanisms such as basic
             (RFC 2616 [3]) and digest authentication (RFC 2617 [10])
             are directly applicable.

        Transport-independent: RTSP does not preclude the use of
             unreliable datagram protocol (UDP) (RFC 768 [11]) as it
             would be possible to implement application-level
             reliability. The use of a connectionless datagram protocol
             such as UDP requires additional definition that may be
             provided as extensions to the core RTSP specification. The
             reliable stream protocol TCP (RFC 793 [12]) and the secured
             reliable stream protocol TLS over TCP [9] are the currently
             defined transport protocols for RTSP messages.

        Media-delivery protocol independent: The operation of RTSP does
             not depend on the transport mechanism used to carry
             continuous media. While most real-time media will use RTP
             as a transport protocol, RTSP does not preclude the use of
             other protocols such as MPEG-2 [27]. The use of other
             protocols requires additional definition that may be
             provided as extensions to the core RTSP specification.

        Multi-server capable: Each media stream within a presentation
             can reside on a different server. The client automatically
             establishes several concurrent control sessions with the
             different media servers. Media synchronization in those
             cases is performed at the transport level.



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        Separation of stream control and conference initiation:  Stream
             control is divorced from inviting a media server to a
             conference. In particular, SIP [28] or H.323 [29] may be
             used to invite a server to a conference; however, the exact
             procedures are unspecified.

        Suitable for professional applications: RTSP supports frame-
             level accuracy through SMPTE time stamps to allow remote
             digital editing.

        Presentation description neutral: The protocol does not impose a
             particular presentation description or metafile format and
             can convey the type of format to be used. However, the
             presentation description is required to contain at least
             one RTSP URI.

        Proxy and firewall friendly: The protocol should be readily
             handled by both application and transport-layer (SOCKS
             [30]) firewalls. A firewall may need to understand the
             SETUP method to open a "hole" for the media stream.

        HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so
             that the existing infrastructure can be reused. This
             infrastructure includes PICS (Platform for Internet Content
             Selection [31,32]) for associating labels with content.
             However, RTSP does not just add methods to HTTP since
             controlling continuous media requires server state in most
             cases.

        Appropriate server control: If a client can start a stream, it
             needs to be able to stop a stream. Servers should not start
             streaming to clients in such a way that clients cannot stop
             the stream.

        Transport negotiation: The client can negotiate the transport
             method prior to actually needing to process a continuous
             media stream.

1.6 Extending RTSP

   Since not all media servers have the same functionality, media
   servers by necessity will support different sets of requests. For
   example:

        o A server may not be capable of seeking (absolute positioning)
          if it is to support live events only.

        o Some servers may not support setting stream parameters and



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          thus not support GET_PARAMETER and SET_PARAMETER.

        o Some server may support an RTSP extension.

   It is up to the creators of presentation descriptions not to ask the
   impossible of a server. This situation is similar in HTTP/1.1 [3],
   where the methods described in [H19.5] are not likely to be supported
   across all servers.

   RTSP can be extended in three ways, listed here in order of the
   magnitude of changes supported:

        o Existing methods can be extended with new parameters, e.g.
          headers, as long as these parameters can be safely ignored by
          the recipient. If the client needs negative acknowledgement
          when a method extension is not supported, a tag corresponding
          to the extension may be added in the field of the Require or
          Proxy-Require headers (see Section 14.37).

        o New methods can be added. If the recipient of the message does
          not understand the request, it MUST respond with error code
          501 (Not Implemented) so that the sender can avoid using this
          method again. A client may also use the OPTIONS method to
          inquire about methods supported by the server. The server MUST
          list the methods it supports using the Public response header.

        o A new version of the protocol can be defined, allowing almost
          all aspects (except the position of the protocol version
          number) to change.  A new version of the protocol MUST be
          registered through an IETF standard track document.

   The basic capability discovery mechanism can be used to both discover
   support for a certain feature and to ensure that a feature is
   available when performing a request. For detailed explanation of this
   see section 10.

1.7 Overall Operation

   Each presentation and media stream is identified by an RTSP URI.  The
   overall presentation and the properties of the media the presentation
   is composed of are defined by a presentation description file, the
   format of which is outside the scope of this specification. The
   presentation description file may be obtained by the client using
   HTTP or other means such as email and may not necessarily be stored
   on the media server.

   For the purposes of this specification, a presentation description is
   assumed to describe one or more presentations, each of which



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   maintains a common time axis. For simplicity of exposition and
   without loss of generality, it is assumed that the presentation
   description contains exactly one such presentation. A presentation
   may contain several media streams.

   The presentation description file contains a description of the media
   streams making up the presentation, including their encodings,
   language, and other parameters that enable the client to choose the
   most appropriate combination of media. In this presentation
   description, each media stream that is individually controllable by
   RTSP is identified by an RTSP URI, which points to the media server
   handling that particular media stream and names the stream stored on
   that server. Several media streams can be located on different
   servers; for example, audio and video streams can be split across
   servers for load sharing. The description also enumerates which
   transport methods the server is capable of.

   Besides the media parameters, the network destination address and
   port need to be determined. Several modes of operation can be
   distinguished:

        Unicast: The media is transmitted to the source of the RTSP
             request or the requested destination, with the port number
             chosen by the client. Alternatively, the media is
             transmitted on the same reliable stream as RTSP.

        Multicast, server chooses address: The media server picks the
             multicast address and port. This is the typical case for a
             live or near-media-on-demand transmission.

        Multicast, client chooses address: If the server is to
             participate in an existing multicast conference, the
             multicast address, port and encryption key are given by the
             conference description, established by means outside the
             scope of this specification, for example by a SIP created
             conference.

1.8 RTSP States

   RTSP controls a stream which may be sent via a separate protocol,
   independent of the control channel. For example, RTSP control may be
   transported on a TCP connection while the media data is conveyed via
   UDP. Thus, data delivery continues even if no RTSP requests are
   received by the media server. Also, during its lifetime a single
   media stream may be controlled by RTSP requests issued sequentially
   on different TCP connections. Therefore, the server needs to maintain
   "session state" to be able to correlate RTSP requests with a stream.
   The state transitions are described in Appendix A.



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   Many methods in RTSP do not contribute to state. However, the
   following play a central role in defining the allocation and usage of
   stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, and
   TEARDOWN.

        SETUP: Causes the server to allocate resources for a stream and
             create an RTSP session.

        PLAY: Starts data transmission on a stream allocated via SETUP.

        PAUSE: Temporarily halts a stream without freeing server
             resources.

        REDIRECT: Indicates that the session should be moved to a new
             server or location

        TEARDOWN: Frees resources associated with the stream.  The RTSP
             session ceases to exist on the server.

   RTSP methods that contribute to state use the Session header field
   (Section 14.42) to identify the RTSP session whose state is being
   manipulated. The server generates session identifiers in response to
   SETUP requests (Section 11.3).

1.9 Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP. It also may
   interact with HTTP in that the initial contact with streaming content
   will often be made through a web page. The current protocol
   specification aims to allow different hand-off points between a web
   server and the media server implementing RTSP. For example, the
   presentation description can be retrieved using HTTP or RTSP, which
   reduces round trips in web-browser-based scenarios, yet also allows
   for stand alone RTSP servers and clients which do not rely on HTTP at
   all. However, RTSP differs fundamentally from HTTP in that most data
   delivery takes place out-of-band in a different protocol. HTTP is an
   asymmetric protocol where the client issues requests and the server
   responds. In RTSP, both the media client and media server can issue
   requests. RTSP requests are also stateful; they may set parameters
   and continue to control a media stream long after the request has
   been acknowledged.


        Re-using HTTP functionality has advantages in at least two
        areas, namely security and proxies. The requirements are
        very similar, so having the ability to adopt HTTP work on
        caches, proxies and authentication is valuable.




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   RTSP assumes the existence of a presentation description format that
   can express both static and temporal properties of a presentation
   containing several media streams. Session Description Protocol (SDP)
   [1] is generally the format of choice; however, RTSP is not bound to
   it. For data delivery, most real-time media will use RTP as a
   transport protocol. While RTSP works well with RTP, it is not tied to
   RTP.

2 RTSP Use Cases

   This section describes the most important and considered use cases
   for RTSP. They are listed in descending order of importance in
   regards to ensuring that all necessary functionality is present.
   This specification only fully supports usage of the two first.  Also
   in these first two cases, there are special cases or exceptions that
   are not supported without extensions, e.g. the redirection of media
   to another address than the controlling entity.

2.1 On-demand Playback of Stored Content

   An RTSP capable server stores content suitable for being streamed to
   a client. A client desiring playback of any of the stored content
   uses RTSP to set up the media transport required to deliver the
   desired content. RTSP is then used to initiate, halt and manipulate
   the actual transmission (playout) of the content.  RTSP is also
   required to provide necessary description and synchronization
   information for the content.

   The above high level description can be broken down into a number of
   functions that RTSP needs to be capable of.

        Presentation Description: Provide initialization information
             about the presentation (content); for example, which media
             codecs are needed for the content. Other information that
             is important includes the number of media stream the
             presentation contains, the transport protocols used for the
             media streams, and identifiers for these media streams.
             This information is required before setup of the content is
             possible and to determine if the client is even capable of
             using the content.

             This information need not be sent using RTSP; other
             external protocols can be used to transmit the transport
             presentation descriptions. Two good examples are the use of
             HTTP [3] or email to fetch or receive presentation
             descriptions like SDP [1].

        Setup: Set up some or all of the media streams in a



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             presentation. The setup itself consist of selecting the
             protocol for media transport and the necessary parameters
             for the protocol, like addresses and ports.

        Control of Transmission: After the necessary media streams have
             been established the client can request the server to start
             transmitting the content. The client must be allowed to
             start or stop the transmission of the content at arbitrary
             times.  The client must also be able to start the
             transmission at any point in the timeline of the
             presentation.

        Synchronization: For media transport protocols like RTP [16] it
             might be beneficial to carry synchronization information
             within RTSP. This may be due to either the lack of inter-
             media synchronization within the protocol itself, or the
             potential delay before the synchronization is established
             (which is the case for RTP when using RTCP).

        Termination Terminate the established contexts.

   For this use case there are a number of assumptions about how it
   works. These are:

        On-Demand content: The content is stored at the server and can
             be accessed at any time during a time period when it is
             intended to be available.

        Independent sessions: A server is capable of serving a number of
             clients simultaneously, including from the same piece of
             content at different points in that presentations time-
             line.

        Unicast Transport: Content for each individual client is
             transmitted to them using unicast traffic.

   It is also possible to redirect the media traffic to a different
   destination than that of the entity controlling the traffic. However,
   allowing this without appropriate mechanisms for checking that the
   destination approves of this allows for distributed denial of service
   attacks (DDoS).

2.2 Unicast distribution of Live Content

   This use cases is similar to the above on-demand content case (see
   section 2.1; the difference is the nature of the content itself.
   Live content is continuously distributed as it becomes available from
   a source; i.e., the main difference from on-demand is that one starts



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   distributing content before the end of it has become available to the
   server.

   In many cases the consumer of live content is only interested in
   consuming what is actually happens "now"; i.e., very similar to
   broadcast TV. However in this case it is assumed that there exist no
   broadcast or multicast channel to the users, and instead the server
   functions as a distribution node, sending the same content to
   multiple receivers, using unicast traffic between server and client.
   This unicast traffic and the transport parameters are individually
   negotiated for each receiving client.

   Another aspect of live content is that it often has a very limited
   time of availability, as it is only is available for the duration of
   the event the content covers. An example of such a live content could
   be a music concert which lasts 2 hour and starts at a predetermined
   time. Thus there is need to announce when and for how long the live
   content is available.

2.3 On-demand Playback using Multicast

   It is possible to use RTSP to request that media be delivered to a
   multicast group. The entity setting up the session (the controller)
   will then control when and what media is delivered to the group. This
   use case has some potential for denial of service attacks by flooding
   a multicast group. Therefore, a mechanism is needed to indicate that
   the group actually accepts the traffic from the RTSP server.

   An open issue in this use case is how one ensures that all receivers
   listening to the multicast or broadcast receives the session
   presentation configuring the receivers.

2.4 Inviting an RTSP server into a conference

   If one has an established conference or group session, it is possible
   to have an RTSP server distribute media to the whole group.
   Transmission to the group is simplest when controlled by a single
   participant or leader of the conference. Shared control might be
   possible, but would require further investigation and possibly
   extensions.

   This use case assumes that there exists either multicast or a
   conference focus that redistribute media to all participants.

   This use case is intended to be able to handle the following
   scenario: A conference leader or participant (hereafter called the
   controller) has some pre-stored content on an RTSP server that he
   wants to share with the group. The controller sets up an RTSP session



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   at the streaming server for this content and retrieves the session
   description for the content.  The destination for the media content
   is set to the shared multicast group or conference focus.  When
   desired by the controller, he/she can start and stop the transmission
   of the media to the conference group.

   There are several issues with this use case that are not solved by
   this core specification for RTSP:

        o To avoid an RTSP server from being an unknowing participant in
          a denial of service attack the server needs to be able to
          verify the destination's acceptance of the media. Such a
          mechanism to verify the approval of received media does not
          yet exist; instead, only policies can be used, which can be
          made to work in controlled environments.

        o To enable a media receiver to correctly decode the content the
          media configuration information needs to be distributed
          reliably to all participants. This will most likely require
          support from an external protocol.

        o If it is desired to pass control of the RTSP session between
          the participants, some support will be required by an external
          protocol to exchange state information and possibly floor
          control of who is controlling the RTSP session.

   If there interest in this use case, further work is required on the
   necessary extensions.

2.5 Live Content using Multicast

   This use case in its simplest form does not require any use of RTSP
   at all; this is what multicast conferences being announced with SAP
   and SDP are intended to handle. However in use cases where more
   advanced features like access control to the multicast session are
   desired, RTSP could be used for session establishment.

   A client desiring to join a live multicasted media session with
   cryptographic (encryption) access control could use RTSP in the
   following way. The source of the session announces the session and
   gives all interested an RTSP URI. The client connects to the server
   and requests the presentation description, allowing configuration for
   reception of the media. In this step it is possible for the client to
   use secured transport and any desired level of authentication; for
   example, for billing or access control. An RTSP link also allows for
   load balancing between multiple servers.

   If these were the only goals, they could be achieved by simply using



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   HTTP.  However, for cases where the sender likes to keep track of
   each individual receiver of a session, and possibly use the session
   as a side channel for distributing key-updates or other information
   on a per-receiver basis, and the full set of receivers is not know
   prior to the session start, the state establishment that RTSP
   provides can be beneficial. In this case a client would establish an
   RTSP session to the multicast group. The RTSP server will not
   transmit any media, but instead will point to the multicast group.
   The client and server will be able to keep the session alive for as
   long as the receiver participates in the session thus enabling, for
   example, the server to push updates to the client.

   This use case will most likely not be able to be implemented without
   some extensions to the server-to-client push mechanism. Here a method
   like ANNOUNCE (see RFC 2326 [25] might be suitable; however, it will
   require a RTSP extension to revive the method.

3 Protocol Parameters

3.1 RTSP Version

   HTTP specification section [H3.1] applies, with "HTTP" replaced by
   "RTSP". This specification defines version 2.0 of RTSP.

3.2 RTSP IRI and URI

   RTSP 2.0 defines and registers three URI schemas "rtsp", "rtsps" and
   "rtspu". The usage of the last, "rtspu", is unspecified in RTSP 2.0,
   and is defined here to register and reserve the URI scheme that is
   defined in RTSP 1.0. The "rtspu" scheme indicates transport of the
   RTSP messages over unreliable transport (UDP). The syntax of "rtsp"
   and "rtsps" URIs has been changed from RTSP 1.0.

   This specification also defines the format of the RTSP IRI [7] that
   can be used as RTSP resource identifiers and locators, in web pages,
   user interfaces, on paper, etc. However, the RTSP request message
   format only allows usage of the absolute URI format. The RTSP IRI
   format SHALL use the rules and transformation for IRIs defined in RFC
   3987 [7]. This way RTSP 2.0 URIs for request can be produced from an
   RTSP IRI.

   The RTSP IRI and URI are both syntax restricted compared to the
   generic syntax defined in RFC 3986 [8] and RFC 3987 [7]:

        o An absolute URI requires the authority part; i.e., a host
          identity must be provided.

        o Parameters in the path element are prefixed with the reserved



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          separator ";".

   The RTSP URI and IRI is case sensitive, with the exception of those
   parts that RFC 3986 [8] and RFC 3987 [7] defines as case-insensitive;
   for example, the scheme and host part.

   The fragment identifier is used as defined in sections 3.5 and 4.3 of
   [8], i.e. the fragment is to be stripped from the URI by the
   requestor and not included in the request. The user agent also needs
   to interpret the value of the fragment based on the media type the
   request relates to; i.e., the media type indicated in Content-Type
   header in the response to DESCRIBE.

   The syntax of any URI query string is unspecified and responder
   (usually the server) specific. The query is, from the requestor's
   perspective, an opaque string and needs to be handled as such.

   The URI scheme "rtsp" requires that commands are issued via a
   reliable protocol (within the Internet, TCP), while the scheme
   "rtsps" identifies a reliable transport using secure transport (TLS
   [9], see Section 18).

   For the scheme "rtsp", if no port number is provided in the authority
   part of the URI port number 554 SHALL be used.  For the scheme
   "rtsps", the TCP port 322 is registered and SHALL be assumed.

   A presentation or a stream is identified by a textual media
   identifier, using the character set and escape conventions of URIs
   (RFC 3986 [8]). URIs may refer to a stream or an aggregate of
   streams; i.e., a presentation. Accordingly, requests described in
   Section 11 can apply to either the whole presentation or an
   individual stream within the presentation. Note that some request
   methods can only be applied to streams, not presentations, and vice
   versa.

   For example, the RTSP URI:

     rtsp://media.example.com:554/twister/audiotrack


   may identify the audio stream within the presentation "twister",
   which can be controlled via RTSP requests issued over a TCP
   connection to port 554 of host media.example.com

   Also, the RTSP URI:

     rtsp://media.example.com:554/twister




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   identifies the presentation "twister", which may be composed of audio
   and video streams, but could also be something else like a random
   media redirector.


        This does not imply a standard way to reference streams in
        URIs. The presentation description defines the hierarchical
        relationships in the presentation and the URIs for the
        individual streams. A presentation description may name a
        stream "a.mov" and the whole presentation "b.mov".

   The path components of the RTSP URI are opaque to the client and do
   not imply any particular file system structure for the server.


        This decoupling also allows presentation descriptions to be
        used with non-RTSP media control protocols simply by
        replacing the scheme in the URI.

3.3 Session Identifiers

   Session identifiers are strings of any arbitrary length. A session
   identifier MUST be chosen randomly and MUST be at least eight
   characters long to make guessing it more difficult. (See Section 20.)

3.4 SMPTE Relative Timestamps

   A SMPTE relative timestamp expresses time relative to the start of
   the clip. Relative timestamps are expressed as SMPTE time codes for
   frame-level access accuracy. The time code has the format
                  hours:minutes:seconds:frames.subframes,
   with the origin at the start of the clip. The default smpte format is
   "SMPTE 30 drop" format, with frame rate is 29.97 frames per second.
   Other SMPTE codes MAY be supported (such as "SMPTE 25") through the
   use of alternative use of "smpte-type". For SMPTE 30, the "frames"
   field in the time value can assume the values 0 through 29. The
   difference between 30 and 29.97 frames per second is handled by
   dropping the first two frame indices (values 00 and 01) of every
   minute, except every tenth minute. If the frame and the subframe
   values are zero, they may be omitted. Subframes are measured in one-
   hundredth of a frame.

   Examples:

     smpte=10:12:33:20-
     smpte=10:07:33-
     smpte=10:07:00-10:07:33:05.01
     smpte-25=10:07:00-10:07:33:05.01



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3.5 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position
   relative to the beginning of the presentation, not to be confused
   with the Network Time Protocol (NTP) [33]. The timestamp consists of
   a decimal fraction. The part left of the decimal may be expressed in
   either seconds or hours, minutes, and seconds. The part right of the
   decimal point measures fractions of a second.

   The beginning of a presentation corresponds to 0.0 seconds.  Negative
   values are not defined. The special constant "now" is defined as the
   current instant of a live event. It MAY only be used for live events,
   and SHALL NOT be used for on-demand (i.e., non-live) content.

   NPT is defined as in DSM-CC [34]:  "Intuitively, NPT is the clock the
   viewer associates with a program. It is often digitally displayed on
   a VCR. NPT advances normally when in normal play mode (scale = 1),
   advances at a faster rate when in fast scan forward (high positive
   scale ratio), decrements when in scan reverse (high negative scale
   ratio) and is fixed in pause mode. NPT is (logically) equivalent to
   SMPTE time codes."

   Examples:

     npt=123.45-125
     npt=12:05:35.3-
     npt=now-




        The syntax conforms to ISO 8601 [35]. The npt-sec notation
        is optimized for automatic generation, the npt-hhmmss
        notation for consumption by human readers. The "now"
        constant allows clients to request to receive the live feed
        rather than the stored or time-delayed version. This is
        needed since neither absolute time nor zero time are
        appropriate for this case.

3.6 Absolute Time

   Absolute time is expressed as ISO 8601 [35] timestamps, using UTC
   (GMT). Fractions of a second may be indicated.

   Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
   UTC:

     19961108T143720.25Z



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3.7 Feature-tags

   Feature-tags are unique identifiers used to designate features in
   RTSP. These tags are used in Require (Section 14.37), Proxy-Require
   (Section 14.31), Proxy-Supported (Section 14.32), Unsupported
   (Section 14.46), and Supported (Section 14.43) header fields.

   A feature-tag definition MUST indicate which combination of clients,
   servers or proxies they applies too.

   The creator of a new RTSP feature-tag should either prefix the
   feature-tag with a reverse domain name (e.g.,
   "com.example.mynewfeature" is an apt name for a feature whose
   inventor can be reached at "example.com"), or register the new
   feature-tag with the Internet Assigned Numbers Authority (IANA) (see
   IANA Section  21).

   The usage of feature-tags is further described in section 10 that
   deals with capability handling.

3.8 Entity Tags

   Entity tags are opaque strings that are used to compare two entities
   from the same resource, for example in caches or to optimize setup
   after a redirect. Further explanation is present in [H3.11]. For an
   explanation of how to compare entity tags see [H13.3]. Entity tags
   can be carried in the ETag header (see section 14.21) or in SDP (see
   section C.1.9).

   Entity tags are used in RTSP to make some methods conditional. The
   methods are made conditional through the inclusion of headers, see
   14.24 and 14.26. Note that RTSP entity tags apply to the complete
   presentation; i.e., both the session description and the individual
   media streams. Thus entity tags can be used to verify at setup time
   after a redirect that the same session description applies to the
   media at the new location using the If-Match header.

4 RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 3629 [13]). Lines SHALL be terminated by CRLF.


        Text-based protocols make it easier to add optional
        parameters in a self-describing manner. Since the number of
        parameters and the frequency of commands is low, processing
        efficiency is not a concern. Text-based protocols, if done
        carefully, also allow easy implementation of research



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        prototypes in scripting languages such as Tcl, Visual Basic
        and Perl.

   The ISO 10646 character set avoids tricky character set switching,
   but is invisible to the application as long as US-ASCII is being
   used.  This is also the encoding used for RTCP. ISO 8859-1 translates
   directly into Unicode with a high-order octet of zero. ISO 8859-1
   characters with the most-significant bit set are represented as
   1100001x 10xxxxxx. (See RFC 3629 [13])

   Requests contain methods, the object the method is operating upon and
   parameters to further describe the method. Methods are idempotent
   unless otherwise noted. Methods are also designed to require little
   or no state maintenance at the media server.

4.1 Message Types

   See [H4.1].

4.2 Message Headers

   See [H4.2].

4.3 Message Body

   See [H4.3]

   Unlike HTTP, the presence of a message-body in either a request or a
   response MUST be signaled by the inclusion of a Content-Length header
   field (see section 14.16).

4.4 Message Length

   When a message body is included with a message, the length of that
   body is determined by one of the following (in order of precedence):

        1.   Any response message which MUST NOT include a message body
             (such as the 1xx, 204, and 304 responses) is always
             terminated by the first empty line after the header fields,
             regardless of the entity-header fields present in the
             message. (Note: An empty line is a line with nothing
             preceding the CRLF.)

        2.   If a Content-Length header field (section 14.16) is
             present, its value in bytes represents the length of the
             message-body. If this header field is not present, a value
             of zero is assumed.




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   Unlike an HTTP message, an RTSP message MUST contain a Content-Length
   header field whenever it contains a message body. Note that RTSP does
   not support the HTTP/1.1 "chunked" transfer coding (see [H3.6.1]).


        Given the moderate length of presentation descriptions
        returned, the server should always be able to determine its
        length, even if it is generated dynamically, making the
        chunked transfer encoding unnecessary.

5 General Header Fields

   See [H4.5], except that the Pragma, Trailer, Transfer-Encoding,
   Upgrade, and Warning headers are not defined. RTSP further defines
   the CSeq, Proxy-Supported and Timestamp headers. The general headers
   are listed in table 1:


                    Header Name      Defined in Section
                    ___________________________________
                    Cache-Control    Section 14.10
                    Connection       Section 14.11
                    CSeq             Section 14.19
                    Date             Section 14.20
                    Proxy-Supported  Section 14.32
                    Supported        Section 14.43
                    Timestamp        Section 14.44
                    Via              Section 14.49


   Table 1: The General headers used in RTSP.


6 Request

   A request message uses the format outlined below regardless of the
   direction of a request, client to server or server to client:

        o Request line, containing the method to be applied to the
          resource, the identifier of the resource, and the protocol
          version in use;

        o Zero or more Header lines, that can be of the following types:
          general (Section 5), request (Section 6.2), or entity (section
          8.1);

        o One empty line (CRLF) to indicate the end of the header
          section;



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        o Optionally a message body (entity), consisting of one or more
          lines. The length of the message body in bytes is indicated by
          the Content-Length entity header.

6.1 Request Line

   The request line provides the key information about the request:
   what method, on what resources and using which RTSP version.  The
   methods that are defined by this specification are listed in Table 2.


                     Method         Defined In Section
                     _________________________________
                     DESCRIBE       Section 11.2
                     GET_PARAMETER  Section 11.7
                     OPTIONS        Section 11.1
                     PAUSE          Section 11.5
                     PLAY           Section 11.4
                     REDIRECT       Section 11.9
                     SETUP          Section 11.3
                     SET_PARAMETER  Section 11.8
                     TEARDOWN       Section 11.6


   Table 2: The RTSP Methods


   The syntax of the RTSP request line is the following:


   <Method> SP <Request-URI> SP <RTSP-Version> CRLF


   Note: This syntax cannot be freely changed in future versions of
   RTSP. This line needs to remain parsable by older RTSP
   implementations since it indicates the RTSP version of the message.

   In contrast to HTTP/1.1 [3], RTSP requests identify the resource
   through an absolute RTSP URI (scheme, host, and port) (see section
   3.2) rather than just the absolute path.


        HTTP/1.1 requires servers to understand the absolute URI,
        but clients are supposed to use the Host request header.
        This is purely needed for backward-compatibility with
        HTTP/1.0 servers, a consideration that does not apply to
        RTSP.




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   An asterisk "*" can be used instead of an absolute URI in the
   Request-URI part to indicate that the request does not apply to a
   particular resource, but to the server or proxy itself, and is only
   allowed when the request method does not necessarily apply to a
   resource.

   For example:

     OPTIONS * RTSP/2.0



   An OPTIONS in this form will determine the capabilities of the server
   or the proxy that first receives the request. If the capability of
   the specific server needs to be determined, without regard to the
   capability of an intervening proxy, the server should be addressed
   explicitly with an absolute URI that contains the server's address.

   For example:


     OPTIONS rtsp://example.com RTSP/2.0



6.2 Request Header Fields

   The RTSP headers in Table 3 can be included in a request, as request
   headers, to modify the specifics of the request. Some of these
   headers may also be used in the response to a request, as response
   headers, to modify the specifics of a response (Section 7.2).


   Detailed headers definition are provided in Section 14.

   New request headers may be defined. If the receiver of the request is
   required to understand the request header, the request MUST include a
   corresponding feature tag in a Require or Proxy-Require header to
   ensure the correct processing of the header.

7 Response

   [H6] applies except that HTTP-Version is replaced by RTSP-Version.
   Also, RTSP defines additional status codes and does not define some
   of the HTTP codes. The valid response codes and the methods they can
   be used with are listed in Table 4.

   After receiving and interpreting a request message, the recipient



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                  Header              Defined in Section
                  ______________________________________
                  Accept              Section 14.1
                  Accept-Credentials  Section 14.2
                  Accept-Encoding     Section 14.3
                  Accept-Language     Section 14.4
                  Authorization       Section 14.7
                  Bandwidth           Section 14.8
                  Blocksize           Section 14.9
                  From                Section 14.23
                  If-Match            Section 14.24
                  If-Modified-Since   Section 14.25
                  If-None-Match       Section 14.26
                  Proxy-Require       Section 14.31
                  Range               Section 14.34
                  Referer             Section 14.35
                  Require             Section 14.37
                  Scale               Section 14.39
                  Session             Section 14.42
                  Speed               Section 14.40
                  Supported           Section 14.43
                  Transport           Section 14.45
                  User-Agent          Section 14.47


   Table 3: The RTSP request headers

   responds with an RTSP response message.

7.1 Status-Line

   The first line of a Response message is the Status-Line, consisting
   of the protocol version followed by a numeric status code and the
   textual phrase associated with the status code, with each element
   separated by SP characters. No CR or LF is allowed except in the
   final CRLF sequence.


   <RTSP-Version> SP <Status-Code> SP <Reason-Phrase> CRLF


7.1.1 Status Code and Reason Phrase

   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request. These codes are fully
   defined in Section 13. The Reason-Phrase is intended to give a short
   textual description of the Status-Code. The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human


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   user. The client is not required to examine or display the Reason-
   Phrase.

   The first digit of the Status-Code defines the class of response. The
   last two digits do not have any categorization role.  There are 5
   values for the first digit:

        o 1xx: Informational - Request received, continuing process

        o 2xx: Success - The action was successfully received,
          understood, and accepted

        o 3rr: Redirection - Further action needs to be taken in order
          to complete the request

        o 4xx: Client Error - The request contains bad syntax or cannot
          be fulfilled

        o 5xx: Server Error - The server failed to fulfill an apparently
          valid request

   The individual values of the numeric status codes defined for
   RTSP/2.0, and an example set of corresponding Reason-Phrases, are
   presented in table 4. The reason phrases listed here are only
   recommended; they may be replaced by local equivalents without
   affecting the protocol. Note that RTSP adopts most HTTP/1.1 [3]
   status codes and adds RTSP-specific status codes starting at x50 to
   avoid conflicts with newly defined HTTP status codes.

   RTSP status codes are extensible. RTSP applications are not required
   to understand the meaning of all registered status codes, though such
   understanding is obviously desirable. However, applications MUST
   understand the class of any status code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 status code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if an
   unrecognized status code of 431 is received by the client, it can
   safely assume that there was something wrong with its request and
   treat the response as if it had received a 400 status code. In such
   cases, user agents SHOULD present to the user the entity returned
   with the response, since that entity is likely to include human-
   readable information which will explain the unusual status.


7.2 Response Header Fields

   The response-header fields allow the request recipient to pass
   additional information about the response which cannot be placed in



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   the Status-Line. These header fields give information about the
   server and about further access to the resource identified by the
   Request-URI. All headers currently classified as response headers are
   listed in table 5.


                Header                  Defined in Section
                __________________________________________
                Accept-Credentials      Section 14.2
                Accept-Ranges           Section 14.5
                Connection-Credentials  Section 14.12
                ETag                    Section 14.21
                Location                Section 14.28
                Proxy-Authenticate      Section 14.29
                Public                  Section 14.33
                Range                   Section 14.34
                Retry-After             Section 14.36
                RTP-Info                Section 14.38
                Scale                   Section 14.39
                Session                 Section 14.42
                Server                  Section 14.41
                Speed                   Section 14.40
                Transport               Section 14.45
                Unsupported             Section 14.46
                Vary                    Section 14.48
                WWW-Authenticate        Section 14.50


   Table 5: The RTSP response headers


   Response-header field names can be extended reliably only in
   combination with a change in the protocol version. However the usage
   of feature-tags in the request allows the responding party to learn
   the capability of the receiver of the response. New or experimental
   header fields MAY be given the semantics of response-header fields if
   all parties in the communication recognize them to be response-header
   fields. Unrecognized header fields in responses are treated as
   entity-header fields.

8 Entity

   Request and Response messages MAY transfer an entity if not otherwise
   restricted by the request method or response status code.  An entity
   consists of entity-header fields and an entity-body, although some
   responses will only include the entity-headers.

   The SET_PARAMETER and GET_PARAMETER request and response, and
   DESCRIBE response MAY have an entity. All 4xx and 5xx responses MAY


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        Code  Reason                               Method
        __________________________________________________________
        100   Continue                             all

        __________________________________________________________
        200   OK                                   all
        __________________________________________________________
        300   Multiple Choices                     all
        301   Moved Permanently                    all
        302   Found                                all
        303   See Other                            all
        305   Use Proxy                            all

        __________________________________________________________
        400   Bad Request                          all
        401   Unauthorized                         all
        402   Payment Required                     all
        403   Forbidden                            all
        404   Not Found                            all
        405   Method Not Allowed                   all
        406   Not Acceptable                       all
        407   Proxy Authentication Required        all
        408   Request Timeout                      all
        410   Gone                                 all
        411   Length Required                      all
        412   Precondition Failed                  DESCRIBE, SETUP
        413   Request Entity Too Large             all
        414   Request-URI Too Long                 all
        415   Unsupported Media Type               all
        451   Parameter Not Understood             SET_PARAMETER
        452   reserved                             n/a
        453   Not Enough Bandwidth                 SETUP
        454   Session Not Found                    all
        455   Method Not Valid In This State       all
        456   Header Field Not Valid               all
        457   Invalid Range                        PLAY, PAUSE
        458   Parameter Is Read-Only               SET_PARAMETER
        459   Aggregate Operation Not Allowed      all
        460   Only Aggregate Operation Allowed     all
        461   Unsupported Transport                all
        462   Destination Unreachable              all
        463   Destination Prohibited               SETUP
        464   Data Transport Not Ready Yet         PLAY
        470   Connection Authorization Required    all
        471   Connection Credentials not accepted  all
        __________________________________________________________
        500   Internal Server Error                all
        501   Not Implemented                      all
        502   Bad Gateway                          all
        503   Service Unavailable                  all
        504   Gateway Timeout                      all
        505   RTSP Version Not Supported           all


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   Table 4: Status codes and their usage with RTSP methods

   also have an entity.

   In this section, both sender and recipient refer to either the client
   or the server, depending on who sends and who receives the entity.

8.1 Entity Header Fields

   Entity-header fields define meta-information about the entity-body
   or, if no body is present, about the resource identified by the
   request. The entity header fields are listed in table 8.1.


                   Header            Defined in Section
                   ____________________________________
                   Allow             Section 14.6
                   Content-Base      Section 14.13
                   Content-Encoding  Section 14.14
                   Content-Language  Section 14.15
                   Content-Length    Section 14.16
                   Content-Location  Section 14.17
                   Content-Type      Section 14.18
                   Expires           Section 14.22
                   Last-Modified     Section 14.27


   Table 6: The RTSP entity headers


   The extension-header mechanism allows additional entity-header fields
   to be defined without changing the protocol, but these fields cannot
   be assumed to be recognizable by the recipient. Unrecognized header
   fields SHOULD be ignored by the recipient and forwarded by proxies.

8.2 Entity Body

   See [H7.2] with the addition that an RTSP message with an entity body
   MUST include the Content-Type and Content-Length headers.

9 Connections

   RTSP requests can be transmitted using the two different connection
   scenarios listed below:

        o persistent - a transport connection is used for several
          request/response transactions;




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        o transient - a transport connection is used for a single
          request/response transaction.

   RFC 2326 attempted to specify an optional mechanism for transmitting
   RTSP messages in connectionless mode over a transport protocol such
   as UDP. However, it was not specified in sufficient detail to allow
   for interoperable implementations. In an attempt to reduce complexity
   and scope, and due to lack of interest, RTSP 2.0 does not attempt to
   define a mechanism for supporting RTSP over UDP or other
   connectionless transport protocols. A side-effect of this is that
   RTSP requests SHALL NOT be sent to multicast groups since no
   connection can be established with a specific receiver in multicast
   environments.

   Certain RTSP headers, such as the CSeq header (Section 14.19), which
   may appear to be relevant only to connectionless transport scenarios
   are still retained and must be implemented according to the
   specification. In the case of CSeq, it is quite useful for matching
   responses to requests if the requests are pipelined (see section
   9.2). It is also useful in proxies for keeping track of the different
   requests when aggregating several client requests on a single TCP
   connection.

9.1 Reliability and Acknowledgements

   When RTSP messages are transmitted using reliable transport
   protocols, they MUST NOT be retransmitted at the RTSP protocol level.
   Instead, the implementation must rely on the underlying transport to
   provide reliability. The RTSP implementation may use any indication
   of reception acknowledgement of the message from the underlying
   transport protocols to optimize the RTSP behavior.


        If both the underlying reliable transport such as TCP and
        the RTSP application retransmit requests, each packet loss
        or message loss may result in two retransmissions. The
        receiver typically cannot take advantage of the
        application-layer retransmission since the transport stack
        will not deliver the application-layer retransmission
        before the first attempt has reached the receiver. If the
        packet loss is caused by congestion, multiple
        retransmissions at different layers will exacerbate the
        congestion.

   Lack of acknowledgement of an RTSP request should be handled within
   the constraints of the connection timeout considerations described
   below (Section 9.4).




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9.2 Using Connections

   A TCP transport can be used for both persistent connections (for
   several message exchanges) and transient connections (for a single
   message exchange). Implementations of this specification MUST support
   RTSP over TCP. The scheme of the RTSP URI (Section 3.2) indicates the
   default port that the server will listen on.

   A server MUST handle both persistent and transient connections.


        Transient connections facilitate mechanisms for fault
        tolerance. They also allow for application layer mobility.
        A server and client pair that support transient connections
        can survive the loss of a TCP connection; e.g., due to a
        NAT timeout. When the client has discovered that the TCP
        connection has been lost, it can set up a new one when
        there is need to communicate again.

   A persistent connection MAY be used for all transactions between the
   server and client, including messages for multiple RTSP sessions.
   However a persistent connection MAY also be closed after a few
   message exchanges. For example, a client may use a persistent
   connection for the initial SETUP and PLAY message exchanges in a
   session and then close the connection. Later, when the client wishes
   to send a new request, such as a PAUSE for the session, a new
   connection would be opened. This connection may either be transient
   or persistent.

   An RTSP agent SHOULD NOT have more than one connection to the server
   at any given point. If a client or proxy handles multiple RTSP
   sessions on the same server, it SHOULD use only one connection for
   managing those sessions.


        This saves connection resources on the server. It also
        reduces complexity by and enabling the server to maintain
        less state about its sessions and connections.

   Unlike HTTP, RTSP allows a server to send requests to a client.
   However, this can be supported only if a client establishes a
   persistent connection with the server. In cases where a persistent
   connection does not exist between a server and its client, due to the
   lack of a signalling channel the server may be forced to drop an RTSP
   session without notifying the client. An example of such a case is
   when the server desires to send a REDIRECT request for an RTSP
   session to the client but is not able to do so because it cannot
   reach the client.



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        Without a persistent connection between the client and the
        server, the media server has no reliable way of reaching
        the client. Also, this is the only way that requests from a
        server to its client are likely to traverse firewalls.

   In light of the above, it is RECOMMENDED that clients use persistent
   connections whenever possible. A client that supports persistent
   connections MAY "pipeline" its requests (i.e., send multiple requests
   without waiting for each response). A server MUST send its responses
   to those requests in the order that the requests were received.

9.3 Closing Connections

   The client MAY close a connection at any point when no outstanding
   request/response transactions exist for any RTSP session being
   managed through the connection. The server, however, SHOULD NOT close
   a connection until all RTSP sessions being managed through the
   connection have been timed out (Section 14.42). A server SHOULD NOT
   close a connection immediately after responding to a session-level
   TEARDOWN request for the last RTSP session being controlled through
   the connection. Instead, it should wait for a reasonable amount of
   time for the client to receive the TEARDOWN response, take
   appropriate action, and initiate the connection closing. The server
   SHOULD wait at least 10 seconds after sending the TEARDOWN response
   before closing the connection.


        This is to ensure that the client has time to issue a SETUP
        for a new session on the existing connection after having
        torn the last one down. 10 seconds should give the client
        ample opportunity get its message to the server.

   A server SHOULD NOT close the connection directly as a result of
   responding to a request with an error code.


        Certain error responses such as "460 Only Aggregate
        Operation Allowed" (Section 13.4.12) are used for
        negotiating capabilities of a server with respect to
        content or other factors. In such cases, it is inefficient
        for the server to close a connection on an error response.
        Also, such behavior would prevent implementation of
        advanced/special types of requests or result in extra
        overhead for the client when testing for new features. On
        the flip side, keeping connections open after sending an
        error response poses a Denial of Service security risk
        (Section 20).




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   If a server initiates a connection close while the client is
   attempting to send a new request, the client will have to close its
   current connection, establish a new connection and send its request
   over the new connection.

   An RTSP message should not be terminated by closing the connection.
   Such a message MAY be considered to be incomplete by the receiver and
   discarded. An RTSP message is properly terminated as defined in
   Section 4.

9.4 Timing Out Connections and RTSP Messages

   Receivers of a request (responder) SHOULD respond to requests in a
   timely manner even when a reliable transport such as TCP is used.
   Similarly, the sender of a request (requestor) SHOULD wait for a
   sufficient time for a response before concluding that the responder
   will not be acting upon its request.

   A responder SHOULD respond to all requests within 5 seconds. If the
   responder recognizes that processing of a request will take longer
   than 5 seconds, it SHOULD send a 100 (Continue) response as soon as
   possible. It SHOULD continue sending a 100 response every 5 seconds
   thereafter until it is ready to send the final response to the
   requestor. After sending a 100 response, the receiver MUST send a
   final response indicating the success or failure of the request.

   A requestor SHOULD wait at least 10 seconds for a response before
   concluding that the responder will not be responding to its request.
   After receiving a 100 response, the requestor SHOULD continue waiting
   for further responses. If more than 10 seconds elapses without
   receiving any response, the requestor MAY assume that the responder
   is unresponsive and abort the connection.

   A requestor SHOULD wait longer than 10 seconds for a response if it
   is experiencing significant transport delays on its connection to the
   responder. The requestor is capable of determining the RTT of the
   request/response cycle using the Timestamp header (section 14.44) in
   any RTSP request.

9.5 Use of IPv6

   Explicit IPv6 support was not present in RTSP 1.0 (RFC 2326). RTSP
   2.0 has been updated for explicit IPv6 support. Implementations of
   RTSP 2.0 MUST understand literal IPv6 addresses in URIs and headers.

10 Capability Handling

   This section describes the capability handling mechanism available in



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   RTSP which allows RTSP to be extended. Extensions to this version of
   the protocol are basically done in two ways. First, new headers can
   be added. Secondly, new methods can be added. The capability handling
   mechanism is designed to handle both cases.

   When a method is added, the involved parties can use the OPTIONS
   method to discover wether it is supported. This is done by issuing a
   OPTIONS request to the other party. Depending on the URI it will
   either apply in regards to a certain media resource, the whole server
   in general, or simply the next hop. The OPTIONS response MUST contain
   a Public header which declares all methods supported for the
   indicated resource.

   It is not necessary to use OPTIONS to discover support of a method,
   the client could simply try the method. If the receiver of the
   request does not support the method it will respond with an error
   code indicating the the method is either not implemented (501) or
   does not apply for the resource (405). The choice between the two
   discovery methods depends on the requirements of the service.

   Feature-Tags are defined to handle functionality additions that are
   not new methods. Each feature-tag represents a certain block of
   functionality. The amount of functionality that a feature-tag
   represents can vary significantly. A feature-tag can for example
   represent the functionality a single RTSP header provides. Another
   feature-tag can represent much more functionality, such as the
   "play.basic" feature-tag which represents the minimal playback
   implementation.

   Feature-tags are used to determine wether the client, server or proxy
   supports the functionality that is necessary to achieve the desired
   service. To determine support of a feature-tag, several different
   headers can be used, each explained below:

        Supported: The supported header is used to determine the
             complete set of functionality that both client and server
             have. The intended usage is to determine before one needs
             to use a functionality that it is supported. It can be used
             in any method, however OPTIONS is the most suitable one as
             it at the same time determines all methods that are
             implemented. When sending a request the requestor declares
             all its capabilities by including all supported feature-
             tags. This results in that the receiver learns the
             requestors feature support. The receiver then includes its
             set of features in the response.

        Proxy-Supported: The Proxy-Supported header is used similar to
             the Supported header, but instead of giving the supported



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             functionality of the client or server it provides both the
             requestor and the responder a view of what functionality
             the proxy chain between the two supports.  Proxies are
             required to add this header whenever the Supported header
             is present, but proxies may independently of the requestor
             add it.

        Require: The Require header can be included in any request where
             the end-point, i.e. the client or server, is required to
             understand the feature to correctly perform the request.
             This can, for example, be a SETUP request where the server
             is required to understand a certain parameter to be able to
             set up the media delivery correctly. Ignoring this
             parameter would not have the desired effect and is not
             acceptable. Therefore the end-point receiving a request
             containing a Require MUST negatively acknowledge any
             feature that it does not understand and not perform the
             request. The response in cases where features are not
             supported are 551 (Option Not Supported).  Also the
             features that are not supported are given in the
             Unsupported header in the response.

        Proxy-Require: This method has the same purpose and workings as
             Require except that it only applies to proxies and not the
             end-point. Features that needs to be supported by both
             proxies and end-point needs to be included in both the
             Require and Proxy-Require header.

        Unsupported: This header is used in a 551 error response, to
             indicate which feature(s) that was not supported.  Such a
             response is only the result of the usage of the Require
             and/or Proxy-Require header where one or more feature where
             not supported. This information allows the requestor to
             make the best of situations as it knows which features are
             not supported.

11 Method Definitions

   The method indicates what is to be performed on the resource
   identified by the Request-URI. The method name is case-sensitive.
   New methods may be defined in the future. Method names SHALL NOT
   start with a $ character (decimal 24) and MUST be a token as defined
   by the ABNF [4] in the syntax chapter 19. The methods are summarized
   in Table 7.



        Note on Table 7: GET_PARAMETER is recommended, but not



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    method         direction       object  Server req.    Client req.
    ___________________________________________________________________
    DESCRIBE       C -> S          P,S     recommended    recommended
    GET_PARAMETER  C -> S, S -> C  P,S     optional       optional
    OPTIONS        C -> S, S -> C  P,S     R=Req, Sd=Opt  Sd=Req, R=Opt
    PAUSE          C -> S          P,S     required       required
    PLAY           C -> S          P,S     required       required
    REDIRECT       S -> C          P,S     optional       required
    SETUP          C -> S          S       required       required
    SET_PARAMETER  C -> S, S -> C  P,S     required       optional
    TEARDOWN       C -> S          P,S     required       required


   Table 7: Overview of RTSP methods, their direction, and what  objects
   (P:  presentation,  S:  stream)  they  operate on. Legend: R=Respond,
   Sd=Send, Opt: Optional, Req: Required

        required. For example, a fully functional server can be
        built to deliver media without any parameters.
        SET_PARAMETER is required however due to its usage for
        keep-alive.  PAUSE is now required due to that it is the
        only way of getting out of the state machines play state
        without terminating the whole session.

   If an RTSP agent does not support a particular method, it MUST return
   501 (Not Implemented) and the requesting RTSP agent, in turn, SHOULD
   NOT try this method again for the given agent / resource combination.

11.1 OPTIONS

   The semantics of the RTSP OPTIONS method is equivalent to that of the
   HTTP OPTIONS method described in [H9.2]. In RTSP however, OPTIONS is
   bi-directional, in that a client can request it to a server and vice
   versa. A client MUST implement the capability to send an OPTIONS
   request and a server or a proxy MUST implement the capability to
   respond to an OPTIONS request. The client, server or proxy MAY also
   implement the converse of their required capability.

   An OPTIONS request may be issued at any time. Such a request does not
   modify the session state. However, it may prolong the session
   lifespan (see below). The URI in an OPTIONS request determines the
   scope of the request and the corresponding response. If the Request-
   URI refers to a specific media resource on a given host, the scope is
   limited to the set of methods supported for that media resource by
   the indicated RTSP agent. A Request-URI with only the host address
   limits the scope to the specified RTSP agent's general capabilities
   without regard to any specific media. If the Request-URI is an
   asterisk ("*"), the scope is limited to the general capabilities of


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   the next hop (i.e.  the RTSP agent in direct communication with the
   request sender).

   Regardless of scope of the request, the Public header MUST always be
   included in the OPTIONS response listing the methods that are
   supported by the responding RTSP agent. In addition, if the scope of
   the request is limited to a media resource, the Allow header MUST be
   included in the response to enumerate the set of methods that are
   allowed for that resource unless the set of methods completely
   matches the set in the Public header. If the given resource is not
   available, the RTSP agent SHOULD return an appropriate response code
   such as 3rr or 4xx. The Supported header MAY be included in the
   request to query the set of features that are supported by the
   responding RTSP agent.

   The OPTIONS method can be used to keep an RTSP session alive.
   However, it is not the preferred means of session keep-alive
   signalling, see section 14.42. An OPTIONS request intended for
   keeping alive an RTSP session MUST include the Session header with
   the associated session ID. Such a request SHOULD also use the media
   or the aggregated control URI as the Request-URI.

   Example:


     C->S:  OPTIONS * RTSP/2.0
            CSeq: 1
            User-Agent: PhonyClient/1.2
            Require:
            Proxy-Require: gzipped-messages
            Supported: play.basic

     S->C:  RTSP/2.0 200 OK
            CSeq: 1
            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
            Supported: play.basic, implicit-play, gzipped-messages
            Server: PhonyServer/1.1



   Note that some of the feature-tags in Require and Proxy-Require are
   fictional features.

11.2 DESCRIBE

   The DESCRIBE method is used to retrieve the description of a
   presentation or media object from a server. The Request-URI of the
   DESCRIBE request identifies the media resource of interest. The



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   client MAY include the Accept header in the request to list the
   description formats that it understands. The server SHALL respond
   with a description of the requested resource and return the
   description in the entity of the response. The DESCRIBE reply-
   response pair constitutes the media initialization phase of RTSP.

   Example:


     C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/2.0
           CSeq: 312
           User-Agent: PhonyClient 1.2
           Accept: application/sdp, application/example

     S->C: RTSP/2.0 200 OK
           CSeq: 312
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.1
           Content-Type: application/sdp
           Content-Length: 367

           v=0
           o=mhandley 2890844526 2890842807 IN IP4 192.0.2.46
           s=SDP Seminar
           i=A Seminar on the session description protocol
           u=http://www.example.com/lectures/sdp.ps
           e=seminar@example.com (Seminar Management)
           c=IN IP4 224.2.17.12/127
           t=2873397496 2873404696
           a=recvonly
           m=audio 3456 RTP/AVP 0
           m=video 2232 RTP/AVP 31
           m=application 32416 UDP WB
           a=orient:portrait



   The DESCRIBE response SHOULD contain all media initialization
   information for the resource(s) that it describes. Servers SHOULD NOT
   use the DESCRIBE response as a means of media indirection by having
   the description point at another server, instead usage of 3rr
   responses are recommended.


        By forcing a DESCRIBE response to contain all media
        initialization for the set of streams that it describes,
        and discouraging the use of DESCRIBE for media indirection,
        any looping problems can be avoided that might have



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        resulted from other approaches.

   Media initialization is a requirement for any RTSP-based system, but
   the RTSP specification does not dictate that this is required to be
   done via the DESCRIBE method. There are three ways that an RTSP
   client may receive initialization information:

        o via an RTSP DESCRIBE request

        o via some other protocol (HTTP, email attachment, etc.)

        o via some form of a user interface

   If a client obtains a valid description from an alternate source, the
   client MAY use this description for initialization purposes without
   issuing a DESCRIBE request for the same media.

   It is RECOMMENDED that minimal servers support the DESCRIBE method,
   and highly recommended that minimal clients support the ability to
   act as "helper applications" that accept a media initialization file
   from a user interface, and/or other means that are appropriate to the
   operating environment of the clients.

11.3 SETUP

   The SETUP request for an URI specifies the transport mechanism to be
   used for the streamed media. The SETUP method may be used in three
   different cases; Create an RTSP session, add a media to a session,
   and change the transport parameters of already set up media stream.
   When in PLAY state, using SETUP to create or add media to a session
   when in PLAY state is unspecified. Otherwise SETUP can be used in all
   three states; INIT, and READY, for both purposes and in PLAY to
   change the transport parameters.

   The Transport header, see section 14.45, specifies the transport
   parameters acceptable to the client for data transmission; the
   response will contain the transport parameters selected by the
   server. This allows the client to enumerate in priority order the
   transport mechanisms and parameters acceptable to it, while the
   server can select the most appropriate. It is expected that the
   session description format used will enable the client to select a
   limited number possible configurations that are offered to the server
   to choose from. All transport related parameters shall be included in
   the Transport header, the use of other headers for this purpose is
   discouraged due to middle boxes such as firewalls, or NATs.

   For the benefit of any intervening firewalls, a client SHALL indicate
   the known transport parameters, even if it has no influence over



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   these parameters, for example, where the server advertises a fixed
   multicast address as destination.


        Since SETUP includes all transport initialization
        information, firewalls and other intermediate network
        devices (which need this information) are spared the more
        arduous task of parsing the DESCRIBE response, which has
        been reserved for media initialization.

   The client SHALL include the Accept-Ranges header in the request
   indicating all supported unit formats in the Range header.  This
   allows the server to know which format it may use in future session
   related responses, such as PLAY response without any range in the
   request. If the client does not support a time format necessary for
   the presentation the server SHALL respond using 456 (Header Field Not
   Valid for Resource) and include the Accept-Ranges header with the
   range unit formats supported for the resource.

   In a SETUP response the server SHALL include the Accept-Ranges header
   (see section  14.5 to indicate which time formats that are acceptable
   to use for this media resource.


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
           CSeq: 302
           Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
                      RTP/AVP/TCP;unicast;interleaved=0-1

     S->C: RTSP/2.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.1
           Session: 47112344;timeout=60
           Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589";
                      src_addr="192.0.2.241:6256"/"192.0.2.241:6257";
                      ssrc=2A3F93ED
           Accept-Ranges: NPT



   In the above example the client wants to create an RTSP session
   containing the media resource "rtsp://example.com/foo/bar/baz.rm".
   The transport parameters acceptable to the client is either
   RTP/AVP/UDP (UDP per default) to be received on client port 4588 and
   4589 or RTP/AVP interleaved on the RTSP control channel. The server
   selects the RTP/AVP/UDP transport and adds the ports it will send and
   received RTP and RTCP from, and the RTP SSRC that will be used by the



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   server.

   The server MUST generate a session identifier in response to a
   successful SETUP request, unless a SETUP request to a server includes
   a session identifier, in which case the server MUST bundle this setup
   request into the existing session (aggregated session) or return
   error 459 (Aggregate Operation Not Allowed) (see Section 13.4.11). An
   Aggregate control URI MUST be used to control an aggregated session.
   This URI MUST be different from the stream control URIs of the
   individual media streams included in the aggregate. The Aggregate
   control URI is to be specified by the session description if the
   server supports aggregated control and aggregated control is desired
   for the session. However even if aggregated control is offered the
   client MAY chose to not set up the session in aggregated control. If
   an Aggregate control URI is not specified in the session description,
   it is normally an indication that non-aggregated control should be
   used. The SETUP of media streams in an aggregate which has not been
   given an aggregated control URI is unspecified.


        While the session ID sometimes has enough information for
        aggregate control of a session, the Aggregate control URI
        is still important for some methods such as SET_PARAMETER
        where the control URI enables the resource in question to
        be easily identified. The Aggregate control URI is also
        useful for proxies, enabling them to route the request to
        the appropriate server, and for logging, where it is useful
        to note the actual resource that a request was operating
        on.

   A session will exist until it is either removed by a TEARDOWN request
   or is timed-out by the server. The server MAY remove a session that
   has not demonstrated liveness signs from the client(s) within a
   certain timeout period. The default timeout value is 60 seconds; the
   server MAY set this to a different value and indicate so in the
   timeout field of the Session header in the SETUP response. For
   further discussion see section 14.42. Signs of liveness for an RTSP
   session are:

        o Any RTSP request from a client(s) which includes a Session
          header with that session's ID.

        o If RTP is used as a transport for the underlying media
          streams, an RTCP sender or receiver report from the client(s)
          for any of the media streams in that RTSP session. RTCP Sender
          Reports may for example be received in sessions where the
          server is invited into a conference session and is as valid
          for keep-alive.



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   If a SETUP request on a session fails for any reason, the session
   state, as well as transport and other parameters for associated
   streams SHALL remain unchanged from their values as if the SETUP
   request had never been received by the server.

11.3.1 Changing Transport Parameters

   A client MAY issue a SETUP request for a stream that is already set
   up or playing in the session to change transport parameters, which a
   server MAY allow. If it does not allow changing of parameters, it
   MUST respond with error 455 (Method Not Valid In This State). Reasons
   to support changing transport parameters, is to allow for application
   layer mobility and flexibility to utilize the best available
   transport as it becomes available. If a client receives a 455 when
   trying to change transport parameters while the server is in play
   state, it MAY try to put the server in ready state using PAUSE,
   before issuing the SETUP request again. If also that fails the
   changing of transport parameters will require that the client
   performs a TEARDOWN of the affected media and then setting it up
   again. In aggregated session avoiding tearing down all the media at
   the same time will avoid the creation of a new session.

   All transport parameters MAY be changed. However the primary usage
   expected is to either change transport protocol completely, like
   switching from Interleaved TCP mode to UDP or vise versa or change
   delivery address.

   In a SETUP response for a request to change the transport parameters
   while in Play state, the server SHALL include the Range to indicate
   from what point the new transport parameters are used. Further, if
   RTP is used for delivery, the server SHALL also include the RTP-Info
   header to indicate from what timestamp and RTP sequence number the
   change has taken place. If both RTP-Info and Range is included in the
   response the "rtp_time" parameter and range MUST be for the
   corresponding time, i.e. be used in the same way as for PLAY to
   ensure the correct synchronization information is available.

   If the transport parameters change while in PLAY state results in a
   change of synchronization related information, for example changing
   RTP SSRC, the server MUST provide in the SETUP response the necessary
   synchronization information. However the server is RECOMMENDED to
   avoid changing the synchronization information if possible.

11.4 PLAY

   The PLAY method tells the server to start sending data via the
   mechanism specified in SETUP. PLAY requests are valid when the
   session is in READY or PLAY states. A PLAY request MUST include a



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   Session header to indicate which session the request applies to.

   For aggregated sessions where the initial SETUP request (creating a
   session) is followed by one or more additional SETUP request, a PLAY
   request MAY be pipelined after those additional SETUP requests
   without awaiting their responses. This can procedure can reduce the
   delay from start of session establishment until media play-out has
   started with one round trip time. However an client needs to be aware
   that using this procedure will result in the playout of the server
   state established at the time of processing the PLAY, i.e. after the
   processing of all the requests prior to the PLAY request in the
   pipeline. This may not be the intended one due to failure of any of
   the prior requests. However a client easily determine this based on
   the responses from those requests. In case of failure the client can
   halt the media playout using PAUSE and try to establish the intended
   state again before issuing another PLAY request.

   In an aggregated session the PLAY request MUST contain an aggregated
   control URI. A server SHALL responde with error 460 (Only Aggregate
   Operation Allowed) if the client PLAY Request-URI is for one of the
   media. The media in an aggregate SHALL be played in sync. If a client
   want individual control of the media it needs to use separate RTSP
   sessions for each media.

   The PLAY request SHALL position the normal play time to the beginning
   of the range specified by the Range header and delivers stream data
   until the end of the range if given, else to the end of the media is
   reached. To allow for precise composition multiple ranges MAY be
   specified in one PLAY Request. The range values are valid if all
   given ranges are part of any media within the aggregate. If a given
   range value points outside of the media, the response SHALL be the
   457 (Invalid Range) error code.

   The below example will first play seconds 10 through 15, then,
   immediately following, seconds 20 to 25, and finally seconds 30
   through the end.


     C->S: PLAY rtsp://audio.example.com/audio RTSP/2.0
           CSeq: 835
           Session: 12345678
           Range: npt=10-15, npt=20-25, npt=30-



   See the description of the PAUSE request for further examples.

   A PLAY request without a Range header is legal. It SHALL start



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   playing a stream from the beginning (npt=0-) unless the stream has
   been paused or is currently playing. If a stream has been paused via
   PAUSE, stream delivery resumes at the pause point.  If a stream is
   currently playing, the new PLAY begins at the current stream
   position. The stream SHALL play until the end of the media.

   The Range header MUST NOT contain a time parameter. The usage of time
   in PLAY method has been deprecated. If a request with time parameter
   is received the server SHOULD respond with a 457 (Invalid Range) to
   indicate that the time parameter is not supported.

   Server MUST include a "Range" header in any PLAY response. The
   response MUST use the same format as the request's range header
   contained. If no Range header was in the request, the NPT time format
   SHOULD be used unless the client showed support for an other format
   more appropriate. Also for a session with live media streams the
   Range header MUST indicate a valid time. It is RECOMMENDED that
   normal play time is used, either the "now" indicator, for example
   "npt=now-", or the time since session start as an open interval, e.g.
   "npt=96.23-". An absolute time value (clock) for the corresponding
   time MAY be given, i.e.  "clock=20030213T143205Z-". The UTC clock
   format SHOULD only be used if client has shown support for it.

   For an on-demand stream, the server MUST reply with the actual range
   that will be played back, i.e. for which duration any media (having
   content at this time) is delivered. This may differ from the
   requested range if alignment of the requested range to valid frame
   boundaries is required for the media source. Note that some media
   streams in an aggregate may need to be delivered from even earlier
   points. Also, some media format have a very long duration per
   individual data unit, therefore it might be necessary for the client
   to parse the data unit, and select where to start.


   Example: Single audio stream (MIDI)

   C->S: PLAY rtsp://example.com/audio RTSP/2.0
         CSeq: 836
         Session: 12345678
         Range: npt=7.05-

   S->C: RTSP/2.0 200 OK
         CSeq: 836
         Date: 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer 1.0
         Range: npt=3.52-
         RTP-Info:url="rtsp://example.com/audio"
            ssrc=0D12F123:seq=14783;rtptime=2345962545



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   S->C: RTP Packet TS=2345962545 => NPT=3.52
         Duration: 4.15 seconds



   In this example the client receives the first media packet that
   stretches all the way up and past the requested playtime. Thus, it is
   the client's decision if to render to the user the time between 3.52
   and 7.05, or to skip it. In most cases it is probably most suitable
   to not render that time period.

   For live media sources it might be impossible to specify from which
   point in time all media streams carrying active content can actually
   be delivered. Therefore a server MAY specify a start time (or now-)
   in the range header, for which not all media will be available from.

   If no range is specified in the request, the start position SHALL
   still be returned in the reply. If the medias that are part of an
   aggregate has different lengths, the PLAY request SHALL be performed
   as long as the given range is valid for any media, for example the
   longest media. Media will be sent whenever it is available for the
   given play-out point.

   A PLAY response MAY include a header(s) carrying synchronization
   information. As the information necessary is dependent on the media
   transport format, further rules specifying the header and its usage
   is needed. For RTP the RTP-Info header is specified, see section
   14.38.

   After playing the desired range, the presentation does NOT transition
   to the READY state, media delivery simply stops. A PAUSE request MUST
   be issued before the stream enters the READY state. A PLAY request
   while the stream is still in the PLAYING state is legal, and can be
   issued without an intervening PAUSE request. Such a request SHALL
   replace the current PLAY action with the new one requested, i.e.
   being handle the same as the request was received in ready state. In
   the case the first time range in Range header has a open start time
   (-endtime), the server SHALL continue to play from where it currently
   was until the specified end point. This is useful to change ongoing
   playback to play another sequence, or end at another point than in
   the previous request.

   A client desiring to play the media from the beginning MUST send a
   PLAY request with a Range header pointing at the beginning, e.g.
   npt=0-. If a PLAY request is received without a Range header when
   media delivery has stopped at the end, the server SHOULD respond with
   a 457 "Invalid Range" error response. In that response the current
   pause point in a Range header SHALL be included.



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   The following example plays the whole presentation starting at SMPTE
   time code 0:10:20 until the end of the clip. Note: The RTP-Info
   headers has been broken into several lines to fit the page.


   C->S: PLAY rtsp://audio.example.com/twister.en RTSP/2.0
         CSeq: 833
         Session: 12345678
         Range: smpte=0:10:20-

   S->C: RTSP/2.0 200 OK
         CSeq: 833
         Date: 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer 1.0
         Range: smpte=0:10:22-0:15:45
         RTP-Info:url="rtsp://example.com/twister.en"
            ssrc=0D12F123:seq=14783;rtptime=2345962545



   For playing back a recording of a live presentation, it may be
   desirable to use clock units:


     C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/2.0
           CSeq: 835
           Session: 12345678
           Range: clock=19961108T142300Z-19961108T143520Z

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:06 GMT
           Server:PhonyServer 1.0
           Range: clock=19961108T142300Z-19961108T143520Z
           RTP-Info:url="rtsp://example.com/meeting.en"
              ssrc=0D12F123:seq=53745;rtptime=484589019




   All range specifiers in this specification allow for ranges with
   unspecified begin times (e.g. "npt=-30"). When used in a PLAY
   request, the server treats this as a request to start/resume playback
   from the current pause point, ending at the end time specified in the
   Range header. If the pause point is located later than the given end
   value, a 457 (Invalid Range) response SHALL be given.

   The possibility to replace a current PLAY request with a new one



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   replaces two RTSP 1.0 functions:

        o The queued play functionality described in RFC 2326 [25] is
          removed and multiple ranges can be used to achieve a similar
          functionality.

        o The use of PLAY for keep-alive signaling, i.e. PLAY request
          without a range header in PLAY state, has also been
          deprecated. Instead a client can use, SET_PARAMETER
          (recommended) or OPTIONS (allowed) for keep alive.

   An example of using PLAY request to change the behavior, if a server
   has received requests to play ranges 10 to 15 and then 13 to 20 (that
   is, overlapping ranges), a PLAY request 4 seconds after the first
   would take effect while the server plays the first range.  Thus
   changing the behavior to continue to play to 25 seconds, i.e.  the
   played range equal play with range: npt=10-25. If the second PLAY
   request would arrive after the second range in the first range was
   playing, then the equivalent request would be play with
   range:npt=10-15,npt=13-25.


     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-15, npt=13-20

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Range: npt=10-15, npt=13-20
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934207921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=789DAF12:seq=57654;rtptime=2792482193
           Session: 12345678

     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 835
           Session: 12345678
           Range: npt=-25

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=14-15, npt=13-25



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           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934239921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=789DAF12:seq=57654;rtptime=2792842193
           Session: 12345678



11.5 PAUSE

   The PAUSE request causes the stream delivery to immediately be
   interrupted (halted). A PAUSE request MUST be done with the
   aggregated control URI for aggregated sessions, resulting in all
   media being halted, or the media URI for non-aggregated sessions.
   Any attempt to do muting of a single media with an PAUSE request in
   an aggregated session SHALL be responded with error 460 (Only
   Aggregate Operation Allowed). After resuming playback,
   synchronization of the tracks MUST be maintained. Any server
   resources are kept, though servers MAY close the session and free
   resources after being paused for the duration specified with the
   timeout parameter of the Session header in the SETUP message.

   Example:

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-



   The PAUSE request causes stream delivery to be interrupted
   immediately on receipt of the message and the pause point is set to
   the current point in the presentation. That pause point in the media
   stream needs to be maintained. A subsequent PLAY request without
   Range header SHALL resume from the pause point and play until media
   end.

   The pause point after any PAUSE request SHALL be returned to the
   client by adding a Range header with what remains unplayed of the
   PLAY request's ranges, i.e. including all the remaining ranges part
   of multiple range specification. If one desires to resume playing a
   ranged request, one simply includes the Range header from the PAUSE
   response.



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     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-30

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Range: npt=10-30
           RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
                   ssrc=0D12F123:seq=5712;rtptime=934207921,
                   url="rtsp://example.com/fizzle/videotrack"
                   ssrc=4FAD8726:seq=57654;rtptime=2792482193
           Session: 12345678

   after 11 seconds, i.e. at 21 seconds into the presentation:
     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 835
           Session: 12345678

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:09 GMT
           Server: PhonyServer 1.0
           Range: npt=21-30
           Session: 12345678



   If a client issues a PAUSE request and the server acknowledges and
   enters the READY state, the proper server response, if the player
   issues another PAUSE, is still 200 OK. The 200 OK response MUST
   include the Range header with the current pause point. See examples
   below:

   Examples:

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/2.0 200 OK
           CSeq: 834
           Session: 12345678
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-98.36




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     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 835
           Session: 12345678

     S->C: RTSP/2.0 200 OK
           CSeq: 835
           Session: 12345678
           Date: 23 Jan 1997 15:35:07 GMT
           Range: npt=45.76-98.36



11.6 TEARDOWN

   The TEARDOWN client to server request stops the stream delivery for
   the given URI, freeing the resources associated with it. A TEARDOWN
   request MAY be performed on either an aggregated or a media control
   URI. However some restrictions apply depending on the current state.
   The TEARDOWN request SHALL contain a Session header indicating what
   session the request applies to.

   A TEARDOWN using the aggregated control URI or the media URI in a
   session under non-aggregated control (single media session) MAY be
   done in any state (Ready, and Play). A successful request SHALL
   result in that media delivery is immediately halted and the session
   state is destroyed. This SHALL be indicated through the lack of a
   Session header in the response.

   A TEARDOWN using a media URI in an aggregated session MAY only be
   done in Ready state. Such a request only removes the indicated media
   stream and associated resources from the session. This may result in
   that a session returns to non-aggregated control, due to that it only
   contains a single media after the requests completion.  A session
   that will exist after the processing of the TEARDOWN request SHALL in
   the response to that TEARDOWN request contain a Session header. Thus
   the presence of the Session header indicates to the receiver of the
   response if the session is still existing or has been removed.


   Example:

     C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 892
           Session: 12345678

     S->C: RTSP/2.0 200 OK
           CSeq: 892
           Server: PhonyServer 1.0



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11.7 GET_PARAMETER

   The GET_PARAMETER request retrieves the value of a parameter or
   parameters for a presentation or stream specified in the URI. If the
   Session header is present in a request, the value of a parameter MUST
   be retrieved in the specified session context.  The content of the
   reply and response is left to the implementation.

   The method MAY also be used without a body (entity). If the this
   request is successful, i.e. a 200 OK response is received, then the
   keep-alive timer has been updated. Any non-required header present in
   such a request may or may not been processed. To allow a client to
   determine if any such header has been processed, it is necessary to
   use a feature-tag and the Require header. Due to this reason it is
   RECOMMENDED that any parameters to be retrieved are sent in the body,
   rather than using any header.

   Example:


     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 431
           Content-Type: text/parameters
           Session: 12345678
           Content-Length: 26

           packets_received
           jitter

     C->S: RTSP/2.0 200 OK
           CSeq: 431
           Content-Length: 38
           Content-Type: text/parameters

           packets_received: 10
           jitter: 0.3838




        The "text/parameters" section is only an example type for a
        body carrying parameters.

11.8 SET_PARAMETER

   This method requests to set the value of a parameter or a set of
   parameters for a presentation or stream specified by the URI. The
   method MAY also be used without a body (entity). It is the



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   RECOMMENDED method to use in request sent for the sole purpose of
   updating the keep-alive timer. If this request is successful, i.e. a
   200 OK response is received, then the keep-alive timer has been
   updated. Any non-required header present in such a request may or may
   not been processed. To allow a client to determine if any such header
   has been processed, it is necessary to use a feature tag and the
   Require header. Due to this reason it is RECOMMENDED that any
   parameters are sent in the body, rather than using any header.

   A request is RECOMMENDED to only contain a single parameter to allow
   the client to determine why a particular request failed. If the
   request contains several parameters, the server MUST only act on the
   request if all of the parameters can be set successfully. A server
   MUST allow a parameter to be set repeatedly to the same value, but it
   MAY disallow changing parameter values.  If the receiver of the
   request does not understand or cannot locate a parameter, error 451
   (Parameter Not Understood) SHALL be used. In the case a parameter is
   not allowed to change, the error code is 458 (Parameter Is Read-
   Only). The response body SHOULD contain only the parameters that have
   errors. Otherwise no body SHALL be returned.

   Note: transport parameters for the media stream MUST only be set with
   the SETUP command.

        Restricting setting transport parameters to SETUP is for
        the benefit of firewalls.


        The parameters are split in a fine-grained fashion so that
        there can be more meaningful error indications. However, it
        may make sense to allow the setting of several parameters
        if an atomic setting is desirable. Imagine device control
        where the client does not want the camera to pan unless it
        can also tilt to the right angle at the same time.

   Example:


     C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 421
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff

     S->C: RTSP/2.0 451 Parameter Not Understood
           CSeq: 421
           Content-length: 10



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           Content-type: text/parameters

           barparam: barstuff




        The "text/parameters" section is only an example type for
        parameters. This method is intentionally loosely defined
        with the intention that the reply content and response
        content will be defined by the one desiring to use this
        mechanism.

11.9 REDIRECT

   The REDIRECT method is issued by a server to inform a client that it
   required to connect to another server location to access the resource
   indicated by the Request-URI. The presence of the Session header in a
   REDIRECT request indicates the scope of the request, and determines
   the specific semantics of the request.

   A REDIRECT request with a Session header has end-to-end (i.e. server
   to client) scope and applies only to the given session.  Any
   intervening proxies SHOULD NOT disconnect the control channel while
   there are other remaining end-to-end sessions. The OPTIONAL Location
   header, if included in such a request, SHALL contain a complete
   absolute URI pointing to the resource to which the client SHOULD
   reconnect. Specifically, the Location SHALL NOT contain just the host
   and port. A client may receive a REDIRECT request with a Session
   header, if and only if, an end-to-end session has been established.

   A client may receive a REDIRECT request without a Session header at
   any time when it has communication or a connection established with a
   server. The scope of such a request is limited to the next-hop (i.e.
   the RTSP agent in direct communication with the server) and applies,
   as well, to the control connection between the next-hop RTSP agent
   and the server.  A REDIRECT request without a Session header
   indicates that all sessions and pending requests being managed via
   the control connection MUST be redirected. The OPTIONAL Location
   header, if included in such a request, SHOULD contain an absolute URI
   with only the host address and the OPTIONAL port number of the server
   to which the RTSP agent SHOULD reconnect. Any intervening proxies
   SHOULD do all of the following in the order listed:

        1.   respond to the REDIRECT request

        2.   disconnect the control channel from the requesting server




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        3.   connect to the server at the given host address

        4.   pass the REDIRECT request to each applicable client
             (typically those clients with an active session or an
             unanswered request)

   Note: The proxy is responsible for accepting REDIRECT responses from
   its clients; these responses MUST NOT be passed on to either the
   original server or the redirected server.

   The lack of a Location header in any REDIRECT request is indicative
   of the server no longer being able to fulfill the current request and
   having no alternatives for the client to continue with its normal
   operation. It is akin to a server initiated TEARDOWN that applies
   both to sessions as well as the general connection associated with
   that client.

   When the Range header is not included in a REDIRECT request, the
   client SHOULD perform the redirection immediately and return a
   response to the server. The server can consider the session as
   terminated and can free any associated state after it receives the
   successful (2xx) response. The server MAY close the signalling
   connection upon receiving the response and the client SHOULD close
   the signalling connection after sending the 2xx response. The
   exception to this is when the client has several sessions on the
   server being managed by the given signalling connection. In this
   case, the client SHOULD close the connection when it has received and
   responded to REDIRECT requests for all the sessions managed by the
   signalling connection.

   If the OPTIONAL Range header is included in a REDIRECT request, it
   indicates when the redirection takes effect.  The range value MUST be
   an open ended single value, e.g. npt=59-, indicating the play out
   time when redirection SHALL occur.  Alternatively, a range with a
   time= parameter indicates the wall clock time by when the redirection
   MUST take place. When the time= parameter is present in the range,
   any range value MUST be ignored even though it MUST be syntactically
   correct. To allow a client to determine that redirect time without
   being time synchronized with the server, the server SHALL include a
   Date header in the request. When the indicated redirect point is
   reached, a client MUST issue a TEARDOWN request and SHOULD close the
   signalling connection after receiving a 2xx response. The normal
   connection considerations apply for the server.


        The differentiation of REDIRECT requests with and without
        range headers is to allow for clear and explicit state
        handling. As the state in the server needs to be kept until



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        the point of redirection, the handling becomes more clear
        if the client is required to TEARDOWN the session at the
        redirect point.

   If the REDIRECT request times out following the rules in Section 9.4
   the server MAY terminate the session or transport connection that
   would be redirected by the request. This is a safeguard against
   misbehaving clients that refuses to respond to a REDIRECT request.
   That should not provide any benefit.

   After a REDIRECT request has been processed, a client that wants to
   continue to send or receive media for the resource identified by the
   Request-URI will have to establish a new session with the designated
   host. If the URI given in the Location header is a valid resource
   URI, a client SHOULD issue a DESCRIBE request for the URI.


        Note: The media resource indicated by the Location header
        can be identical, slightly different or totally different.
        This is the reason why a new DESCRIBE request SHOULD be
        issued.

   If the Location header contains only a host address, the client MAY
   assume that the media on the new server is identical to the media on
   the old server, i.e. all media configuration information from the old
   session is still valid except for the host address. However the usage
   of conditional SETUP using ETag identifiers are RECOMMENDED to verify
   the assumption.

   This example request redirects traffic for this session to the new
   server at the given absolute time:


     S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 732
           Location: rtsp://s2.example.com:8001
           Range: npt=0- ;time=19960213T143205Z
           Session: uZ3ci0K+Ld-M

     C->S: RTSP/2.0 200 OK
           CSeq: 732



12 Embedded (Interleaved) Binary Data

   In order to fulfill certain requirements on the network side, e.g.
   in conjunction with network address translators that block RTP



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   traffic over UDP, it may be necessary to interleave RTSP messages and
   media stream data. This interleaving should generally be avoided
   unless necessary since it complicates client and server operation and
   imposes additional overhead. Also head of line blocking may cause
   problems. Interleaved binary data SHOULD only be used if RTSP is
   carried over TCP.

   Stream data such as RTP packets is encapsulated by an ASCII dollar
   sign (24 decimal), followed by a one-byte channel identifier,
   followed by the length of the encapsulated binary data as a binary,
   two-byte integer in network byte order. The stream data follows
   immediately afterwards, without a CRLF, but including the upper-layer
   protocol headers. Each $ block SHALL contain exactly one upper-layer
   protocol data unit, e.g., one RTP packet.




       0                   1                   2                   3
       0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      | "$" = 24      | Channel ID    | Length in bytes               |
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
      : Length number of bytes of binary data                         :
      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+





   The channel identifier is defined in the Transport header with the
   interleaved parameter(Section 14.45).

   When the transport choice is RTP, RTCP messages are also interleaved
   by the server over the TCP connection. The usage of RTCP messages is
   indicated by including a range containing a second channel in the
   interleaved parameter of the Transport header, see section 14.45. If
   RTCP is used, packets SHALL be sent on the first available channel
   higher than the RTP channel. The channels are bi-directional and
   therefore RTCP traffic are sent on the second channel in both
   directions.


        RTCP is sometime needed for synchronization when two or
        more streams are interleaved in such a fashion. Also, this
        provides a convenient way to tunnel RTP/RTCP packets
        through the TCP control connection when required by the
        network configuration and transfer them onto UDP when



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        possible.



     C->S: SETUP rtsp://example.com/bar.file RTSP/2.0
           CSeq: 2
           Transport: RTP/AVP/TCP;unicast;interleaved=0-1

     S->C: RTSP/2.0 200 OK
           CSeq: 2
           Date: 05 Jun 1997 18:57:18 GMT
           Transport: RTP/AVP/TCP;unicast;interleaved=5-6
           Session: 12345678

     C->S: PLAY rtsp://example.com/bar.file RTSP/2.0
           CSeq: 3
           Session: 12345678

     S->C: RTSP/2.0 200 OK
           CSeq: 3
           Session: 12345678
           Date: 05 Jun 1997 18:59:15 GMT
           RTP-Info: url="rtsp://example.com/bar.file"
             ssrc=0D12F123:seq=232433;rtptime=972948234

     S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $006{2 byte length}{"length" bytes  RTCP packet}



13 Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused. Status codes
   that have the same meaning are not repeated here. See Table 4 for a
   listing of which status codes may be returned by which requests. All
   error messages, 4xx and 5xx MAY return a body containing further
   information about the error.

13.1 Success 1xx

13.1.1 100 Continue

   See, [H10.1.1].

13.2 Success 2xx

13.3 Redirection 3xx



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   The notation "3rr" indicates response codes from 300 to 399 inclusive
   which are meant for redirection. The response code 304 is excluded
   from this set, as it is not used for redirection.

   See [H10.3] for definition of status code 300 to 305. However
   comments are given for some to how they apply to RTSP.

   Within RTSP, redirection may be used for load balancing or
   redirecting stream requests to a server topologically closer to the
   client.  Mechanisms to determine topological proximity are beyond the
   scope of this specification.

   A 3rr code MAY be used to respond to any request. It is RECOMMENDED
   that they are used if necessary before a session is established, i.e.
   in response to DESCRIBE or SETUP. However in cases where a server is
   not able to send a REDIRECT request to the client, the server MAY
   need to resort to using 3rr responses to inform a client with a
   established session about the need for redirecting the session. If an
   3rr response is received for an request in relation to a established
   session, the client SHOULD send a TEARDOWN request for the session,
   and MAY reestablish the session using the resource indicated by the
   Location.

   If the the Location header is used in a response it SHALL contain an
   absolute URI pointing out the media resource the client is redirected
   to, the URI SHALL NOT only contain the host name.

13.3.1 300 Multiple Choices

   See [H10.3.1].

13.3.2 301 Moved Permanently

   The request resource are moved permanently and resides now at the URI
   given by the location header. The user client SHOULD redirect
   automatically to the given URI. This response MUST NOT contain a
   message-body. The Location header MUST be included in the response.

13.3.3 302 Found

   The requested resource resides temporarily at the URI given by the
   Location header. The Location header MUST be included in the
   response. This response is intended to be used for many types of
   temporary redirects; e.g., load balancing. It is RECOMMENDED that the
   server set the reason phrase to something more meaningful than
   "Found" in these cases. The user client SHOULD redirect automatically
   to the given URI. This response MUST NOT contain a message-body.




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   This example shows a client being redirected to a different server:


     C->S: SETUP rtsp://example.com/fizzle/foo RTSP/2.0
           CSeq: 2
           Transport: RTP/AVP/TCP;unicast;interleaved=0-1

     C->S: RTSP/2.0 302 Try Other Server
           CSeq: 2
           Location: rtsp://s2.example.com:8001/fizzle/foo



13.3.4 303 See Other

   This status code SHALL NOT be used in RTSP. However as it was allowed
   to use in RTSP 1.0 (RFC 2326).

13.3.5 304 Not Modified

   If the client has performed a conditional DESCRIBE or SETUP (see
   14.25) and the requested resource has not been modified, the server
   SHOULD send a 304 response. This response MUST NOT contain a
   message-body.

   The response MUST include the following header fields:

        o Date

        o ETag and/or Content-Location, if the header(s) would have been
          sent in a 200 response to the same request.

        o Expires, Cache-Control, and/or Vary, if the field-value might
          differ from that sent in any previous response for the same
          variant.

   This response is independent for the DESCRIBE and SETUP requests.
   That is, a 304 response to DESCRIBE does NOT imply that the resource
   content is unchanged (only the session description) and a 304
   response to SETUP does NOT imply that the resource description is
   unchanged. The ETag and If-Match headers may be used to link the
   DESCRIBE and SETUP in this manner.

13.3.6 305 Use Proxy

   See [H10.3.6].

13.4 Client Error 4xx



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13.4.1 400 Bad Request

   The request could not be understood by the server due to malformed
   syntax. The client SHOULD NOT repeat the request without
   modifications [H10.4.1]. If the request does not have a CSeq header,
   the server MUST NOT include a CSeq in the response.

13.4.2 405 Method Not Allowed

   The method specified in the request is not allowed for the resource
   identified by the Request-URI. The response MUST include an Allow
   header containing a list of valid methods for the requested resource.
   This status code is also to be used if a request attempts to use a
   method not indicated during SETUP, e.g., if a RECORD request is
   issued even though the mode parameter in the Transport header only
   specified PLAY.

13.4.3 451 Parameter Not Understood

   The recipient of the request does not support one or more parameters
   contained in the request. When returning this error message the
   sender SHOULD return a entity body containing the offending
   parameter(s).

13.4.4 452 reserved

   This error code was removed from RFC 2326 [25] and is obsolete.

13.4.5 453 Not Enough Bandwidth

   The request was refused because there was insufficient bandwidth.
   This may, for example, be the result of a resource reservation
   failure.

13.4.6 454 Session Not Found

   The RTSP session identifier in the Session header is missing,
   invalid, or has timed out.

13.4.7 455 Method Not Valid in This State

   The client or server cannot process this request in its current
   state. The response SHALL contain an Allow header to make error
   recovery possible.

13.4.8 456 Header Field Not Valid for Resource

   The server could not act on a required request header. For example,



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   if PLAY contains the Range header field but the stream does not allow
   seeking. This error message may also be used for specifying when the
   time format in Range is impossible for the resource. In that case the
   Accept-Ranges header SHALL be returned to inform the client of which
   format(s) that are allowed.

13.4.9 457 Invalid Range

   The Range value given is out of bounds, e.g., beyond the end of the
   presentation.

13.4.10 458 Parameter Is Read-Only

   The parameter to be set by SET_PARAMETER can be read but not
   modified. When returning this error message the sender SHOULD return
   a entity body containing the offending parameter(s).

13.4.11 459 Aggregate Operation Not Allowed

   The requested method may not be applied on the URI in question since
   it is an aggregate (presentation) URI. The method may be applied on a
   media URI.

13.4.12 460 Only Aggregate Operation Allowed

   The requested method may not be applied on the URI in question since
   it is not an aggregate control (presentation) URI.  The method may be
   applied on the aggregate control URI.

13.4.13 461 Unsupported Transport

   The Transport field did not contain a supported transport
   specification.

13.4.14 462 Destination Unreachable

   The data transmission channel could not be established because the
   client address could not be reached. This error will most likely be
   the result of a client attempt to place an invalid dest_addr
   parameter in the Transport field.

13.4.15 463 Destination Prohibited

   The data transmission channel was not established because the server
   prohibited access to the client address. This error is most likely
   the result of a client attempt to redirect media traffic to another
   destination with a dest_addr parameter in the Transport header.




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13.4.16 464 Data Transport Not Ready Yet

   The data transmission channel to the media destination is not yet
   ready for carrying data. However the responding entity still expects
   that the data transmission channel will be established at this point
   in time. Note however that this may result in a permanent failure
   like 462 "Destination Unreachable".

   An example when this error may occur is in the case a client sends a
   PLAY request to a server prior to ensuring that the TCP connections
   negotiated for carrying media data was successful established (In
   violation of this specification). The server would use this error
   code to indicate that the requested action could not be performed due
   to the failure of completing the connection establishment.

13.4.17 470 Connection Authorization Required

   The secured connection attempt need user or client authorization
   before proceeding. The next hops certificate is included in this
   response in the Accept-Credentials header.

13.4.18 471 Connection Credentials not accepted

   When performing a secure connection over multiple connections, a
   intermediary has refused to connect to the next hop and carry out the
   request due to unacceptable credentials for the used policy.

13.5 Server Error 5xx

13.5.1 551 Option not supported

   A feature-tag given in the Require or the Proxy-Require fields was
   not supported. The Unsupported header SHALL be returned stating the
   feature for which there is no support.

14 Header Field Definitions


   The general syntax for header fields is covered in Section 4.2 This
   section lists the full set of header fields along with notes on
   meaning, and usage. The syntax definition for header fields are
   present in section 19.2.3. Throughout this section, we use [HX.Y] to
   refer to Section X.Y of the current HTTP/1.1 specification RFC 2616
   [3]. Examples of each header field are given.

   Information about header fields in relation to methods and proxy
   processing is summarized in Tables 9, 10, 11, and 12.




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             method        direction      object acronym Body
             _________________________________________________
             DESCRIBE      C -> S         P,S    DES     r
             GET_PARAMETER C -> S, S -> C P,S    GPR     R,r
             OPTIONS       C -> S         P,S    OPT
                           S -> C
             PAUSE         C -> S         P,S    PSE
             PLAY          C -> S         P,S    PLY
             REDIRECT      S -> C         P,S    RDR
             SETUP         C -> S         S      STP
             SET_PARAMETER C -> S, S -> C P,S    SPR     R,r
             TEARDOWN      C -> S         P,S    TRD


   Table 8: Overview of RTSP methods, their direction, and what  objects
   (P:  presentation, S: stream) they operate on. Body notes if a method
   is allowed to carry  body  and  in  which  direction,  R  =  Request,
   r=response. Note: It is allowed for all error messages 4xx and 5xx to
   have a body

   The "where" column describes the request and response types in which
   the header field can be used. Values in this column are:

        R: header field may only appear in requests;

        r: header field may only appear in responses;

        2xx, 4xx, etc.: A numerical value or range indicates response
             codes with which the header field can be used;

        c: header field is copied from the request to the response.

   An empty entry in the "where" column indicates that the header field
   may be present in both requests and responses.

   The "proxy" column describes the operations a proxy may perform on a
   header field. An empty proxy column indicates that the proxy SHALL
   NOT do any changes to that header, all allowed operations are
   explicitly stated:

        a: A proxy can add or concatenate the header field if not
             present.

        m: A proxy can modify an existing header field value.

        d: A proxy can delete a header field value.




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        r: A proxy needs to be able to read the header field, and thus
             this header field cannot be encrypted.

   The rest of the columns relate to the presence of a header field in a
   method. The method names when abbreviated, are according to table 8:

        c: Conditional; requirements on the header field depend on the
             context of the message.

        m: The header field is mandatory.

        m*: The header field SHOULD be sent, but clients/servers need to
             be prepared to receive messages without that header field.

        o: The header field is optional.

        *: The header field is SHALL be present if the message body is
             not empty. See sections 14.16, 14.18 and 4.3 for details.

        -: The header field is not applicable.

   "Optional" means that a Client/Server MAY include the header field in
   a request or response. The Client/Server behavior when receiving such
   headers varies, for some it may ignore the header field, in other
   case it is request to process the header. This is regulated by the
   method and header descriptions. Example of such headers that require
   processing are the Require and Proxy-Require header fields discussed
   in 14.37 and 14.31. A "mandatory" header field MUST be present in a
   request, and MUST be understood by the Client/Server receiving the
   request. A mandatory response header field MUST be present in the
   response, and the header field MUST be understood by the
   Client/Server processing the response. "Not applicable" means that
   the header field MUST NOT be present in a request. If one is placed
   in a request by mistake, it MUST be ignored by the Client/Server
   receiving the request. Similarly, a header field labeled "not
   applicable" for a response means that the Client/Server MUST NOT
   place the header field in the response, and the Client/Server MUST
   ignore the header field in the response.

   An RTSP agent SHALL ignore extension headers that are not understood.

   The From and Location header fields contain an URI. If the URI
   contains a comma, or semicolon, the URI MUST be enclosed in double
   quotas ("). Any URI parameters are contained within these quotas. If
   the URI is not enclosed in double quotas, any semicolon- delimited
   parameters are header-parameters, not URI parameters.





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   Header                  Where  Proxy DES OPT SETUP PLAY PAUSE TRD
   _________________________________________________________________
   Accept                    R           o   -    -    -     -   -
   Accept-Credentials        R      r    o   o    o    o     o   o
   Accept-Encoding           R      r    o   -    -    -     -   -
   Accept-Language           R      r    o   -    -    -     -   -
   Accept-Ranges             R      r    -   -    m    -     -   -
   Accept-Ranges             r      r    -   -    o    -     -   -
   Accept-Ranges            456     r    -   -    -    o     -   -
   Allow                     r     am    c   c    c    -     -   -
   Allow                    405    am    m   m    m    m     m   m
   Authorization             R           o   o    o    o     o   o
   Bandwidth                 R           o   o    o    o     -   -
   Blocksize                 R           o   -    o    o     -   -
   Cache-Control                    r    o   -    o    -     -   -
   Connection                            o   o    o    o     o   o
   Connection-Credentials 470,407  ar    o   o    o    o     o   o
   Content-Base              r           o   -    -    -     -   -
   Content-Base           4xx,5xx        o   o    o    o     o   o
   Content-Encoding          R      r    -   -    -    -     -   -
   Content-Encoding          r      r    o   -    -    -     -   -
   Content-Encoding       4xx,5xx   r    o   o    o    o     o   o
   Content-Language          R      r    -   -    -    -     -   -
   Content-Language          r      r    o   -    -    -     -   -
   Content-Language       4xx,5xx   r    o   o    o    o     o   o
   Content-Length            r      r    *   -    -    -     -   -
   Content-Length         4xx,5xx   r    *   *    *    *     *   *
   Content-Location          r           o   -    -    -     -   -
   Content-Location       4xx,5xx        o   o    o    o     o   o
   Content-Type              r           *   -    -    -     -   -
   Content-Type           4xx,5xx        *   *    *    *     *   *
   CSeq                     Rc     rm    m   m    m    m     m   m
   Date                            am    o   o    o    o     o   o
   ETag                      r      r    o   -    o    -     -   -
   Expires                   r      r    o   -    -    -     -   -
   From                      R      r    o   o    o    o     o   o
   If-Match                  R      r    -   -    o    -     -   -
   If-Modified-Since         R      r    o   -    o    -     -   -
   If-None-Match             R      r    o   -    -    -     -   -
   Last-Modified             r      r    o   -    -    -     -   -
   Location                 3rr          o   o    o    o     o   o


   Table 9: Overview of RTSP header  fields  (A-L)  related  to  methods
   DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.




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   Header               Where  Proxy DES OPT SETUP PLAY PAUSE TRD
   ______________________________________________________________
   Proxy-Authenticate    407    amr   m   m    m    m     m    m
   Proxy-Authorization    R     rd    o   o    o    o     o    o
   Proxy-Require          R     ar    o   o    o    o     o    o
   Proxy-Require          r      r    c   c    c    c     c    c
   Proxy-Supported        R     amr   c   c    c    c     c    c
   Proxy-Supported        r           c   c    c    c     c    c
   Public                 r    admr   -   m    -    -     -    -
   Public                501   admr   m   m    m    m     m    m
   Range                  R           -   -    -    o     -    -
   Range                  r           -   -    c    m     m    -
   Referer                R           o   o    o    o     o    o
   Require                R           o   o    o    o     o    o
   Retry-After         3rr,503        o   o    o    -     -    -
   RTP-Info               r           -   -    o    c     -    -
   Scale                              -   -    -    o     -    -
   Session                R      r    -   o    o    m     m    m
   Session                r      r    -   c    m    m     m    o
   Server                 R      r    -   o    -    -     -    -
   Server                 r      r    o   o    o    o     o    o
   Speed                              -   -    -    o     -    -
   Supported              R     amr   o   o    o    o     o    o
   Supported              r     amr   c   c    c    c     c    c
   Timestamp              R    admr   o   o    o    o     o    o
   Timestamp              c    admr   m   m    m    m     m    m
   Transport                    amr   -   -    m    -     -    -
   Unsupported            r           c   c    c    c     c    c
   User-Agent             R          m*  m*   m*    m*   m*   m*
   Vary                   r           c   c    c    c     c    c
   Via                    R     amr   o   o    o    o     o    o
   Via                    c     dr    m   m    m    m     m    m
   WWW-Authenticate      401          m   m    m    m     m    m

______________________________________________________________
   Header               Where  Proxy DES OPT SETUP PLAY PAUSE TRD



   Table 10: Overview of RTSP header fields  (P-W)  related  to  methods
   DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.





14.1 Accept




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   The Accept request-header field can be used to specify certain
   presentation description content types which are acceptable for the
   response.

   See [H14.1] for syntax.

   Example of use:


     Accept: application/example q=1.0, application/sdp



14.2 Accept-Credentials

   The Accept-Credentials header is a request header used to indicate to
   any trusted intermediary how to handle further secured connections to
   proxies or servers. See Section 18 for the usage of this header. It
   SHALL NOT be included in server to client requests.

   In a request the header SHALL contain the method (User, Proxy, or
   Any) for approving credentials selected by the requestor. The method
   SHALL NOT be changed by any proxy. If the method is "User" the header
   contains zero or more of credentials that the client accept.  The
   header may contain zero credentials in the first RTSP request to a
   RTSP server when using the "User" method. This as the client has not
   yet received any credentials to accept. Each credential SHALL consist
   of one URI identifying the proxy or server, the hash algorithm
   identifier, and the hash over that entity's DER encoded certificate
   [14] in Base64. All RTSP clients and proxies SHALL implement the
   SHA-1 [15] algorithm for computation of the hash of the DER encoded
   certificate. The SHA-1 algorithm is identified by the token "sha-1".

   The intention with allowing for other hash algorithms is to enable
   the future retirement of algorithms that are not implemented
   somewhere else than here. Thus the definition of future algorithms
   for this purpose is intended to be extremely limited.


   Example:
     Accept-Credentials:User,
       "rtsps://proxy2.example.com/";sha-1;exaIl9VMbQMOFGClx5rXnPJKVNI=,
       "rtsps://server.example.com/";sha-1;lurbjj5khhB0NhIuOXtt4bBRH1M=



14.3 Accept-Encoding




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   Header                  Where  Proxy GPR SPR RDR
   __________________________________________________
   Accept-Credentials        R      r    o   o   o
   Allow                    405    amr   m   m   m
   Authorization             R           o   o   o
   Bandwidth                 R           -   o   -
   Blocksize                 R           -   o   -
   Connection                            o   o   o
   Connection-Credentials 470,407  ar    o   o   o
   Content-Base              R           o   o  -
   Content-Base              r           o   o  -
   Content-Base           4xx,5xx        o   o  o
   Content-Encoding          R      r    o   o  -
   Content-Encoding          r      r    o   o  -
   Content-Encoding       4xx,5xx   r    o   o  o
   Content-Language          R      r    o   o  -
   Content-Language          r      r    o   o  -
   Content-Language       4xx,5xx   r    o   o  o
   Content-Length            R      r    *   *  -
   Content-Length            r      r    *   *  -
   Content-Length         4xx,5xx   r    *   *  *
   Content-Location          R           o   o  -
   Content-Location          r           o   o  -
   Content-Location       4xx,5xx        o   o  o
   Content-Type              R           *   *  -
   Content-Type              r           *   *  -
   Content-Type             4xx          *   *  *
   CSeq                     R,c    mr    m   m  m
   Date                      R      a    o   o  m
   Date                      r     am    o   o  o
   From                      R      r    o   o  o
   Last-Modified             R      r    -   -  -
   Last-Modified             r      r    o   -  -
   Location                 3rr          o   o  o
   Location                  R           -   -  m
   Proxy-Authenticate       407    amr   m   m  m
   Proxy-Authorization       R     rd    o   o  o
   Proxy-Require             R     ar    o   o  o
   Proxy-Require             r      r    c   c  c
   Proxy-Supported           R     amr   c   c  c
   Proxy-Supported           r           c   c  c
   Public                   501   admr   m   m  m

__________________________________________________
   Header                  Where  Proxy GPR SPR RDR


   Table 11: Overview of RTSP header fields  (A-P)  related  to  methods
   GET_PARAMETER, SET_PARAMETER, and REDIRECT.


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               Header            Where  Proxy GPR SPR RDR
               ____________________________________________
               Range               R           -   -  o
               Referer             R           o   o  o
               Require             R      r    o   o  o
               Retry-After      3rr,503        o   o  -
               Scale                           -   -  -
               Session             R      r    o   o  o
               Session             r      r    c   c  o
               Server              R      r    o   o  o
               Server              r      r    o   o  -
               Supported           R    adrm   o   o  o
               Supported           r    adrm   c   c  c
               Timestamp           R    adrm   o   o  o
               Timestamp           c    adrm   m   m  m
               Unsupported         r     arm   c   c  c
               User-Agent          R      r   m*  m*  -
               User-Agent          r      r    -   -  m*
               Vary                r           c   c  -
               Via                 R     amr   o   o  o
               Via                 c     dr    m   m  m
               WWW-Authenticate   401          m   m  m

____________________________________________
               Header            Where  Proxy GPR SPR RDR


   Table 12: Overview of RTSP header fields  (R-W)  related  to  methods
   GET_PARAMETER, SET_PARAMETER, and REDIRECT.


   See [H14.3]

14.4 Accept-Language

   See [H14.4]. Note that the language specified applies to the
   presentation description and any reason phrases, not the media
   content.

14.5 Accept-Ranges

   The Accept-Ranges request and response-header field allows indication
   of the format supported in the Range header. The client SHALL include
   the header in SETUP requests to indicate which formats it support to
   receive in PLAY and PAUSE responses, and REDIRECT requests. The
   server SHALL include the header in SETUP and 456 error responses to
   indicate the formats supported for the resource indicated by the
   request URI.




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   Accept-Ranges: NPT, SMPTE



   This header has the same syntax as [H14.5] and the syntax is defined
   in 19.2.3. However, new range-units are defined.

14.6 Allow

   The Allow entity-header field lists the methods supported by the
   resource identified by the Request-URI. The purpose of this field is
   to strictly inform the recipient of valid methods associated with the
   resource. An Allow header field MUST be present in a 405 (Method Not
   Allowed) response. See [H14.7] for syntax definition. The Allow
   header MUST also be present in all OPTIONS responses where the
   content of the header will not include exactly the same methods as
   listed in the Public header.

   The Allow SHALL also be included in SETUP and DESCRIBE responses, if
   the methods allowed for the resource is different than the minimal
   implementation set.

   Example of use:

     Allow: SETUP, PLAY, SET_PARAMETER, DESCRIBE



14.7 Authorization

   See [H14.8]

14.8 Bandwidth

   The Bandwidth request-header field describes the estimated bandwidth
   available to the client, expressed as a positive integer and measured
   in bits per second. The bandwidth available to the client may change
   during an RTSP session, e.g., due to mobility, congestion, etc.

   Example:

     Bandwidth: 62360



14.9 Blocksize

   The Blocksize request-header field is sent from the client to the



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   media server asking the server for a particular media packet size.
   This packet size does not include lower-layer headers such as IP,
   UDP, or RTP. The server is free to use a blocksize which is lower
   than the one requested. The server MAY truncate this packet size to
   the closest multiple of the minimum, media-specific block size, or
   override it with the media-specific size if necessary. The block size
   MUST be a positive decimal number, measured in octets. The server
   only returns an error (4xx) if the value is syntactically invalid.

14.10 Cache-Control

   The Cache-Control general-header field is used to specify directives
   that MUST be obeyed by all caching mechanisms along the
   request/response chain.

   Cache directives MUST be passed through by a proxy or gateway
   application, regardless of their significance to that application,
   since the directives may be applicable to all recipients along the
   request/response chain. It is not possible to specify a cache-
   directive for a specific cache.

   Cache-Control should only be specified in a SETUP request and its
   response. Note: Cache-Control does not govern the caching of
   responses as for HTTP, instead it applies to the media stream
   identified by the SETUP request. The RTSP requests are generally not
   cacheable, for further information see section 16. Below is the
   description of the cache directives that can be included in the
   Cache-Control header.

        no-cache: Indicates that the media stream MUST NOT be cached
             anywhere. This allows an origin server to prevent caching
             even by caches that have been configured to return stale
             responses to client requests.

        public: Indicates that the media stream is cacheable by any
             cache.

        private: Indicates that the media stream is intended for a
             single user and MUST NOT be cached by a shared cache. A
             private (non-shared) cache may cache the media stream.

        no-transform: An intermediate cache (proxy) may find it useful
             to convert the media type of a certain stream. A proxy
             might, for example, convert between video formats to save
             cache space or to reduce the amount of traffic on a slow
             link.  Serious operational problems may occur, however,
             when these transformations have been applied to streams
             intended for certain kinds of applications. For example,



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             applications for medical imaging, scientific data analysis
             and those using end-to-end authentication all depend on
             receiving a stream that is bit-for-bit identical to the
             original media stream. Therefore, if a response includes
             the no-transform directive, an intermediate cache or proxy
             MUST NOT change the encoding of the stream.  Unlike HTTP,
             RTSP does not provide for partial transformation at this
             point, e.g., allowing translation into a different
             language.

        only-if-cached: In some cases, such as times of extremely poor
             network connectivity, a client may want a cache to return
             only those media streams that it currently has stored, and
             not to receive these from the origin server. To do this,
             the client may include the only-if-cached directive in a
             request. If it receives this directive, a cache SHOULD
             either respond using a cached media stream that is
             consistent with the other constraints of the request, or
             respond with a 504 (Gateway Timeout) status. However, if a
             group of caches is being operated as a unified system with
             good internal connectivity, such a request MAY be forwarded
             within that group of caches.

        max-stale: Indicates that the client is willing to accept a
             media stream that has exceeded its expiration time. If
             max-stale is assigned a value, then the client is willing
             to accept a response that has exceeded its expiration time
             by no more than the specified number of seconds. If no
             value is assigned to max-stale, then the client is willing
             to accept a stale response of any age.

        min-fresh: Indicates that the client is willing to accept a
             media stream whose freshness lifetime is no less than its
             current age plus the specified time in seconds. That is,
             the client wants a response that will still be fresh for at
             least the specified number of seconds.

        must-revalidate: When the must-revalidate directive is present
             in a SETUP response received by a cache, that cache MUST
             NOT use the entry after it becomes stale to respond to a
             subsequent request without first revalidating it with the
             origin server. That is, the cache is required to do an
             end-to-end revalidation every time, if, based solely on the
             origin server's Expires, the cached response is stale.)

        proxy-revalidate: The proxy-revalidate directive has the same
             meaning as the must-revalidate directive, except that it
             does not apply to non-shared user agent caches. It can be



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             used on a response to an authenticated request to permit
             the user's cache to store and later return the response
             without needing to revalidate it (since it has already been
             authenticated once by that user), while still requiring
             proxies that service many users to revalidate each time (in
             order to make sure that each user has been authenticated).
             Note that such authenticated responses also need the public
             cache control directive in order to allow them to be cached
             at all.

        max-age: When an intermediate cache is forced, by means of a
             max-age=0 directive, to revalidate its own cache entry, and
             the client has supplied its own validator in the request,
             the supplied validator might differ from the validator
             currently stored with the cache entry. In this case, the
             cache MAY use either validator in making its own request
             without affecting semantic transparency.

             However, the choice of validator might affect performance.
             The best approach is for the intermediate cache to use its
             own validator when making its request. If the server
             replies with 304 (Not Modified), then the cache can return
             its now validated copy to the client with a 200 (OK)
             response. If the server replies with a new entity and cache
             validator, however, the intermediate cache can compare the
             returned validator with the one provided in the client's
             request, using the strong comparison function. If the
             client's validator is equal to the origin server's, then
             the intermediate cache simply returns 304 (Not Modified).
             Otherwise, it returns the new entity with a 200 (OK)
             response.

14.11 Connection

   See [H14.10]. The use of the connection option "close" in RTSP
   messages SHOULD be limited to error messages when the server is
   unable to recover and therefore see it necessary to close the
   connection. The reason is that the client has the choice of
   continuing using a connection indefinitely, as long as it sends valid
   messages.

14.12 Connection-Credentials

   The Connection-Credentials response header is used to carry the
   credentials of any next hop that need to be approved by the
   requestor. It SHALL only be used in server to client responses.

   The Connection-Credentials header in an RTSP response SHALL, if



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   included, contain the credentials information of the next hop that an
   intermediary needs to securely connect to. The credential MUST
   include the URI of the next proxy or server and the DER encoded
   X.509v3 [14] certificate in base64 [36].


   Example:

   Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...



14.13 Content-Base

   The Content-Base entity-header field may be used to specify the base
   URI for resolving relative URIs within the entity.

   Content-Base: rtsp://media.example.com/movie/twister


   If no Content-Base field is present, the base URI of an entity is
   defined either by its Content-Location (if that Content-Location URI
   is an absolute URI) or the URI used to initiate the request, in that
   order of precedence. Note, however, that the base URI of the contents
   within the entity-body may be redefined within that entity-body.

14.14 Content-Encoding

   See [H14.11]

14.15 Content-Language

   See [H14.12]

14.16 Content-Length

   The Content-Length general-header field contains the length of the
   body (entity) of the message (i.e. after the double CRLF following
   the last header). Unlike HTTP, it MUST be included in all messages
   that carry body beyond the header portion of the message. If it is
   missing, a default value of zero is assumed. It is interpreted
   according to [H14.13].

14.17 Content-Location

   See [H14.14]

14.18 Content-Type



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   See [H14.17]. Note that the content types suitable for RTSP are
   likely to be restricted in practice to presentation descriptions and
   parameter-value types.

14.19 CSeq

   The CSeq general-header field specifies the sequence number for an
   RTSP request-response pair. This field MUST be present in all
   requests and responses. For every RTSP request containing the given
   sequence number, the corresponding response will have the same
   number. Any retransmitted request MUST contain the same sequence
   number as the original (i.e. the sequence number is not incremented
   for retransmissions of the same request). For each new RTSP request
   the CSeq value SHALL be incremented by one.  The initial sequence
   number MAY be any number, however it is RECOMMENDED to start at 0.
   Each sequence number series is unique between each requester and
   responder, i.e.  the client has one series for its request to a
   server and the server has another when sending request to the client.
   Each requester and responder is identified with its network address.

   Proxies that aggregate several sessions on the same transport will
   regularly need to renumber the CSeq header field in requests and
   responses to fulfill the rules for the header.

   Example:

   CSeq: 239



14.20 Date

   See [H14.18]. An RTSP message containing a body MUST include a Date
   header if the sending host has a clock. Servers SHOULD include a Date
   header in all other RTSP messages.

14.21 ETag

   The ETag response header MAY be included in DESCRIBE or SETUP
   responses. The entity tags (Section 3.8) returned in a DESCRIBE
   response, and the one in SETUP refers to the presentation, i.e. both
   the returned session description and the media stream. This allows
   for verification that one has the right session description to a
   media resource at the time of the SETUP request. However it has the
   disadvantage that a change in any of the parts results in
   invalidation of all the parts.

   If the ETag is provided both inside the entity, e.g. within the



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   "a=etag" attribute in SDP, and in the response message, then both
   tags SHALL be identical. It is RECOMMENDED that the ETag is primarily
   given in the RTSP response message, to ensure that caches can use the
   ETag without requiring content inspection. However for session
   descriptions that are distributed outside of RTSP, for example using
   HTTP, etc. it will be necessary to include the entity tag in the
   session description as specified in Section C.1.9.

   SETUP and DESCRIBE requests can be made conditional upon the ETag
   using the headers If-Match (Section 14.24) and If-None-Match (Section
   14.26).

14.22 Expires

   The Expires entity-header field gives a date and time after which the
   description or media-stream should be considered stale. The
   interpretation depends on the method:

        DESCRIBE response: The Expires header indicates a date and time
             after which the presentation description (body) SHOULD be
             considered stale.

        SETUP response: The Expires header indicate a date and time
             after which the media stream SHOULD be considered stale.

   A stale cache entry may not normally be returned by a cache (either a
   proxy cache or an user agent cache) unless it is first validated with
   the origin server (or with an intermediate cache that has a fresh
   copy of the entity). See section 16 for further discussion of the
   expiration model.

   The presence of an Expires field does not imply that the original
   resource will change or cease to exist at, before, or after that
   time.

   The format is an absolute date and time as defined by HTTP-date in
   [H3.3]; it MUST be in RFC1123-date format:

   An example of its use is

     Expires: Thu, 01 Dec 1994 16:00:00 GMT



   RTSP/2.0 clients and caches MUST treat other invalid date formats,
   especially including the value "0", as having occurred in the past
   (i.e., already expired).




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   To mark a response as "already expired," an origin server should use
   an Expires date that is equal to the Date header value. To mark a
   response as "never expires," an origin server SHOULD use an Expires
   date approximately one year from the time the response is sent.
   RTSP/2.0 servers SHOULD NOT send Expires dates more than one year in
   the future.

   The presence of an Expires header field with a date value of some
   time in the future on a media stream that otherwise would by default
   be non-cacheable indicates that the media stream is cacheable, unless
   indicated otherwise by a Cache-Control header field (Section 14.10).

14.23 From

   See [H14.22].

14.24 If-Match

   See [H14.24].

   The If-Match request-header field is especially useful for ensuring
   the integrity of the presentation description, in both the case where
   it is fetched via means external to RTSP (such as HTTP), or in the
   case where the server implementation is guaranteeing the integrity of
   the description between the time of the DESCRIBE message and the
   SETUP message. By including the ETag given in or with the session
   description in a SETUP request, the client ensures that resources set
   up are matching the description. A SETUP request for which the ETag
   validation check fails, SHALL responde using 412 (Precondition
   Failed).

   This validation check is also very useful if a session has been
   redirected from one server to another.

14.25 If-Modified-Since

   The If-Modified-Since request-header field is used with the DESCRIBE
   and SETUP methods to make them conditional. If the requested variant
   has not been modified since the time specified in this field, a
   description will not be returned from the server (DESCRIBE) or a
   stream will not be set up (SETUP). Instead, a 304 (Not Modified)
   response SHALL be returned without any message-body.

   An example of the field is:

     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT





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14.26 If-None-Match

   See [H14.26].

   This request header can be used with one or several entity tags to
   make DESCRIBE requests conditional. A new session description is
   retrieved only if another entity than the ones already available
   would be included. If the entity available for delivery is matching
   the one the client already has, then a 304 (Not Modified) response is
   given.

14.27 Last-Modified

   The Last-Modified entity-header field indicates the date and time at
   which the origin server believes the presentation description or
   media stream was last modified. See [H14.29]. For the methods
   DESCRIBE, the header field indicates the last modification date and
   time of the description, for SETUP that of the media stream.

14.28 Location

   See [H14.30].

14.29 Proxy-Authenticate

   See [H14.33].

14.30 Proxy-Authorization

   See [H14.34].

14.31 Proxy-Require

   The Proxy-Require request-header field is used to indicate proxy-
   sensitive features that MUST be supported by the proxy. Any Proxy-
   Require header features that are not supported by the proxy MUST be
   negatively acknowledged by the proxy to the client using the
   Unsupported header. The proxy SHALL use the 551 (Option Not
   Supported) status code in the response. Any feature-tag included in
   the Proxy-Require does not apply to the end-point (server or client).
   To ensure that a feature is supported by both proxies and servers the
   tag needs to be included in also a Require header.

   See Section 14.37 for more details on the mechanics of this message
   and a usage example.

   Example of use:




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      Proxy-Require: play.basic



14.32 Proxy-Supported

   The Proxy-Supported header field enumerates all the extensions
   supported by the proxy using feature-tags. The header carries the
   intersection of extensions supported by the forwarding proxies. The
   Proxy-Supported header MAY be included in any request by a proxy. It
   SHALL be added by any proxy if the Supported header is present in a
   request. When present in a request, the receiver MUST in the response
   copy the received Proxy-Supported header.

   The Proxy-Supported header field contains a list of feature-tags
   applicable to proxies, as described in Section 3.7. The list are the
   intersection of all feature-tags understood by the proxies. To
   achieve an intersection, the proxy adding the Proxy-Supported header
   includes all proxy feature-tags it understands. Any proxy receiving a
   request with the header, checks the list and removes any feature-tag
   it do not support. A Proxy-Supported header present in the response
   SHALL NOT be touched by the proxies.

   Example:

     C->P1: OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech

    P1->P2: OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech
            Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
            Via: 2.0 prox1.example.com

    P2->S:  OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech
            Proxy-Supported: proxy-foo, proxy-blech
            Via: 2.0 prox1.example.com, 2.0 prox2.example.com

     S->C:  RTSP/2.0 200 OK
            Supported: foo, bar, baz
            Proxy-Supported: proxy-foo, proxy-blech
            Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
            Via: 2.0 prox1.example.com, 2.0 prox2.example.com




14.33 Public



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   The Public response header field lists the set of methods supported
   by the response sender. This header applies to the general
   capabilities of the sender and its only purpose is to indicate the
   sender's capabilities to the recipient. The methods listed may or may
   not be applicable to the Request-URI; the Allow header field (section
   14.7) MAY be used to indicate methods allowed for a particular URI.

   Example of use:

      Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN



   In the event that there are proxies between the sender and the
   recipient of a response, each intervening proxy MUST modify the
   Public header field to remove any methods that are not supported via
   that proxy. The resulting Public header field will contain an
   intersection of the sender's methods and the methods allowed through
   by the intervening proxies.

        In general proxies should allow all methods to
        transparently pass through from the sending RTSP agent to
        the receiving RTSP agent, but there may be cases where this
        is not desirable for a given proxy. Modification of the
        Public response header field by the intervening proxies
        ensures that the request sender gets an accurate response
        indicating the methods that can be used on the target agent
        via the proxy chain.

14.34 Range

   The Range header specifies a time range in PLAY (Section 11.4), PAUSE
   (Section 11.5), SETUP (Section 11.3), and REDIRECT (Section 11.9)
   requests and responses.

   The range can be specified in a number of units. This specification
   defines smpte (Section 3.4), npt (Section 3.5), and clock (Section
   3.6) range units. While byte ranges [H14.35.1] and other extended
   units MAY be used, their behavior is unspecified since they are not
   normally meaningful in RTSP. Servers supporting the Range header MUST
   understand the NPT range format and SHOULD understand the SMPTE range
   format. If the Range header is sent in a time format that is not
   understood, the recipient SHOULD return 456 (Header Field Not Valid
   for Resource) and include an Accept-Ranges header indicating the
   supported time formats for the given resource.

   The Range header MAY contain a time parameter in UTC, specifying the
   time at which the operation is to be made effective. This



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   functionality SHALL be used only with the REDIRECT method.

   Ranges are half-open intervals, including the first point, but
   excluding the second point. In other words, a range of A-B starts
   exactly at time A, but stops just before B. Only the start time of a
   media unit such as a video or audio frame is relevant. For example,
   assume that video frames are generated every 40 ms. A range of
   10.0-10.1 would include a video frame starting at 10.0 or later time
   and would include a video frame starting at 10.08, even though it
   lasted beyond the interval. A range of 10.0-10.08, on the other hand,
   would exclude the frame at 10.08.

   Example:

     Range: clock=19960213T143205Z-;time=19970123T143720Z



        The notation is similar to that used for the HTTP/1.1 [3]
        byte-range header. It allows clients to select an excerpt
        from the media object, and to play from a given point to
        the end as well as from the current location to a given
        point.

   By default, range intervals increase, where the second point is
   larger than the first point.

   Example:

       Range: npt=10-15



   However, range intervals can also decrease if the Scale header (see
   section 14.39) indicates a negative scale value. For example, this
   would be the case when a playback in reverse is desired.

   Example:

       Scale: -1
       Range: npt=15-10



   Decreasing ranges are still half open intervals as described above.
   Thus, for range A-B, A is closed and B is open. In the above example,
   15 is closed and 10 is open. An exception to this rule is the case
   when B=0 in a decreasing range. In this case, the range is closed on



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   both ends, as otherwise there would be no way to reach 0 on a reverse
   playback for formats that have such a notion, like NPT and SMPTE.

   Example:

       Scale: -1
       Range: npt=15-0



   In this range both 15 and 0 are closed.

   A decreasing range interval without a corresponding negative Scale
   header is not valid.

14.35 Referer

   See [H14.36]. The URI refers to that of the presentation description,
   typically retrieved via HTTP.

14.36 Retry-After

   See [H14.37].

14.37 Require

   The Require request-header field is used by clients or servers to
   ensure that the other end-point supports features that are required
   in respect to this request. It can also be used to query if the other
   end-point supports certain features, however the use of the Supported
   (Section 14.43) is much more effective in this purpose. The server
   MUST respond to this header by using the Unsupported header to
   negatively acknowledge those feature-tags which are NOT supported.
   The response SHALL use the error code 551 (Option Not Supported).
   This header does not apply to proxies, for the same functionality in
   respect to proxies see, header Proxy-Require (Section 14.31).


        This is to make sure that the client-server interaction
        will proceed without delay when all features are understood
        by both sides, and only slow down if features are not
        understood (as in the example below). For a well-matched
        client-server pair, the interaction proceeds quickly,
        saving a round-trip often required by negotiation
        mechanisms. In addition, it also removes state ambiguity
        when the client requires features that the server does not
        understand.




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   Example:

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/2.0
           CSeq: 302
           Require: funky-feature
           Funky-Parameter: funkystuff

   S->C:   RTSP/2.0 551 Option not supported
           CSeq: 302
           Unsupported: funky-feature




   In this example, "funky-feature" is the feature-tag which indicates
   to the client that the fictional Funky-Parameter field is required.
   The relationship between "funky-feature" and Funky-Parameter is not
   communicated via the RTSP exchange, since that relationship is an
   immutable property of "funky-feature" and thus should not be
   transmitted with every exchange.

   Proxies and other intermediary devices SHALL ignore this header. If a
   particular extension requires that intermediate devices support it,
   the extension should be tagged in the Proxy-Require field instead
   (see Section 14.31).

14.38 RTP-Info

   The RTP-Info response-header field is used to set RTP-specific
   parameters in the PLAY response. For streams using RTP as transport
   protocol the RTP-Info header SHOULD be part of a 200 response to
   PLAY.


        The exclusion of the RTP-Info in a PLAY response for RTP
        transported media will result in that a client needs to
        synchronize the media streams using RTCP. This may have
        negative impact as the RTCP can be lost, and does not need
        to be particulary timely in their arrival. Also
        functionality as informing the client from which packet a
        seek has occurred is affected.

   The RTP-Info MAY also be included in SETUP responses to provide
   synchronization information when changing transport parameters, see
   11.3.

   The header can carry the following parameters:




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        url: Indicates the stream URI which for which the following RTP
             parameters correspond, this URI MUST be the same used in
             the SETUP request for this media stream. Any relative URI
             SHALL use the Request-URI as base URI. This parameter SHALL
             be present.

        ssrc: The Synchronization source (SSRC) that the RTP timestamp
             and sequence number provide applies to. This parameter
             SHALL be present.

        seq: Indicates the sequence number of the first packet of the
             stream that is direct result of the request. This allows
             clients to gracefully deal with packets when seeking. The
             client uses this value to differentiate packets that
             originated before the seek from packets that originated
             after the seek. Note that a client may not receive the
             packet with the expressed sequence number, and instead
             packets with a higher sequence number, due to packet loss
             or reordering. This parameter is RECOMMENDED to be present.

        rtptime: SHALL indicate the RTP timestamp value corresponding to
             the start time value in the Range response header, or if
             not explicitly given the implied start point. The client
             uses this value to calculate the mapping of RTP time to NPT
             or other media timescale. This parameter SHOULD be present
             to ensure inter-media synchronization is achieved. There
             exist no requirement that any received RTP packet will have
             the same RTP timestamp value as the one in the parameter
             used to establish synchronization.


        A mapping from RTP timestamps to NTP timestamps (wall
        clock) is available via RTCP. However, this information is
        not sufficient to generate a mapping from RTP timestamps to
        media clock time (NPT, etc.). Furthermore, in order to
        ensure that this information is available at the necessary
        time (immediately at startup or after a seek), and that it
        is delivered reliably, this mapping is placed in the RTSP
        control channel.

   In order to compensate for drift for long, uninterrupted
   presentations, RTSP clients should additionally map NPT to NTP, using
   initial RTCP sender reports to do the mapping, and later reports to
   check drift against the mapping.

   Example:

   Range:npt=3.25-15



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   RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
            rtptime=12345678,url="rtsp://example.com/foo/video"
            ssrc=9A9DE123:seq=30211;rtptime=29567112

   Lets assume that audio uses a 16kHz RTP timestamp clock and Video
   a 90kHz RTP timestamp clock. Then the media synchronization is
   depicted in the following way.

   NPT    3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
   Audio               PA A
   Video                  V    PV

   X: NPT time value = 3.25, from Range header.
   A: RTP timestamp value for Audio from RTP-Info header (12345678).
   V: RTP timestamp value for Video from RTP-Info header (29567112).
   PA: RTP audio packet carrying an RTP timestamp of 12344878. Which
       corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
   PV: RTP video packet carrying an RTP timestamp of 29573412. Which
       corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32



14.39 Scale

   A scale value of 1 indicates normal play at the normal forward
   viewing rate. If not 1, the value corresponds to the rate with
   respect to normal viewing rate. For example, a ratio of 2 indicates
   twice the normal viewing rate ("fast forward") and a ratio of 0.5
   indicates half the normal viewing rate. In other words, a ratio of 2
   has normal play time increase at twice the wallclock rate. For every
   second of elapsed (wallclock) time, 2 seconds of content will be
   delivered. A negative value indicates reverse direction. For certain
   media transports this may require certain considerations to work
   consistent, see section B.1 for description on how RTP handles this.

   Unless requested otherwise by the Speed parameter, the data rate
   SHOULD not be changed. Implementation of scale changes depends on the
   server and media type. For video, a server may, for example, deliver
   only key frames or selected key frames. For audio, it may time-scale
   the audio while preserving pitch or, less desirably, deliver
   fragments of audio.

   The server should try to approximate the viewing rate, but may
   restrict the range of scale values that it supports. The response
   MUST contain the actual scale value chosen by the server.

   If the server does not implement the possibility to scale, it will
   not return a Scale header. A server supporting Scale operations for



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   PLAY SHALL indicate this with the use of the "play.scale" feature-
   tags.

   When indicating a negative scale for a reverse playback, the Range
   header MUST indicate a decreasing range as described in section
   14.34.

   Example of playing in reverse at 3.5 times normal rate:

     Scale: -3.5
     Range: npt=15-10



14.40 Speed

   The Speed request-header field requests the server to deliver data to
   the client at a particular speed, contingent on the server's ability
   and desire to serve the media stream at the given speed.
   Implementation by the server is OPTIONAL. The default is the bit rate
   of the stream.

   The parameter value is expressed as a decimal ratio, e.g., a value of
   2.0 indicates that data is to be delivered twice as fast as normal. A
   speed of zero is invalid. All speeds may not be possible to support.
   Therefore the actual used speed MUST be included in the response. The
   lack of a response header is indication of lack of support from the
   server of this functionality. Support of the speed functionality are
   indicated by the "play.speed" featuretag.

   Example:

     Speed: 2.5



   Use of this field changes the bandwidth used for data delivery. It is
   meant for use in specific circumstances where preview of the
   presentation at a higher or lower rate is necessary. Implementors
   should keep in mind that bandwidth for the session may be negotiated
   beforehand (by means other than RTSP), and therefore re-negotiation
   may be necessary. When data is delivered over UDP, it is highly
   recommended that means such as RTCP be used to track packet loss
   rates. If the data transport is performed over non-dedicated best-
   effort networks the sender is required to perform congestion control
   of the stream(s). This can result in that the communicated speed is
   impossible to maintain.




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14.41 Server

   See [H14.38], however the header syntax is corrected in section
   19.2.3.

14.42 Session

   The Session request-header and response-header field identifies an
   RTSP session. An RTSP session is created by the server as a result of
   a successful SETUP request and in the response the session identifier
   is given to the client. The RTSP session exist until destroyed by a
   TEARDOWN or timed out by the server.

   The session identifier is chosen by the server (see Section 3.3) and
   MUST be returned in the SETUP response. Once a client receives a
   session identifier, it SHALL be included in any request related to
   that session.  This means that the Session header MUST be included in
   a request using the following methods: PLAY, PAUSE, and TEARDOWN, and
   MAY be included in SETUP, OPTIONS, SET_PARAMETER, GET_PARAMETER, and
   REDIRECT, and SHALL NOT be included in DESCRIBE. In an RTSP response
   the session header SHALL be included in methods, SETUP, PLAY, and
   PAUSE, and MAY be included in methods, TEARDOWN, and REDIRECT, and if
   included in the request of the following methods it SHALL also be
   included in the response, OPTIONS, GET_PARAMETER, and SET_PARAMETER,
   and SHALL NOT be included in DESCRIBE.

   The timeout parameter MAY be included in a SETUP response, and SHALL
   NOT be included in requests. The server uses it to indicate to the
   client how long the server is prepared to wait between RTSP commands
   or other signs of life before closing the session due to lack of
   activity (see below and Section A). The timeout is measured in
   seconds, with a default of 60 seconds (1 minute). The length of the
   session timeout SHALL NOT be changed in a established session.

   The mechanisms for showing liveness of the client is, any RTSP
   request with a Session header, if RTP & RTCP is used an RTCP message,
   or through any other used media protocol capable of indicating
   liveness of the RTSP client. It is RECOMMENDED that a client does not
   wait to the last second of the timeout before trying to send a
   liveness message. The RTSP message may be lost or when using reliable
   protocols, such as TCP, the message may take some time to arrive
   safely at the receiver. To show liveness between RTSP request issued
   to accomplish other things, the following mechanisms can be used, in
   descending order of preference:

        RTCP: If RTP is used for media transport RTCP SHOULD be used. If
             RTCP is used to report transport statistics, it SHALL also
             work as keep alive. The server can determine the client by



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             used network address and port together with the fact that
             the client is reporting on the servers SSRC(s). A downside
             of using RTCP is that it only gives statistical guarantees
             to reach the server. However that probability is so low
             that it can be ignored in most cases. For example, a
             session with 60 seconds timeout and enough bitrate assigned
             to RTCP messages to send a message from client to server on
             average every 5 seconds. That client have for a network
             with 5 % packet loss, the probability to fail showing
             liveness sign in that session within the timeout interval
             of 2.4*E-16. In sessions with shorter timeout times, or
             much higher packet loss, or small RTCP bandwidths SHOULD
             also use any of the mechanisms below.

        SET_PARAMETER: When using SET_PARAMETER for keep alive, no body
             SHOULD be included. This method is the RECOMMENDED RTSP
             method to use in request only intended to perform keep-
             alive.

        OPTIONS: This method does also work. However it causes the
             server to perform more unnecessary processing and result in
             bigger responses than necessary for the task. The reason
             for this is that the server needs to determine what
             capabilities that are associated with the media resource to
             correctly populate the Public and Allow headers.

   Note that a session identifier identifies an RTSP session across
   transport sessions or connections. RTSP requests for a given session
   can use different URIs (Presentation and media URIs). Note, that
   there are restrictions depending on the session which URIs that are
   acceptable for a given method. However, multiple "user" sessions for
   the same URI from the same client will require use of different
   session identifiers.

        The session identifier is needed to distinguish several
        delivery requests for the same URI coming from the same
        client.

   The response 454 (Session Not Found) SHALL be returned if the session
   identifier is invalid.

14.43 Supported

   The Supported header field enumerates all the extensions supported by
   the client or server using feature tags. The header carries the
   extensions supported by the message sending entity.  The Supported
   header MAY be included in any request.  When present in a request,
   the receiver MUST respond with its corresponding Supported header.



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   Note, also in 4xx and 5xx responses is the supported header included.

   The Supported header field contains a list of feature-tags, described
   in Section 3.7, that are understood by the client or server.

   Example:

     C->S:  OPTIONS rtsp://example.com/ RTSP/2.0
            Supported: foo, bar, blech

     S->C:  RTSP/2.0 200 OK
            Supported: bar, blech, baz



14.44 Timestamp

   The Timestamp general-header field describes when the agent sent the
   request. The value of the timestamp is of significance only to the
   agent and may use any timescale.  The responding agent MUST echo the
   exact same value and MAY, if it has accurate information about this,
   add a floating point number indicating the number of seconds that has
   elapsed since it has received the request. The timestamp is used by
   the agent to compute the round-trip time to the responding agent so
   that it can adjust the timeout value for retransmissions. It also
   resolves retransmission ambiguities for unreliable transport of RTSP.

14.45 Transport

   The Transport request and response header field indicates which
   transport protocol is to be used and configures its parameters such
   as destination address, compression, multicast time-to-live and
   destination port for a single stream. It sets those values not
   already determined by a presentation description.

   Transports are comma separated, listed in order of preference.
   Parameters may be added to each transport, separated by a semicolon.
   The server SHOULD return a Transport response-header field in the
   response to indicate the values actually chosen. The Transport header
   field MAY also be used to change certain transport parameters. A
   server MAY refuse to change parameters of an existing stream.

   A Transport request header field MAY contain a list of transport
   options acceptable to the client, in the form of multiple
   transportspec entries. In that case, the server MUST return the
   single (transport-spec) which was actually chosen. The number of
   transportspec entries is expected to be limited as the client will
   get guidance on what configurations that are possible from the



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   presentation description.

   A transport-spec transport option may only contain one of any given
   parameter within it. Parameters MAY be given in any order.
   Additionally, it may only contain the unicast or the multicast
   transport type parameter. Unknown parameters SHALL be ignored. The
   requester need to ensure that the responder understands the
   parameters through the use of feature tags and the Require header.

   Any parameters part of future extensions requires clarification if
   they are safe to ignore in accordance to this specification, or are
   required to be understood. If a parameter is required to be
   understood, then a feature-tag MUST be defined for the functionality
   and used in the Require or Proxy-Require headers.


        The Transport header field is restricted to describing a
        single media stream. (RTSP can also control multiple
        streams as a single entity.) Making it part of RTSP rather
        than relying on a multitude of session description formats
        greatly simplifies designs of firewalls.


   The general syntax for the transport specifier is a list of slash
   separated tokens:
   Value1/Value2/Value3...
   Which for RTP transports take the form:
   RTP/profile/lower-transport.


   The default value for the "lower-transport" parameters is specific to
   the profile. For RTP/AVP, the default is UDP.

   There are two different methods for how to specify where the media
   should be delivered:

        o The presence of this parameter and its values indicates the
          destination address or addresses (host address and port pairs
          for IP flows) necessary for the media transport.

        o The lack of the dest_addr parameter indicates that the server
          SHALL send media to same address for which the RTSP messages
          originates. Does not work for transports requiring explicitly
          given destination ports.

   The choice of method for indicating where the media is to be
   delivered depends on the use case. In many case the only allowed
   method will be to use no explicit address indication and have the



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   server deliver media to the source of the RTSP messages.

   An RTSP proxy will need to take care. If the media is not desired to
   be routed through the proxy, the proxy will need to introduce the
   destination indication.

   Below are the configuration parameters associated with transport:

   General parameters:

        unicast / multicast: This parameter is a mutually exclusive
             indication of whether unicast or multicast delivery will be
             attempted. One of the two values MUST be specified. Clients
             that are capable of handling both unicast and multicast
             transmission needs to indicate such capability by including
             two full transport-specs with separate parameters for each.

        layers: The number of multicast layers to be used for this media
             stream. The layers are sent to consecutive addresses
             starting at the dest_addr address. If the parameter is not
             included, it defaults to a single layer.

        dest_addr: A general destination address parameter that can
             contain one or more address specifications.  Each
             combination of Protocol/Profile/Lower Transport needs to
             have the format and interpretation of its address
             specification defined.  For RTP/AVP/UDP and RTP/AVP/TCP,
             the address specification is a tuple containing a host
             address and port. Note, only a single destination entity
             per transport spec is intended. The usage of multiple
             destination to distribute a single media to multiple
             entities is unspecified.

             The client originating the RTSP request MAY specify the
             destination address of the stream recipient with the host
             address part of the tuple. When the destination address is
             specified, the recipient may be a different party than the
             originator of the request. To avoid becoming the unwitting
             perpetrator of a remote-controlled denial-of-service
             attack, a server MUST perform security checks (see Section
             20.1) and SHOULD log such attempts before allowing the
             client to direct a media stream to a recipient address not
             chosen by the server. Implementations cannot rely on TCP as
             reliable means of client identification. If the server does
             not allow the host address part of the tuple to be set, it
             SHALL return 463 (Destination Prohibited).

             The host address part of the tuple MAY be empty, for



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             example ":58044", in cases when only destination port is
             desired to be specified.

        src_addr: A general source address parameter that can contain
             one or more address specifications.  Each combination of
             Protocol/Profile/Lower Transport needs to have the format
             and interpretation of its address specification defined.
             For RTP/AVP/UDP and RTP/AVP/TCP, the address specification
             is a tuple containing a host address and port.

             This parameter MUST be specified by the server if it
             transmits media packets from another address than the one
             RTSP messages are sent to. This will allow the client to
             verify source address and give it a destination address for
             its RTCP feedback packets if RTP is used. The address or
             addresses indicated in the src_addr parameter SHOULD be
             used both for sending and receiving of the media streams
             data packets. The main reasons are threefold: First,
             indicating the port and source address(s) lets the receiver
             know where from the packets is expected to originate.
             Secondly, traversal of NATs are greatly simplified when
             traffic is flowing symmetrically over a NAT binding.
             Thirdly, certain NAT traversal mechanisms, needs to know to
             which address and port to send so called "binding packets"
             from the receiver to the sender, thus creating a address
             binding in the NAT that the sender to receiver packet flow
             can use.


             This information may also be available through SDP.
             However, since this is more a feature of transport
             than media initialization, the authoritative source
             for this information should be in the SETUP response.

        mode: The mode parameter indicates the methods to be supported
             for this session. Valid values are PLAY and RECORD. If not
             provided, the default is PLAY. The RECORD value was defined
             in RFC 2326 and is in this specification unspecified but
             reserved.

        interleaved: The interleaved parameter implies mixing the media
             stream with the control stream in whatever protocol is
             being used by the control stream, using the mechanism
             defined in Section 12. The argument provides the channel
             number to be used in the $ statement and MUST be present.
             This parameter MAY be specified as a range, e.g.,
             interleaved=4-5 in cases where the transport choice for the
             media stream requires it, e.g. for RTP with RTCP.  The



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             channel number given in the request are only a guidance
             from the client to the server on what channel number(s) to
             use. The server MAY set any valid channel number in the
             response. The declared channel(s) are bi-directional, so
             both end-parties MAY send data on the given channel. One
             example of such usage is the second channel used for RTCP,
             where both server and client sends RTCP packets on the same
             channel.


             This allows RTP/RTCP to be handled similarly to the
             way that it is done with UDP, i.e., one channel for
             RTP and the other for RTCP.

   Multicast-specific:

        ttl: multicast time-to-live. When included in requests the value
             indicate the TTL value that the client desires to use. In
             response the value actually being used is returned. A
             server will need to consider what values that are
             reasonable and also the authority of the user to set this
             value.

   RTP-specific:

   These parameters are MAY only be used if the media transport protocol
   is RTP.

        ssrc: The ssrc parameter, if included in a SETUP response,
             indicates the RTP SSRC [16] value(s) that will be used by
             the media server for RTP packets within the stream. It is
             expressed as an eight digit hexadecimal value.

             The ssrc parameter SHALL NOT be specified in requests. The
             functionality of specifying the ssrc parameter in a SETUP
             request is deprecated as it is incompatible with the
             specification of RTP in RFC 3550 [16]. If the parameter is
             included in the Transport header of a SETUP request, the
             server MAY ignore it, and choose appropriate SSRCs for the
             stream. The server MAY set the ssrc parameter in the
             Transport header of the response.

   The parameters defined below MAY only be used if the media transport
   protocol if the lower-level transport is connection-oriented (such as
   TCP). However, these parameters MUST NOT be used when interleaving
   data over the RTSP control connection.

        setup: Clients use the setup parameter on the Transport line in



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             a SETUP request, to indicate the roles it wishes to play in
             a TCP connection.  This parameter is adapted from [37]. We
             discuss the use of this parameter in RTP/AVP/TCP non-
             interleaved transport in Appendix B.2.2; the discussion
             below is limited to syntactic issues.

             Clients may specify the following values for the setup
             parameter:

             "active": The client will initiate an outgoing connection.

             "passive": The client will accept an incoming connection.

             "actpass": The client is willing to accept an incoming
                  connection or to initiate an outgoing connection.

             If a client does not specify a setup value, the "active"
             value is assumed.

             In response to a client SETUP request where the setup
             parameter is set to "active", a server's 2xx reply MUST
             assign the setup parameter to "passive" on the Transport
             header line.

             In response to a client SETUP request where the setup
             parameter is set to "passive", a server's 2xx reply MUST
             assign the setup parameter to "active" on the Transport
             header line.

             In response to a client SETUP request where the setup
             parameter is set to "actpass", a server's 2xx reply MUST
             assign the setup parameter to "active" or "passive" on the
             Transport header line.

             Note that the "holdconn" value for setup is not defined for
             RTSP use, and MUST NOT appear on a Transport line.

        connection: Clients use the setup parameter on the Transport
             line in a SETUP request, to indicate the SETUP request
             prefers the reuse of an existing connection between client
             and server (in which case the client sets the "connection"
             parameter to "existing"), or that the client requires the
             creation of a new connection between client and server (in
             which cast the client sets the "connection" parameter to
             "new"). Typically, clients use the "new" value for the
             first SETUP request for a URL, and "existing" for
             subsequent SETUP requests for a URL.




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             If a client SETUP request assigns the "new" value to
             "connection", the server response MUST also assign the
             "new" value to "connection" on the Transport line.

             If a client SETUP request assigns the "existing" value to
             "connection", the server response MUST assign a value of
             "existing" or "new" to "connection" on the Transport line,
             at its discretion.

             The default value of "connection" is "existing", for all
             SETUP requests (initial and subsequent).

   The combination of transport protocol, profile and lower transport
   needs to be defined. A number of combinations are defined in the
   appendix B.

   Below is a usage example, showing a client advertising the capability
   to handle multicast or unicast, preferring multicast.  Since this is
   a unicast-only stream, the server responds with the proper transport
   parameters for unicast.


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/2.0
           CSeq: 302
           Transport: RTP/AVP;multicast;mode="PLAY",
               RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
               "192.0.2.5:3457";mode="PLAY"

     S->C: RTSP/2.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Session: 47112344
           Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
           "192.0.2.5:3457";src_addr="192.0.2.224:6256"
           /"192.0.2.224:6257";mode="PLAY"



14.46 Unsupported

   The Unsupported response-header field lists the features not
   supported by the server. In the case where the feature was specified
   via the Proxy-Require field (Section 14.31), if there is a proxy on
   the path between the client and the server, the proxy MUST send a
   response message with a status code of 551 (Option Not Supported).
   The request SHALL NOT be forwarded.

   See Section 14.37 for a usage example.



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14.47 User-Agent

   See [H14.43] for explanation, however the syntax is clarified due to
   an error in RFC 2616. A Client SHOULD include this header in all RTSP
   messages it sends.

14.48 Vary

   See [H14.44]

14.49 Via

   See [H14.45].

14.50 WWW-Authenticate

   See [H14.47].

15 Proxies

   RTSP Proxies are RTSP agents that sit in between a client and a
   server. A proxy can take on both the role as a client and as server
   depending on what it tries to accomplish. Proxies are also introduced
   for several different reasons.

        Caching Proxy: This type of proxy is used to reduce the workload
             on servers and connections. By caching a presentation, both
             description and media streams the proxy can serve a client
             content without requesting it from the server once it has
             been cached and hasn't become stale. See the caching
             Section 16.

        Access Proxy: This type of proxy is used to ensure that a RTSP
             client get access to servers on an external network. Thus
             this proxy is placed on the border between two domains,
             e.g. a private address space and the public internet. The
             proxy performs the necessary translation, usually
             addresses, and often also media stream translation or
             redirection.

        Security Proxy: This type of proxy is used to help facilitate
             security functions around RTSP. For example when having a
             firewalled network, the security proxy request that the
             necessary pinholes in the firewall is opened when a client
             in the protected network want to access media streams on
             the external side. It can also provide network owners with
             a logging and audit point for RTSP sessions, e.g. for
             corporations that tracks or limits their employees access



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             to certain type of content.

   All type of proxies can be used also when using secured communication
   with TLS as RTSP 2.0 allows the client to approve certificates for
   connection establishment from a proxy, see Section 18.3.2. However
   that trust model may not be suitable for all type of deployment, and
   instead secured sessions do by-pass of the proxies.

   Access proxies SHOULD NOT be used in equipment like NATs and
   firewalls that aren't expected to be regularly maintained, like home
   or small office equipment. In these cases it is better to use the NAT
   traversal procedures defined for RTSP 2.0 [38].  The reason for these
   recommendations is that any extensions of RTSP resulting in new media
   transport protocols or profiles, new parameters etc may fail in a
   proxy that isn't maintained. Thus resulting in blocking further
   development of RTSP and its usage.

   The existence of proxies must always be considered when developing
   new RTSP extensions. There must be definition of how proxies may
   handle the extension, if it is required to understand it, thus
   requiring a feature-tag to be used in the Proxy-Require header.

16 Caching

   In HTTP, response-request pairs are cached. RTSP differs
   significantly in that respect. Responses are not cacheable, with the
   exception of the presentation description returned by DESCRIBE.
   (Since the responses for anything but DESCRIBE and GET_PARAMETER do
   not return any data, caching is not really an issue for these
   requests.) However, it is desirable for the continuous media data,
   typically delivered out-of-band with respect to RTSP, to be cached,
   as well as the session description.

   On receiving a SETUP or PLAY request, a proxy ascertains whether it
   has an up-to-date copy of the continuous media content and its
   description. It can determine whether the copy is up-to-date by
   issuing a SETUP or DESCRIBE request, respectively, and comparing the
   Last-Modified header with that of the cached copy. If the copy is not
   up-to-date, it modifies the SETUP transport parameters as appropriate
   and forwards the request to the origin server. Subsequent control
   commands such as PLAY or PAUSE then pass the proxy unmodified. The
   proxy delivers the continuous media data to the client, while
   possibly making a local copy for later reuse. The exact behavior
   allowed to the cache is given by the cache-response directives
   described in Section 14.10. A cache MUST answer any DESCRIBE requests
   if it is currently serving the stream to the requestor, as it is
   possible that low-level details of the stream description may have
   changed on the origin-server.



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   Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
   through" variety. Rather than retrieving the whole resource from the
   origin server, the cache simply copies the streaming data as it
   passes by on its way to the client. Thus, it does not introduce
   additional latency.

   To the client, an RTSP proxy cache appears like a regular media
   server, to the media origin server like a client. Just as an HTTP
   cache has to store the content type, content language, and so on for
   the objects it caches, a media cache has to store the presentation
   description. Typically, a cache eliminates all transport-references
   (that is, e.g. multicast information) from the presentation
   description, since these are independent of the data delivery from
   the cache to the client. Information on the encodings remains the
   same. If the cache is able to translate the cached media data, it
   would create a new presentation description with all the encoding
   possibilities it can offer.

17 Examples

   This section contains several different examples trying to illustrate
   possible ways of using RTSP. The examples can also help with the
   understanding of how functions of RTSP work. However remember that
   this is examples and the normative and syntax description in the
   other sections takes precedence. Please also note that many of the
   example contain syntax illegal line breaks to accommodate the
   formatting restriction that the RFC series impose.

17.1 Media on Demand (Unicast)

   The is an example of media on demand streaming of a media stored in a
   container file. For purposes of this example, a container file is a
   storage entity in which multiple continuous media types pertaining to
   the same end-user presentation are present. In effect, the container
   file represents an RTSP presentation, with each of its components
   being RTSP controlled media streams. Container files are a widely
   used means to store such presentations. While the components are
   transported as independent streams, it is desirable to maintain a
   common context for those streams at the server end.


        This enables the server to keep a single storage handle
        open easily. It also allows treating all the streams
        equally in case of any prioritization of streams by the
        server.

   It is also possible that the presentation author may wish to prevent
   selective retrieval of the streams by the client in order to preserve



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   the artistic effect of the combined media presentation.  Similarly,
   in such a tightly bound presentation, it is desirable to be able to
   control all the streams via a single control message using an
   aggregate URI.

   The following is an example of using a single RTSP session to control
   multiple streams. It also illustrates the use of aggregate URIs. In a
   container file it is also desirable to not write any URI parts which
   is not kept, when the container is distributed, like the host and
   most of the path element. Therefore this example also uses the "*"
   and relative URI in the delivered SDP.

   Client C requests a presentation from media server M. The movie is
   stored in a container file. The client has obtained an RTSP URI to
   the container file.


   C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2

   M->C: RTSP/2.0 200 OK
         CSeq: 1
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 257
         Content-Base: rtsp://example.com/twister.3gp/
         Expires: 24 Jan 1997 15:35:06 GMT

         v=0
         o=- 2890844256 2890842807 IN IP4 192.0.2.5
         s=RTSP Session
         i=An Example of RTSP Session Usage
         e=adm@example.com
         a=control: *
         a=range: npt=0-0:10:34.10
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control: trackID=1
         m=video 0 RTP/AVP 26
         a=control: trackID=4

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"



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   M->C: RTSP/2.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001;
                    src_addr="192.0.2.5:9000"/"192.0.2.5:9001"
                    ssrc=93CB001E
         Session: 12345678
         Expires: 24 Jan 1997 15:35:12 GMT
         Date: 23 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT

   C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Require: play.basic
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003;
                    src_addr="192.0.2.5:9002"/"192.0.2.5:9003";
                    ssrc=A813FC13
         Session: 12345678
         Expires: 24 Jan 1997 15:35:13 GMT
         Date: 23 Jan 1997 15:35:13 GMT
         Accept-Range: NPT

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 4
         User-Agent: PhonyClient/1.2
         Range: npt=0-10, npt=30-
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 4
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:14 GMT
         Session: 12345678
         Range: npt=0-10, npt=30-623.10
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12345;rtptime=3450012,
           url="rtsp://example.com/twister.3gp/trackID=1";
            ssrc=4F312DD8:seq=54321;rtptime=2876889

   C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 5



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         User-Agent: PhonyClient/1.2
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 5
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10

   C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
         CSeq: 6
         User-Agent: PhonyClient/1.2
         Range: npt=34.57-623.10
         Session: 12345678

   M->C: RTSP/2.0 200 OK
         CSeq: 6
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:01 GMT
         Session: 12345678
         Range: npt=34.57-623.10
         RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
            ssrc=0D12F123:seq=12555;rtptime=6330012,
           url="rtsp://example.com/twister.3gp/trackID=1"
            ssrc=4F312DD8:seq=55021;rtptime=3132889




17.2 Media on Demand (Unicast)

   An alternative example of media on demand with a bit more tweaks is
   the following. Client C requests a movie distributed from two
   different media servers A (audio.example.com ) and V
   (video.example.com ). The media description is stored on a web server
   W. The media description contains descriptions of the presentation
   and all its streams, including the codecs that are available, dynamic
   RTP payload types, the protocol stack, and content information such
   as language or copyright restrictions. It may also give an indication
   about the timeline of the movie.

   In this example, the client is only interested in the last part of
   the movie.


   C->W: GET /twister.sdp HTTP/1.1
         Host: www.example.com



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         Accept: application/sdp

   W->C: HTTP/1.0 200 OK
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 264
         Expires: 23 Jan 1998 15:35:06 GMT

         v=0
         o=- 2890844526 2890842807 IN IP4 192.0.2.5
         s=RTSP Session
         e=adm@example.com
         a=range:npt=0-1:49:34
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://audio.example.com/twister/audio.en
         m=video 0 RTP/AVP 31
         a=control:rtsp://video.example.com/twister/video

   C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
                    RTP/AVP/TCP;unicast;interleaved=0-1

   A->C: RTSP/2.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057";
                    src_addr="192.0.2.5:5000"/"192.0.2.5:5001"
         Date: 23 Jan 1997 15:35:12 GMT
         Server: PhonyServer/1.0
         Expires: 24 Jan 1997 15:35:12 GMT
         Cache-Control: public
         Accept-Ranges: NPT, SMPTE

   C->V: SETUP rtsp://video.example.com/twister/video RTSP/2.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;dest_addr=":3058"/":3059",
                    RTP/AVP/TCP;unicast;interleaved=0-1

   V->C: RTSP/2.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;dest_addr=":3058"/":3059";
            src_addr="192.0.2.5:5002"/"192.0.2.5:5003"
         Date: 23 Jan 1997 15:35:12 GMT



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         Server: PhonyServer/1.0
         Cache-Control: public
         Expires: 24 Jan 1997 15:35:12 GMT
         Accept-Ranges: NPT, SMPTE

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/2.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-1:49:23
         RTP-Info: url="rtsp://video.example.com/twister/video"
                   ssrc=A17E189D:seq=12312232;rtptime=78712811
         Server: PhonyServer/2.0
         Date: 23 Jan 1997 15:35:13 GMT

   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/2.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 12345678
         Range: smpte=0:10:00-

   A->C: RTSP/2.0 200 OK
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-1:49:23
         RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
                   ssrc=3D124F01:seq=876655;rtptime=1032181
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:13 GMT



   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 12345678

   A->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:36:52 GMT

   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/2.0



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         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 23456789

   V->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/2.0
         Date: 23 Jan 1997 15:36:52 GMT




   Even though the audio and video track are on two different servers,
   may start at slightly different times, and may drift with respect to
   each other, the client can perform initial synchronize of the two
   media using RTP-Info and Range received in the PLAY responses. If the
   two servers are time synchronized the RTCP packets can also be used
   to maintain synchronization.

17.3 Single Stream Container Files

   Some RTSP servers may treat all files as though they are "container
   files", yet other servers may not support such a concept. Because of
   this, clients needs to use the rules set forth in the session
   description for Request-URIs, rather than assuming that a consistent
   URI may always be used throughout. Below are an example of how a
   multi-stream server might expect a single-stream file to be served:


   C->S: DESCRIBE rtsp://foo.com/test.wav RTSP/2.0
         Accept: application/x-rtsp-mh, application/sdp
         CSeq: 1
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 1
         Content-base: rtsp://foo.com/test.wav/
         Content-type: application/sdp
         Content-length: 148
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Expires: 23 Jan 1997 17:00:00 GMT

         v=0
         o=- 872653257 872653257 IN IP4 192.0.2.5
         s=mu-law wave file
         i=audio test
         t=0 0



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         a=control: *
         m=audio 0 RTP/AVP 0
         a=control:streamid=0

   C->S: SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/2.0
         Transport: RTP/AVP/UDP;unicast;
            dest_addr=":6970"/":6971";mode="PLAY"
         CSeq: 2
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         Transport: RTP/AVP/UDP;unicast;dest_addr=":6970"/":6971";
             src_addr="192.0.2.5:6970"/"192.0.2.5:6971";
             mode="PLAY";ssrc=EAB98712
         CSeq: 2
         Session: 2034820394
         Expires: 23 Jan 1997 16:00:00 GMT
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:07 GMT

   C->S: PLAY rtsp://foo.com/test.wav/ RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 2034820394

   S->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:08 GMT
         Session: 2034820394
         Range: npt=0-600
         RTP-Info: url="rtsp://foo.com/test.wav/streamid=0"
            ssrc=0D12F123:seq=981888;rtptime=3781123



   Note the different URI in the SETUP command, and then the switch back
   to the aggregate URI in the PLAY command.  This makes complete sense
   when there are multiple streams with aggregate control, but is less
   than intuitive in the special case where the number of streams is
   one. However the server has declared that the aggregated control URI
   in the SDP and therefore this is legal.

   In this case, it is also required that servers accept implementations
   that use the non-aggregated interpretation and use the individual
   media URI, like this:





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   C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2



17.4 Live Media Presentation Using Multicast

   The media server M chooses the multicast address and port. Here, it
   is assumed that the web server only contains a pointer to the full
   description, while the media server M maintains the full description.


   C->W: GET /sessions.html HTTP/2.0
         Host: www.example.com

   W->C: HTTP/2.0 200 OK
         Content-Type: text/html

         <html>
           ...
           <href "Stremed Live Music performance"
              src="rtsp://live.example.com/concert/audio">
           ...
         </html>

   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 1
         Supported: play.basic, play.scale

   M->C: RTSP/2.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 182
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Supported: play.basic

         v=0
         o=- 2890844526 2890842807 IN IP4 192.0.2.5
         s=RTSP Session
         m=audio 3456 RTP/AVP 0
         c=IN IP4 224.2.0.1/16
         a=control: rtsp://live.example.com/concert/audio
         a=range:npt=0-

   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 2



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         Transport: RTP/AVP;multicast

   M->C: RTSP/2.0 200 OK
         CSeq: 2
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:06 GMT
         Transport: RTP/AVP;multicast;dest_addr="224.2.0.1:3456"/"
                    224.2.0.1:3457";ttl=16
         Session: 0456804596
         Accept-Ranges: NPT, UTC

   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/2.0
         CSeq: 3
         Session: 0456804596

   M->C: RTSP/2.0 200 OK
         CSeq: 3
         Server: PhonyServer/1.0
         Date: 23 Jan 1997 15:35:07 GMT
         Session: 0456804596
         Range:npt=1256-
         RTP-Info: url="rtsp://live.example.com/concert/audio"
                   ssrc=0D12F123:seq=1473; rtptime=80000



17.5 Capability Negotiation

   This examples illustrate how the client and server determines their
   capability to support a special feature, in this case "play.scale".
   The server, through the clients request and the included Supported
   header, learns the client supports RTSP 2.0, and also supports the
   playback time scaling feature of RTSP. The server's response contains
   the following feature related information to the client; it supports
   the basic playback (play.basic), the extended functionality of time
   scaling of content (play.scale), and one "example.com" proprietary
   feature (com.example.flight). The client also learns the methods
   supported (Public header) by the server for the indicated resource.


   C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/2.0
         CSeq: 1
         Supported: play.basic, play.scale
         User-Agent: PhonyClient/1.2

   S->C: RTSP/2.0 200 OK
         CSeq: 1
         Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN



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         Server: PhonyServer/2.0
         Supported: play.basic, play.scale, com.example.flight



   When the client sends its SETUP request it tells the server that it
   is requires support of the play.scale feature for this session by
   including the Require header.


   C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/2.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
                    RTP/AVP/TCP;unicast;interleaved=0-1
         Require: play.scale

   S->C: RTSP/2.0 200 OK
         CSeq: 3
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057";
            src_addr="192.0.2.5:5000"/"192.0.2.5:5001"
         Server: PhonyServer/2.0
         Accept-Ranges: NPT, SMPTE



18 Security Framework

   The RTSP security framework consists of two high level components:
   the pure authentication mechanisms based on HTTP authentication, and
   the transport protection based on TLS, which is independent of RTSP.
   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security for HTTP is re-used to a large extent.

18.1 RTSP and HTTP Authentication

   RTSP and HTTP share common authentication schemes, and thus follow
   the same usage guidelines as specified in [10] and also in [H15].
   Servers SHOULD implement both basic and digest [10] authentication.

   It should be stressed that using the HTTP authentication alone does
   not provide full control message security. Therefore, in environments
   requiring tighter security for the control messages, TLS SHOULD be
   used, see Section 18.2.

18.2 RTSP over TLS




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   RTSP SHALL follow the same guidelines with regards to TLS [9] usage
   as specified for HTTP, see [17]. RTSP over TLS is separated from
   unsecured RTSP both on URI level and port level. Instead of using the
   "rtsp" scheme identifier in the URI, the "rtsps" scheme identifier
   MUST be used to signal RTSP over TLS. If no port is given in a URI
   with the "rtsps" scheme, port 322 SHALL be used for TLS over TCP/IP.

   When a client tries to setup an insecure channel to the server (using
   the "rtsp" URI), and the policy for the resource requires a secure
   channel, the server SHALL redirect the client to the secure service
   by sending a 301 redirect response code together with the correct
   Location URI (using the "rtsps" scheme). A user or client MAY upgrade
   a non secured URI to a secured by changing the scheme from "rtsp" to
   "rtsps". A server implementing support for "rtsps" SHALL allow this.

   It should be noted that TLS allows for mutual authentication (when
   using both server and client certificates). Still, one of the more
   common way TLS is used is to only provide server side authentication
   (often to avoid client certificates). TLS is then used in addition to
   HTTP authentication, providing transport security and server
   authentication, while HTTP Authentication is used to authenticate the
   client.

   RTSP includes the possibility to keep a TCP session up between the
   client and server, throughout the RTSP session lifetime. It may be
   convenient to keep the TCP session, not only to save the extra setup
   time for TCP, but also the extra setup time for TLS (even if TLS uses
   the resume function, there will be almost two extra roundtrips).
   Still, when TLS is used, such behavior introduces extra active state
   in the server, not only for TCP and RTSP, but also for TLS. This may
   increase the vulnerability to DoS attacks.

   In addition to these recommendations, Section 18.3 gives further
   recommendations of TLS usage with proxies.

18.3 Security and Proxies

   The nature of a proxy is often to act as a "man-in-the-middle", while
   security is often about preventing the existence of a "man-in-the-
   middle". This section provides the clients with the possibility to
   use proxies even when applying secure transports (TLS). The client
   needs to select between using the procedure specified below or using
   a TLS connection directly (by-passing any proxies) to the server. The
   choice may be dependent on policies.

   There are basically two categories of proxies, the transparent
   proxies (of which the client is not aware) and the non-transparent
   proxies (of which the client is aware). An infrastructure based on



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   proxies requires that the trust model is such that both client and
   servers can trust the proxies to handle the RTSP messages correctly.
   To be able to trust a proxy, the client and server also needs to be
   aware of the proxy. Hence, transparent proxies cannot generally be
   seen as trusted and will not work well with security (unless they
   work only at transport layer). In the rest of this section any
   reference to proxy will be to a non-transparent proxy, which inspects
   or manipulate the RTSP messages.

   HTTP Authentication is built on the assumption of proxies and can
   provide user-proxy authentication and proxy-proxy/server
   authentication in addition to the client-server authentication.

   When TLS is applied and a proxy is used, the client will connect to
   the proxy's address when connecting to any RTSP server. This implies
   that for TLS, the client will authenticate the proxy server and not
   the end server. Note that, when the client checks the server
   certificate in TLS, it MUST check the proxy's identity (URI or
   possibly other known identity) against the proxy's identity as
   presented in the proxy's Certificate message.

   The problem is that for a proxy accepted by the client, the proxy
   needs to be provided information on which grounds it should accept
   the next-hop certificate. Both the proxy and the user may have rules
   for this, and the user have the possibility to select the desired
   behavior. To handle this case, the Accept-Credentials header (See
   Section 14.2) is used, where the client can force the proxy/proxies
   to relay back the certificates used by any intermediate proxies as
   well as the server. Given the assumption that the proxies are viewed
   as trusted, it gives the user a possibility to enforce policies to
   each trusted proxy of whether it should accept the next entity in the
   chain.

   A proxy MUST use TLS for the next hop if the RTSP request includes a
   "rtsps" URI. TLS MAY be applied on intermediate links (e.g. between
   client and proxy, or between proxy and proxy), even if the resource
   and the end server does not require to use it.

18.3.1 Accept-Credentials

   The Accept-Credentials header can be used by the client to distribute
   simple authorization policies to intermediate proxies. The client
   includes the Accept-Credentials header to dictate how the proxy
   treats the server/next proxy certificate. There are currently three
   methods defined:

        Any, which means that the proxy (or proxies) SHALL accept
             whatever certificate presented. This is of course not a



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             recommended option to use, but may be useful in certain
             circumstances (such as testing).

        Proxy, which means that the proxy (or proxies) MUST use its own
             policies to validate the certificate and decide whether to
             accept it or not. This is convenient in cases where the
             user has a strong trust relation with the proxy. Reason why
             a strong trust relation may exist are; personal/company
             proxy, proxy has a out-of-band policy configuration
             mechanism.

        User, which means that the proxy (or proxies) MUST send
             credential information about the next hop to the client for
             authorization. The client can then decide whether the proxy
             should accept the certificate or not. See section 18.3.2
             for further details.

   If the Accept-Credentials header is not included in the RTSP request
   from the client, then the "Proxy" method SHALL be used as default. If
   an other method than the "Proxy" is to be used, then the Accept-
   Credentials header SHALL be included in all of the RTSP request from
   the client. This is because it cannot be assumed that the proxy
   always keeps the TLS state or the users previously preference between
   different RTSP messages (in particular if the time interval between
   the messages is long).

   With the "Any" and "Proxy" methods the proxy will apply the policy as
   defined for respectively method. If the policy do not accept the
   credentials of the next hop, the entity SHALL respond with a message
   using status code 471 (Connection Credentials not accepted).

   An RTSP request in the direction server to client MUST NOT include
   the Accept-Credential header. As for the non-secured communication,
   the possibility for these request depends on the presence of a client
   established connection.  However if the server to client request is
   in relation to a session established over a TLS secured channel, if
   MUST be sent in a TLS secured connection. That secured connection
   MUST also be the one used by the last client to server request. If no
   such transport connection exist at the time when the server desire to
   send the request, it silently fails.

   Further policies MAY be defined and registered, but should be done so
   with caution.

18.3.2 User approved TLS procedure

   For the "User" method each proxy MUST perform the the following
   procedure for each RTSP request:



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        o Setup the TLS session to the next hop if not already present
          (i.e. run the TLS handshake, but do not send the RTSP
          request).

        o Extract the peer certificate for the TLS session.

        o Check if a matching identity and hash of the peer certificate
          is present in the Accept-Credentials header.  If present, send
          the message to the next hop, and conclude these procedures. If
          not, go to the next step.

        o The proxy responds to the RTSP request with a 470 or 407
          response code. The 407 response code MAY be used when the
          proxy requires both user and connection authorization from
          user or client. In this message the proxy SHALL include a
          Connection-Credentials header, see section 14.12 with the next
          hop's identity and certificate.

   The client MUST upon receiving a 470 or 407 response with
   Connection-Credentials header take the decision on whether to accept
   the certificate or not (if it cannot do so, the user SHOULD be
   consulted). If the certificate is accepted, the client has to again
   send the RTSP request. In that request the client has to include the
   Accept-Credentials header including the hash over the DER encoded
   certificate for all trusted proxies in the chain.


   Example:
   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
                    "192.0.2.5:4589"

   P->C: RTSP/2.0 470 Connection Authorization Required
         CSeq: 2
         Connection-Credentials: "rtsps://test.example.org";
         MIIDNTCCAp...

   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
         "192.0.2.5:4589"
         Accept-Credentials: User "rtsps://test.example.org" ;
         dPYD 7txp oGTb AqZZ QJ+v aeOk yH4= ...

   C->P: SETUP rtsps://test.example.org/secret/audio RTSP/2.0
         CSeq: 2
         Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/



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         "192.0.2.5:4589"
         Via: RTSP/2.0 proxy.example.org
         Accept-Credentials: User "rtsps://test.example.org" ;
         dPYD 7txp oGTb AqZZ QJ+v aeOk yH4= ...



   One implication of this process is that the connection for secured
   RTSP messages may take significantly more round-trip times for the
   first message. An complete extra message exchange between the proxy
   connecting to the next hop and the client results because of the
   process for approval for each hop. However after the first message
   exchange the remaining message should not be delayed, if each message
   contains the chain of proxies that the requestor accepts. The
   procedure of including the credentials in each request rather than
   building state in each proxy, avoids the need for revocation
   procedures.

19 Syntax

   The RTSP syntax is described in an Augmented Backus-Naur Form (ABNF)
   as defined in RFC 4234 [4]. It uses the basic definitions present in
   RFC 4234.

   Please note that ABNF strings, e.g. "Accept", are case insensitive as
   specified in section 2.3 of RFC 4234.

19.1 Base Syntax

   RTSP header field values can be folded onto multiple lines if the
   continuation line begins with a space or horizontal tab. All linear
   white space, including folding, has the same semantics as SP. A
   recipient MAY replace any linear white space with a single SP before
   interpreting the field value or forwarding the message downstream.
   This is intended to behave exactly as HTTP/1.1 as described in RFC
   2616 [3]. The SWS construct is used when linear white space is
   optional, generally between tokens and separators.

   To separate the header name from the rest of value, a colon is used,
   which, by the above rule, allows whitespace before, but no line
   break, and whitespace after, including a linebreak. The HCOLON
   defines this construct.


   OCTET      =  %x00-FF ; any 8-bit sequence of data
   CHAR       =  %x01-7F ; any US-ASCII character (octets 1 - 127)
   UPALPHA    =  %x41-5A ; any US-ASCII uppercase letter "A".."Z"
   LOALPHA    =  %x61-7A ;any US-ASCII lowercase letter "a".."z"



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   ALPHA      =  UPALPHA / LOALPHA
   DIGIT      =  %x30-39 ; any US-ASCII digit "0".."9"
   CTL        =  %x00-1F / %x7F ; any US-ASCII control character
                 ; (octets 0 - 31) and DEL (127)
   CR         =  %x0D ; US-ASCII CR, carriage return (13
   LF         =  %x0A ; US-ASCII LF, linefeed (10)
   SP         =  %x20 ; US-ASCII SP, space (32)
   HT         =  %x09 ; US-ASCII HT, horizontal-tab (9)
   DQ         =  %x22 ; US-ASCII double-quote mark (34)
   BACKSLASH  =  %x5C ; US-ASCII backslash (92)
   CRLF       =  CR LF



   LWS            =  [CRLF] 1*( SP / HT )
   SWS            =  [LWS] ; sep whitespace
   HCOLON         =  *( SP / HT ) ":" SWS
   TEXT           =  %x20-7D / %x80-FF ; any OCTET except CTLs
   tspecials      =  "(" / ")" / "<" / ">" / "@"
                  /  "," / ";" / ":" / BACKSLASH / DQ
                  /  "/" / "[" / "]" / "?" / "="
                  /  "{" / "}" / SP / HT
   token          =  1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
                  /  %x41-5A / %x5E-7A / %x7C / %x7E)
                     ; 1*<any CHAR except CTLs or tspecials>
   quoted-string  =  ( DQ *qdtext DQ )
   qdtext         =  %x20-21 / %x23-7D / %x80-FF ; any TEXT except <">
   quoted-pair    =  BACKSLASH CHAR
   ctext          =  %x20-27 / %x2A-7D
                  /  %x80-FF ; any OCTET except CTLs, "(" and ")"
   generic-param  =  token [ EQUAL gen-value ]
   gen-value      =  token / host / quoted-string



   safe             =  "$" / "-" / "_" / "." / "+"
   extra            =  "!" / "*" / "'" / "(" / ")" / ","
   rtsp-extra       =  "!" / "*" / "'" / "(" / ")"
   HEX              =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
                    /  "a" / "b" / "c" / "d" / "e" / "f"
   LHEX             =  DIGIT / %x61-66 ;lowercase a-f
   reserved         =  ";" / "/" / "?" / ":" / "@" / "&" / "="
   unreserved       =  ALPHA / DIGIT / safe / extra
   rtsp-unreserved  =  ALPHA / DIGIT / safe / rtsp-extra
   base64           =  *base64-unit [base64-pad]
   base64-unit      =  4base64-char
   base64-pad       =  (2base64-char "==") / (3base64-char "=")
   base64-char      =  ALPHA / DIGIT / "+" / "/"



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   SLASH          =  SWS "/" SWS ; slash
   EQUAL          =  SWS "=" SWS ; equal
   LPAREN         =  SWS "(" SWS ; left parenthesis
   RPAREN         =  SWS ")" SWS ; right parenthesis
   COMMA          =  SWS "," SWS ; comma
   SEMI           =  SWS ";" SWS ; semicolon
   COLON          =  SWS ":" SWS ; colon
   LDQUOT         =  SWS DQ ; open double quotation mark
   RDQUOT         =  DQ SWS ; close double quotation mark
   RAQUOT         =  ">" SWS ; right angle quote
   LAQUOT         =  SWS "<" ; left angle quote
   TEXT-UTF8char  =  %x21-7E / UTF8-NONASCII
   UTF8-NONASCII  =  %xC0-DF 1UTF8-CONT
                  /  %xE0-EF 2UTF8-CONT
                  /  %xF0-F7 3UTF8-CONT
                  /  %xF8-FB 4UTF8-CONT
                  /  %xFC-FD 5UTF8-CONT
   UTF8-CONT      =  %x80-BF


19.2 RTSP Protocol Definition

19.2.1 Generic Protocol elements


   RTSP-IRI        =  schemes ":" IRI-rest
   IRI-rest        =  ihier-part [ "?" iquery ] [ "#" ifragment ]
   ihier-part      =  "//" iauthority ipath-abempty
   RTSP-IRI-ref    =  RTSP-IRI / irelative-ref
   irelative-ref   =  irelative-part [ "?" iquery ] [ "#" ifragment ]
   irelative-part  =  "//" iauthority ipath-abempty
                      / ipath-absolute
                      / ipath-noscheme
                      / ipath-empty
   iauthority      =  < As defined in RFC 3987 [7]>
   ipath           =  ipath-abempty ; begins with "/" or is empty
                      / ipath-absolute ; begins with "/" but not "//"
                      / ipath-noscheme ; begins with a non-colon segment
                      / ipath-rootless ; begins with a segment
                      / ipath-empty ; zero characters
   ipath-abempty   =  *( "/" isegment )
   ipath-absolute  =  "/" [ isegment-nz *( "/" isegment ) ]
   ipath-noscheme  =  isegment-nz-nc *( "/" isegment )
   ipath-rootless  =  isegment-nz *( "/" isegment )
   ipath-empty     =  0<ipchar>
   isegment        =  *ipchar [";" *ipchar]
   isegment-nz     =  1*ipchar [";" *ipchar]
                      / ";" *ipchar



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   isegment-nz-nc  =  (1*ipchar-nc [";" *ipchar-nc])
                      / ";" *ipchar-nc
                      ; non-zero-length segment without any colon ":"
   ipchar          =  iunreserved / pct-encoded / sub-delims / ":" / "@"
   ipchar-nc       =  iunreserved / pct-encoded / sub-delims / "@"
   iquery          =  < As defined in RFC 3987 [7]>
   ifragment       =  < As defined in RFC 3987 [7]>
   iunreserved     =  < As defined in RFC 3987 [7]>
   pct-encoded     =  < As defined in RFC 3987 [7]>



   RTSP-URI       =  schemes ":" URI-rest
   RTSP-URI-Ref   =  RTSP-URI / RTSP-Relative
   schemes        =  "rtsp" / "rtsps" / scheme
   scheme         =  < As defined in RFC 3986 [8]>
   URI-rest       =  hier-part [ "?" query ]
   hier-part      =  "//" authority path-abempty
   RTSP-Relative  =  relative-part [ "?" query ]
   relative-part  =  "//" authority path-abempty
                     / path-absolute
                     / path-noscheme
                     / path-empty
   authority      =  < As defined in RFC 3986 [8]>
   query          =  < As defined in RFC 3986 [8]>
   path           =  path-abempty ; begins with "/" or is empty
                     / path-absolute ; begins with "/" but not "//"
                     / path-noscheme ; begins with a non-colon segment
                     / path-rootless ; begins with a segment
                     / path-empty ; zero characters
   path-abempty   =  *( "/" segment )
   path-absolute  =  "/" [ segment-nz *( "/" segment ) ]
   path-noscheme  =  segment-nz-nc *( "/" segment )
   path-rootless  =  segment-nz *( "/" segment )
   path-empty     =  0<pchar>
   segment        =  *pchar [";" *pchar]
   segment-nz     =  ( 1*pchar [";" *pchar]) / (";" *pchar)
   segment-nz-nc  =  ( 1*pchar-nc [";" *pchar-nc]) / (";" *pchar-nc)
                     ; non-zero-length segment without any colon ":"
   pchar          =  unreserved / pct-encoded / sub-delims / ":" / "@"
   pchar-nc       =  unreserved / pct-encoded / sub-delims / "@"
   sub-delims     =  "!" / "$" / "&" / "'" / "(" / ")"
                     / "*" / "+" / "," / "="



   smpte-range           =  smpte-type "=" smpte-range-spec
                            ;Section 3.4



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   smpte-range-spec      =  ( smpte-time "-" [ smpte-time ] )
                         /  ( "-" smpte-time )
   smpte-type            =  "smpte" / "smpte-30-drop"
                         /  "smpte-25" / smpte-type-extension
                            ; other timecodes may be added
   smpte-type-extension  =  token
   smpte-time            =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
                            [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]



   npt-range       =  "npt=" npt-range-spec ; Section 3.5
   npt-range-spec  =  ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
   npt-time        =  "now" / npt-sec / npt-hhmmss
   npt-sec         =  1*DIGIT [ "." *DIGIT ]
   npt-hhmmss      =  npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
   npt-hh          =  1*DIGIT ; any positive number
   npt-mm          =  1*2DIGIT ; 0-59
   npt-ss          =  1*2DIGIT ; 0-59



   utc-range       =  "clock=" utc-range-spec ; Section 3.6
   utc-range-spec  =  ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
   utc-time        =  utc-date "T" utc-clock "Z"
   utc-date        =  8DIGIT ; < YYYYMMDD >
   utc-clock       =  6DIGIT [ "." fraction ]; < HHMMSS.fraction >
   fraction        =  1*DIGIT



   feature-tag       =  token
   session-id        =  8*( ALPHA / DIGIT / safe )
   extension-header  =  header-name HCOLON header-value
   header-name       =  token
   header-value      =  *(TEXT-UTF8char / UTF8-CONT / LWS)


19.2.2 Message Syntax



        RTSP-message  =   Request / Response ; RTSP/2.0 messages
        Request       =   Request-Line        ; Section 6.1
                      *(  general-header      ; Section 5
                      /   request-header      ; Section 6.2
                      /   entity-header )     ; Section 8.1
                          CRLF



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                          [ message-body ]                      ;
        Section 4.3      Response      =   Status-Line         ; Section
        7.1                    *(  general-header      ; Section
        5                      /   response-header     ; Section
        7.2                    /   entity-header )     ; Section
        8.1
        CRLF                                                     [
        message-body ]    ; Section 4.3



   Request-Line  =  Method SP Request-URI SP RTSP-Version CRLF
   Status-Line   =  RTSP-Version SP Status-Code SP Reason-Phrase CRLF



   Method            =  "DESCRIBE"        ; Section 11.2
                     /  "GET_PARAMETER"   ; Section 11.7
                     /  "OPTIONS"         ; Section 11.1
                     /  "PAUSE"           ; Section 11.5
                     /  "PLAY"            ; Section 11.4
                     /  "REDIRECT"        ; Section 11.9
                     /  "SETUP"           ; Section 11.3
                     /  "SET_PARAMETER"   ; Section 11.8
                     /  "TEARDOWN"        ; Section 11.6
                     /  extension-method
   extension-method  =  token



   Request-URI             =  "*" / RTSP-URI
   RTSP-Version            =  "RTSP/" 1*DIGIT "." 1*DIGIT
   message-body = 1*OCTET



   Status-Code     =  "100" ; Continue
                   /  "200" ; OK
                   /  "300" ; Multiple Choices
                   /  "301" ; Moved Permanently
                   /  "302" ; Found
                   /  "303" ; See Other
                   /  "304" ; Not Modified
                   /  "305" ; Use Proxy
                   /  "400" ; Bad Request
                   /  "401" ; Unauthorized
                   /  "402" ; Payment Required
                   /  "403" ; Forbidden



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                   /  "404" ; Not Found
                   /  "405" ; Method Not Allowed
                   /  "406" ; Not Acceptable
                   /  "407" ; Proxy Authentication Required
                   /  "408" ; Request Time-out
                   /  "410" ; Gone
                   /  "411" ; Length Required
                   /  "412" ; Precondition Failed
                   /  "413" ; Request Entity Too Large
                   /  "414" ; Request-URI Too Large
                   /  "415" ; Unsupported Media Type
                   /  "451" ; Parameter Not Understood
                   /  "452" ; reserved
                   /  "453" ; Not Enough Bandwidth
                   /  "454" ; Session Not Found
                   /  "455" ; Method Not Valid in This State
                   /  "456" ; Header Field Not Valid for Resource
                   /  "457" ; Invalid Range
                   /  "458" ; Parameter Is Read-Only
                   /  "459" ; Aggregate operation not allowed
                   /  "460" ; Only aggregate operation allowed
                   /  "461" ; Unsupported Transport
                   /  "462" ; Destination Unreachable
                   /  "463" ; Destination Prohibited
                   /  "464" ; Data Transport Not Ready Yet
                   /  "470" ; Connection Authorization Required
                   /  "471" ; Connection Credentials not accepted
                   /  "500" ; Internal Server Error
                   /  "501" ; Not Implemented
                   /  "502" ; Bad Gateway
                   /  "503" ; Service Unavailable
                   /  "504" ; Gateway Time-out
                   /  "505" ; RTSP Version not supported
                   /  "551" ; Option not supported
                   /  extension-code
   extension-code  =  3DIGIT
   Reason-Phrase   =  *TEXT



   general-header  =  Cache-Control     ; Section 14.10
                   /  Connection        ; Section 14.11
                   /  CSeq              ; Section 14.19
                   /  Date              ; Section 14.20
                   /  Proxy-Supported   ; Section 14.32
                   /  Supported         ; Section 14.43
                   /  Timestamp         ; Section 14.44
                   /  Via               ; Section 14.49



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                   /  extension-header



   request-header  =  Accept              ; Section 14.1 and [H14.1]
                   /  Accept-Credentials  ; Section 14.2
                   /  Accept-Encoding     ; Section 14.3 and [H14.3]
                   /  Accept-Language     ; Section 14.4 and [H14.4]
                   /  Authorization       ; Section 14.7 and [H14.8]
                   /  Bandwidth           ; Section 14.8
                   /  Blocksize           ; Section 14.9
                   /  From                ; Section 14.23
                   /  If-Match            ; Section 14.24
                   /  If-Modified-Since   ; Section 14.25 and [H14.25]
                   /  If-None-Match       ; Section 14.26
                   /  Proxy-Require       ; Section 14.31
                   /  Range               ; Section 14.34
                   /  Referer             ; Section 14.35
                   /  Require             ; Section 14.37
                   /  Scale               ; Section 14.39
                   /  Session             ; Section 14.42
                   /  Speed               ; Section 14.40
                   /  Supported           ; Section 14.43
                   /  Transport           ; Section 14.45
                   /  User-Agent          ; Section 14.47
                   /  extension-header



   response-header  =  Accept-Credentials      ; Section 14.2
                    /  Accept-Ranges           ; Section 14.5
                    /  Connection-Credentials  ; Section 14.12
                    /  ETag                    ; Section 14.21
                    /  Location                ; Section 14.28
                    /  Proxy-Authenticate      ; Section 14.29
                    /  Public                  ; Section 14.33
                    /  Range                   ; Section 14.34
                    /  Retry-After             ; Section 14.36
                    /  RTP-Info                ; Section 14.38
                    /  Scale                   ; Section 14.39
                    /  Session                 ; Section 14.42
                    /  Server                  ; Section 14.41
                    /  Speed                   ; Section 14.40
                    /  Transport               ; Section 14.45
                    /  Unsupported             ; Section 14.46
                    /  Vary                    ; Section 14.48
                    /  WWW-Authenticate        ; Section 14.50
                    /  extension-header



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   entity-header  =  Allow             ; Section 14.6
                  /  Content-Base      ; Section 14.13
                  /  Content-Encoding  ; Section 14.14
                  /  Content-Language  ; Section 14.15
                  /  Content-Length    ; Section 14.16
                  /  Content-Location  ; Section 14.17
                  /  Content-Type      ; Section 14.18
                  /  Expires           ; Section 14.22 and [H14.21]
                  /  Last-Modified     ; Section 14.27
                  /  extension-header


19.2.3 Header Syntax

   All header syntaxes not defined in this section are defined in
   section 14 of the HTTP 1.1 specification [3].


   Accept              =  "Accept" HCOLON
                          [ accept-range *(COMMA accept-range) ]
   accept-range        =  media-range *(SEMI accept-param)
   media-range         =  ( "*/*"
                       /  ( m-type SLASH "*" )
                       /  ( m-type SLASH m-subtype )
                          ) *( SEMI m-parameter )
   accept-param        =  ("q" EQUAL qvalue) / generic-param
   qvalue              =  ( "0" [ "." *3DIGIT ] )
                       /  ( "1" [ "." *3("0") ] )
   Accept-Credentials  =  "Accept-Credentials" HCOLON cred-decision CRLF
   cred-decision       =  ("User" COMMA [cred-info])
                       /  "Proxy"
                       /  "Any"
                       /  token ; For future extensions
   cred-info           =  cred-info-data *(COMMA cred-info-data)
   cred-info-data      =  DQ RTSP-URI DQ SEMI hash-alg SEMI base64
   hash-alg            =  "sha-1" / extension-alg
   extension-alg       =  token
   Accept-Encoding     =  "Accept-Encoding" HCOLON
                          [ encoding *(COMMA encoding) ]
   encoding            =  codings *(SEMI accept-param)
   codings             =  content-coding / "*"
   content-coding      =  token
   Accept-Language     =  "Accept-Language" HCOLON
                          [ language *(COMMA language) ]
   language            =  language-range *(SEMI accept-param)
   language-range      =  ( ( 1*8ALPHA *( "-" 1*8ALPHA ) ) / "*" )
   Accept-Ranges       =  "Accept-Ranges" HCOLON acceptable-ranges CRLF
   acceptable-ranges   =  (range-unit *(COMMA range-unit))



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                       /  "none"
   range-unit          =  "NPT" / "SMPTE" / "UTC" / extension-format
   extension-format    =  token
   Allow               =  "Allow" HCOLON [Method *(COMMA Method)]
   Authorization       =  "Authorization" HCOLON credentials
   credentials         =  ("Digest" LWS digest-response)
                       /  other-response
   digest-response     =  dig-resp *(COMMA dig-resp)
   dig-resp            =  username / realm / nonce / digest-uri
                       /  dresponse / algorithm / cnonce
                       /  opaque / message-qop
                       /  nonce-count / auth-param
   username            =  "username" EQUAL username-value
   username-value      =  quoted-string
   digest-uri          =  "uri" EQUAL LDQUOT digest-uri-value RDQUOT
   digest-uri-value    =  Request-URI
                          ; by HTTP/1.1
   message-qop         =  "qop" EQUAL qop-value
   cnonce              =  "cnonce" EQUAL cnonce-value
   cnonce-value        =  nonce-value
   nonce-count         =  "nc" EQUAL nc-value
   nc-value            =  8LHEX
   dresponse           =  "response" EQUAL request-digest
   request-digest      =  LDQUOT 32LHEX RDQUOT
   auth-param          =  auth-param-name EQUAL
                          ( token / quoted-string )
   auth-param-name     =  token
   other-response      =  auth-scheme LWS auth-param
                          *(COMMA auth-param)
   auth-scheme         =  token
   Bandwidth           =  "Bandwidth" HCOLON 1*DIGIT CRLF
   Blocksize           =  "Blocksize" HCOLON 1*DIGIT CRLF



   Cache-Control          =  "Cache-Control" HCOLON cache-directive CRLF
                             *(COMMA cache-directive)
   cache-directive        =  cache-rqst-directive
                          /  cache-rspns-directive
   cache-rqst-directive   =  "no-cache"
                          /  "max-stale" [EQUAL delta-seconds]
                          /  "min-fresh" EQUAL delta-seconds
                          /  "only-if-cached"
                          /  cache-extension
   cache-rspns-directive  =  "public"
                          /  "private"
                          /  "no-cache"
                          /  "no-transform"



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                          /  "must-revalidate"
                          /  "proxy-revalidate"
                          /  "max-age" EQUAL delta-seconds
                          /  cache-extension
   cache-extension        =  token [EQUAL (token / quoted-string)]
   delta-seconds          =  1*DIGIT



   Connection-Creds    =  "Connection-Credentials" HCOLON cred-info CRLF
   Connection          =  "Connection" HCOLON (connection-token)
                          *(COMMA connection-token) CRLF
   connection-token    =  token
   Content-Base        =  "Content-Base" HCOLON RTSP-URI-Ref CRLF
   Content-Encoding    =  "Content-Encoding" HCOLON
                          content-coding *(COMMA content-coding)
   Content-Language    =  "Content-Language" HCOLON
                          language-tag *(COMMA language-tag)
   language-tag        =  primary-tag *( "-" subtag )
   primary-tag         =  1*8ALPHA
   subtag              =  1*8ALPHA
   Content-Length      =  "Content-Length" HCOLON 1*DIGIT
   Content-Location    =  "Content-Location" HCOLON RTSP-URI-Ref
   Content-Type        =  ( "Content-Type" / "c" ) HCOLON media-type
   media-type          =  m-type SLASH m-subtype *(SEMI m-parameter)
   m-type              =  discrete-type / composite-type
   discrete-type       =  "text" / "image" / "audio" / "video"
                       /  "application" / extension-token
   composite-type      =  "message" / "multipart" / extension-token
   extension-token     =  ietf-token / x-token
   ietf-token          =  token
   x-token             =  "x-" token
   m-subtype           =  extension-token / iana-token
   iana-token          =  token
   m-parameter         =  m-attribute EQUAL m-value
   m-attribute         =  token
   m-value             =  token / quoted-string
   CSeq                =  "Cseq" HCOLON 1*DIGIT CRLF
   Date                =  "Date" HCOLON RTSP-date
   RTSP-date           =  rfc1123-date ; HTTP-date
   rfc1123-date        =  wkday "," SP date1 SP time SP "GMT"
   date1               =  2DIGIT SP month SP 4DIGIT
                          ; day month year (e.g., 02 Jun 1982)
   time                =  2DIGIT ":" 2DIGIT ":" 2DIGIT
                          ; 00:00:00 - 23:59:59
   wkday               =  "Mon" / "Tue" / "Wed"
                       /  "Thu" / "Fri" / "Sat" / "Sun"
   month               =  "Jan" / "Feb" / "Mar" / "Apr"



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                       /  "May" / "Jun" / "Jul" / "Aug"
                       /  "Sep" / "Oct" / "Nov" / "Dec"
   ETag                =  "ETag" HCOLON entity-tag
   Expires             =  "Expires" HCOLON delta-seconds
   From                =  "From" HCOLON from-spec
   from-spec           =  ( name-addr / addr-spec ) *( SEMI from-param )
   name-addr           =  [ display-name ] LAQUOT addr-spec RAQUOT
   addr-spec           =  RTSP-URI / absolute-URI
   absolute-URI        =  < As defined in RFC 3986 [8]>
   display-name        =  *(token LWS)/ quoted-string
   from-param          =  tag-param / generic-param
   tag-param           =  "tag" EQUAL token
   If-Match            =  "If-Match" HCOLON ( "*" / entity-tag-list)
   entity-tag-list     =  entity-tag *(COMMA entity-tag)
   entity-tag          =  [ weak ] opaque-tag
   weak                =  "W/"
   opaque-tag          =  quoted-string
   If-Modified-Since   =  "If-Modified-Since" HCOLON RTSP-date
   If-None-Match       =  "If-None-Match" HCOLON ("*" / entity-tag-list)
   Last-Modified       =  "Last-Modified" HCOLON RTSP-date
   Location            =  "Location" HCOLON RTSP-URI
   Proxy-Authenticate  =  "Proxy-Authenticate" HCOLON challenge
   challenge           =  ("Digest" LWS digest-cln *(COMMA digest-cln))
                          / other-challenge
   other-challenge     =  auth-scheme LWS auth-param
                          *(COMMA auth-param)
   digest-cln          =  realm / domain / nonce
                       /  opaque / stale / algorithm
                       /  qop-options / auth-param
   realm               =  "realm" EQUAL realm-value
   realm-value         =  quoted-string
   domain              =  "domain" EQUAL LDQUOT URI
                          *( 1*SP URI ) RDQUOT
   URI                 =  RTSP-URI / RTSP-URI-Ref
   nonce               =  "nonce" EQUAL nonce-value
   nonce-value         =  quoted-string
   opaque              =  "opaque" EQUAL quoted-string
   stale               =  "stale" EQUAL ( "true" / "false" )
   algorithm           =  "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
   qop-options         =  "qop" EQUAL LDQUOT qop-value
                          *("," qop-value) RDQUOT
   qop-value           =  "auth" / "auth-int" / token
   Proxy-Require       =  "Proxy-Require" HCOLON feature-tag CRLF
                          *(COMMA feature-tag)
   Proxy-Supported     =  "Proxy-Supported" HCOLON feature-tag
                          *(COMMA feature-tag) CRLF
   Public              =  "Public" HCOLON Method *(COMMA Method) CRLF
   Range               =  "Range" HCOLON ranges-list [exec-time] CRLF



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   ranges-list         =  ranges-spec *(COMMA ranges-spec)
   exec-time           =  SEMI "time" EQUAL utc-time
   ranges-spec         =  npt-range / utc-range / smpte-range
                       /  range-ext
   range-ext           =  extension-format "=" range-value
   range-value         =  1*(rtsp-unreserved / quoted-string / ":" )
   Referer             =  "Referer" HCOLON RTSP-URI-Ref
   Require             =  "Require" HCOLON feature-tag-list CRLF
   feature-tag-list    =  feature-tag *(COMMA feature-tag)



   RTP-Info        =  "RTP-Info" HCOLON rtsp-info-spec
                      *(COMMA rtsp-info-spec) CRLF
   rtsp-info-spec  =  stream-url 1*ssrc-parameter
   stream-url      =  "url" EQUAL DQ RTSP-URI-Ref DQ
   ssrc-parameter  =  LWS "ssrc" EQUAL ssrc HCOLON
                      ri-parameter *(SEMI ri-parameter)
   ri-parameter    =  "seq" EQUAL 1*DIGIT
                   /  "rtptime" EQUAL 1*DIGIT
   Retry-After     =  "Retry-After" HCOLON delta-seconds
                      [ comment ] *( SEMI retry-param )
   retry-param     =  ("duration" EQUAL delta-seconds)
                   /  generic-param



   Scale            =  "Scale" HCOLON ["-"] 1*DIGIT [ "." *DIGIT ] CRLF
   Speed            =  "Speed" HCOLON 1*DIGIT [ "." *DIGIT ] CRLF
   Server           =  "Server" HCOLON ( product / comment )
                       *(LWS (product / comment)) CRLF
   product          =  token [SLASH product-version]
   product-version  =  token
   comment          =  LPAREN *( ctext / quoted-pair) RPAREN
   Session          =  "Session" HCOLON session-id
                       [ SEMI "timeout" EQUAL delta-seconds ] CRLF
   Supported        =  "Supported" HCOLON [feature-tag-list] CRLF



   Timestamp        =  "Timestamp" HCOLON timestamp-value LWS [delay]
   timestamp-value  =  *DIGIT [ "." *DIGIT ]
   delay            =  *DIGIT [ "." *DIGIT ]
   Transport        =  "Transport" HCOLON transport-spec
                       *(COMMA transport-spec) CRLF
   transport-spec   =  transport-id *tr-parameter
   transport-id     =  trans-id-rtp / other-trans
   trans-id-rtp     =  "RTP/" profile ["/" lower-transport]



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                       ; no LWS is allowed inside transport-id
   other-trans      =  token *("/" token)



   profile          =  "AVP" / "SAVP" / "AVPF" / token
   lower-transport  =  "TCP" / "UDP" / token
   tr-parameter     =  SEMI ( "unicast" / "multicast" )
                    /  SEMI "interleaved" EQUAL channel [ "-" channel ]
                    /  SEMI "append"
                    /  SEMI "ttl" EQUAL ttl
                    /  SEMI "layers" EQUAL 1*DIGIT
                    /  SEMI "ssrc" EQUAL ssrc *(SLASH ssrc)
                    /  SEMI "client_ssrc" EQUAL ssrc
                    /  SEMI "mode" EQUAL mode-spec
                    /  SEMI "dest_addr" EQUAL addr-list
                    /  SEMI "src_addr" EQUAL addr-list
                    /  SEMI trn-param-ext
                    /  SEMI "setup" EQUAL contrans-setup
                    /  SEMI "connection" EQUAL contrans-con
   contrans-setup   =  "active" / "passive" / "actpass"
   contrans-con     =  "new" / "existing"
   trn-param-ext    =  par-name EQUAL trn-par-value
   par-name         =  token
   trn-par-value    =  *(rtsp-unreserved / DQ *TEXT DQ)
   ttl              =  1*3DIGIT ; 0 to 255
   ssrc             =  8HEX
   channel          =  1*3DIGIT
   mode-spec        =  ( DQ mode *(COMMA mode) DQ )
   mode             =  "PLAY" / "RECORD" / token
   addr-list        =  quoted-addr *(SLASH quoted-addr)
   quoted-addr      =  DQ (host-port / extension-addr) DQ
   host-port        =  host [":" port]
                    /  ":" port
   extension-addr   =  1*qdtext
   host             =  < As defined in RFC 3986 [8]>
   port             =  < As defined in RFC 3986 [8]>



   Unsupported       =  "Unsupported" HCOLON feature-tag-list CRLF
   User-Agent        =  "User-Agent" HCOLON ( product / comment )
                        0*(LWS (product / comment)) CRLF
   Vary              =  "Vary" HCOLON ( "*" / field-name-list)
   field-name-list   =  field-name *(COMMA field-name)
   field-name        =  token
   Via               =  "Via" HCOLON via-parm *(COMMA via-parm)
   via-parm          =  sent-protocol LWS sent-by *( SEMI via-params )



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   via-params        =  via-ttl / via-maddr
                     /  via-received / via-branch
                     /  via-extension
   via-ttl           =  "ttl" EQUAL ttl
   via-maddr         =  "maddr" EQUAL host
   via-received      =  "received" EQUAL (IPv4address / IPv6address)
   IPv4address       =  < As defined in RFC 3986 [8]>
   IPv6address       =  < As defined in RFC 3986 [8]>
   via-branch        =  "branch" EQUAL token
   via-extension     =  generic-param
   sent-protocol     =  protocol-name SLASH protocol-version
                        SLASH transport-prot
   protocol-name     =  "RTSP" / token
   protocol-version  =  token
   transport-prot    =  "UDP" / "TCP" / "TLS" / other-transport
   other-transport   =  token
   sent-by           =  host [ COLON port ]
   WWW-Authenticate  =  "WWW-Authenticate" HCOLON challenge


19.3 SDP extension Syntax

   This section defines in ABNF the SDP extensions defined for RTSP.
   See section C for the definition of the extensions in text.


   control-attribute  =  "a=control:" *SP RTSP-URI
   a-range-def        =  "a=range:" ranges-spec CRLF
   a-etag-def         =  "a=etag:" entity-tag CRLF


20 Security Considerations

   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security considerations outlined in [H15]
   apply.  Specifically, please note the following:

        Abuse of Server Log Information: RTSP and HTTP servers will
             presumably have similar logging mechanisms, and thus should
             be equally guarded in protecting the contents of those
             logs, thus protecting the privacy of the users of the
             servers. See [H15.1.1] for HTTP server recommendations
             regarding server logs.

        Transfer of Sensitive Information: There is no reason to believe
             that information transferred via RTSP may be any less
             sensitive than that normally transmitted via HTTP.
             Therefore, all of the precautions regarding the protection



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             of data privacy and user privacy apply to implementors of
             RTSP clients, servers, and proxies. See [H15.1.2] for
             further details.

        Attacks Based On File and Path Names: Though RTSP URIs are
             opaque handles that do not necessarily have file system
             semantics, it is anticipated that many implementations will
             translate portions of the Request-URIs directly to file
             system calls. In such cases, file systems SHOULD follow the
             precautions outlined in [H15.5], such as checking for ".."
             in path components.

        Personal Information: RTSP clients are often privy to the same
             information that HTTP clients are (user name, location,
             etc.)  and thus should be equally sensitive. See [H15.1]
             for further recommendations.

        Privacy Issues Connected to Accept Headers: Since may of the
             same "Accept" headers exist in RTSP as in HTTP, the same
             caveats outlined in [H15.1.4] with regards to their use
             should be followed.

        DNS Spoofing: Presumably, given the longer connection times
             typically associated to RTSP sessions relative to HTTP
             sessions, RTSP client DNS optimizations should be less
             prevalent.  Nonetheless, the recommendations provided in
             [H15.3] are still relevant to any implementation which
             attempts to rely on a DNS-to-IP mapping to hold beyond a
             single use of the mapping.

        Location Headers and Spoofing: If a single server supports
             multiple organizations that do not trust each another, then
             it needs to check the values of Location and Content-
             Location header fields in responses that are generated
             under control of said organizations to make sure that they
             do not attempt to invalidate resources over which they have
             no authority. ([H15.4])

   In addition to the recommendations in the current HTTP specification
   (RFC 2616 [3], as of this writing) and also of the previous RFC2068
   [39], future HTTP specifications may provide additional guidance on
   security issues.

   The following are added considerations for RTSP implementations.

        Concentrated denial-of-service attack: The protocol offers the
             opportunity for a remote-controlled denial-of-service
             attack.  See Section 20.1.



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        Session hijacking: Since there is no or little relation between
             a transport layer connection and an RTSP session, it is
             possible for a malicious client to issue requests with
             random session identifiers which would affect unsuspecting
             clients. The server SHOULD use a large, random and non-
             sequential session identifier to minimize the possibility
             of this kind of attack. For real session security, client
             authentication needs to be performed.

        Authentication: Servers SHOULD implement both basic and digest
             [10] authentication. In environments requiring tighter
             security for the control messages, the transport layer
             mechanism TLS (RFC 4346 [9]) SHOULD be used.

        Stream issues: RTSP only provides for stream control. Stream
             delivery issues are not covered in this section, nor in the
             rest of this draft. RTSP implementations will most likely
             rely on other protocols such as RTP, IP multicast, RSVP and
             IGMP, and should address security considerations brought up
             in those and other applicable specifications.

        Persistently suspicious behavior: RTSP servers SHOULD return
             error code 403 (Forbidden) upon receiving a single instance
             of behavior which is deemed a security risk. RTSP servers
             SHOULD also be aware of attempts to probe the server for
             weaknesses and entry points and MAY arbitrarily disconnect
             and ignore further requests clients which are deemed to be
             in violation of local security policy.

        Scope of Multicast: If RTSP is used to control the transmission
             of media onto a multicast network it is need to consider
             the scope that delivery has. RTSP supports the TTL
             Transport header parameter to indicate this scope. However
             such scope control is risk as it may be set to large and
             distribute media beyond the intended scope.

        TLS through proxies: If one uses the possibility to connect TLS
             in multiple legs (Section 18.3 one really needs to be aware
             of the trust model. That procedure requires full faith and
             trust in all proxies that one allows to connect through.
             They are man in the middle and has access to all that goes
             on over the TLS connection. Thus it is important to
             consider if that trust model is acceptable in the actual
             application.

20.1 Remote denial of Service Attack

   The attacker may initiate traffic flows to one or more IP addresses



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   by specifying them as the destination in SETUP requests. While the
   attacker's IP address may be known in this case, this is not always
   useful in prevention of more attacks or ascertaining the attackers
   identity. Thus, an RTSP server MUST only allow client-specified
   destinations for RTSP-initiated traffic flows if the server has
   ensured that the specified destination address accepts receiving
   media through different security mechanisms. Security mechanism that
   are acceptable in an increased generality are; verification of the
   client's identity, either against a database of known users using
   RTSP authentication mechanisms (preferably digest authentication or
   stronger); a list of addresses that accept to be media destinations,
   especially considering user identity; and media path based
   verification.

   The server SHOULD NOT allow the destination field to be set unless a
   mechanism exists in the system to authorize the request originator to
   direct streams to the recipient. It is preferred that this
   authorization be performed by the media recipient (destination)
   itself and the credentials passed along to the server. However, in
   certain cases, such as when recipient address is a multicast group,
   or when the recipient is unable to communicate with the server in an
   out-of-band manner, this may not be possible. In these cases server
   may chose another method such as a server-resident authorization list
   to ensure that the request originator has the proper credentials to
   request stream delivery to the recipient.

   One solution that performs the necessary verification of acceptance
   of media suitable for unicast based delivery is the ICE based NAT
   traversal method described in [38]. By using random passwords and
   username the probability of unintended indication as a valid media
   destination is very low. If the server include in its STUN requests a
   cookie (consisting of random material) that is the destination echo
   back the solution is also safe against having a off-path attacker
   being able to spoof the STUN checks. Leaving this solution vulnerable
   only to on-path attackers that can see the STUN requests go to the
   target of attack.

   For delivery to multicast addresses there is need for another
   solution which is not specified here.

21 IANA Considerations

   This section sets up a number of registries for RTSP 2.0 that should
   be maintained by IANA. For each registry there is a description on
   what it is required to contain, what specification is needed when
   adding a entry with IANA, and finally the entries that this document
   needs to register. See also the section 1.6 "Extending RTSP". There
   is also an IANA registration of two SDP attributes.



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   The sections describing how to register an item uses some of the
   requirements level described in RFC 2434 [18], namely "First Come,
   First Served", "Specification Required", and "Standards Action".

   A registration request to IANA MUST contain the following
   information:

        o A name of the item to register according to the rules
          specified by the intended registry.

        o Indication of who has change control over the feature (for
          example, IETF, ISO, ITU-T, other international standardization
          bodies, a consortium, a particular company or group of
          companies, or an individual);

        o A reference to a further description, if available, for
          example (in order of preference) an RFC, a published standard,
          a published paper, a patent filing, a technical report,
          documented source code or a computer manual;

        o For proprietary features, contact information (postal and
          email address);

21.1 Feature-tags

21.1.1 Description

   When a client and server try to determine what part and functionality
   of the RTSP specification and any future extensions that its counter
   part implements there is need for a namespace.  This registry
   contains named entries representing certain functionality.

   The usage of feature-tags is explained in section 10 and 11.1.

21.1.2 Registering New Feature-tags with IANA

   The registering of feature-tags is done on a first come, first served
   basis.

   The name of the feature MUST follow these rules: The name may be of
   any length, but SHOULD be no more than twenty characters long.  The
   name MUST NOT contain any spaces, or control characters. The
   registration SHALL indicate if the feature-tag applies to clients,
   servers, or proxies only or any combinations of these. Any
   proprietary feature SHALL have as the first part of the name a vendor
   tag, which identifies the organization.

21.1.3 Registered entries



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   The following feature-tags are in this specification defined and
   hereby registered. The change control belongs to the IETF.

        play.basic: The minimal implementation for playback operations
             according to section D. Applies for both clients, servers
             and proxies.

        play.scale: Support of scale operations for media playback.
             Applies only for servers.

        play.speed: Support of the speed functionality for playback.
             Applies only for servers

21.2 RTSP Methods

21.2.1 Description

   What a method is, is described in section 11.  Extending the protocol
   with new methods allow for totally new functionality.

21.2.2 Registering New Methods with IANA

   A new method MUST be registered through an IETF standard track
   document. The reason is that new methods may radically change the
   protocols behavior and purpose.

   A specification for a new RTSP method MUST consist of the following
   items:

        o A method name which follows the ABNF rules for methods.

        o A clear specification on what action and response a request
          with the method will result in. Which directions the method is
          used, C -> S or S -> C or both. How the use of headers, if
          any, modifies the behavior and effect of the method.

        o A list or table specifying which of the registered headers
          that are allowed to use with the method in request or/and
          response.

        o Describe how the method relates to network proxies.

21.2.3 Registered Entries

   This specification, RFCXXXX, registers 9 methods: DESCRIBE,
   GET_PARAMETER, OPTIONS, PAUSE, PLAY, REDIRECT, SETUP, SET_PARAMETER,
   and TEARDOWN.




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21.3 RTSP Status Codes

21.3.1 Description

   A status code is the three digit numbers used to convey information
   in RTSP response messages, see 7. The number space is limited and
   care should be taken not to fill the space.

21.3.2 Registering New Status Codes with IANA

   A new status code can only be registered by an IETF standards track
   document. A specification for a new status code MUST specify the
   following:

        o The requested number.

        o A description what the status code means and the expected
          behavior of the sender and receiver of the code.

21.3.3 Registered Entries

   RFCXXX, registers the numbered status code defined in the ABNF entry
   "Status-Code" except "extension-code" in section 19.2.2.

21.4 RTSP Headers

21.4.1 Description

   By specifying new headers a method(s) can be enhanced in many
   different ways. An unknown header will be ignored by the receiving
   entity. If the new header is vital for a certain functionality, a
   feature-tag for the functionality can be created and demanded to be
   used by the counter-part with the inclusion of a Require header
   carrying the feature-tag.

21.4.2 Registering New Headers with IANA

   A public available specification is required to register a header.
   The specification SHOULD be a standards document, preferable an IETF
   RFC.

   The specification MUST contain the following information:

        o The name of the header.

        o An ABNF specification of the header syntax.

        o A list or table specifying when the header may be used,



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          encompassing all methods, their request or response, the
          direction (C -> S or S -> C).

        o How the header is to be handled by proxies.

        o A description of the purpose of the header.

21.4.3 Registered entries

   All headers specified in section 14 in RFCXXXX are to be registered.

   Furthermore the following RTSP headers defined in other
   specifications are registered:

        o x-wap-profile defined in [40].

        o x-wap-profile-diff defined in [40].

        o x-wap-profile-warning defined in [40].

        o x-predecbufsize defined in [40].

        o x-initpredecbufperiod defined in [40].

        o x-initpostdecbufperiod defined in [40].

        o 3gpp-videopostdecbufsize defined in [40].

        o 3GPP-Link-Char defined in [40].

        o 3GPP-Adaptation defined in [40].

        o 3GPP-QoE-Metrics defined in [40].

        o 3GPP-QoE-Feedback defined in [40].

   The use of "X-" is NOT RECOMMENDED but the above headers in the
   register list was defined prior to the clarification.

21.5 Transport Header registries

   The transport header contains a number of parameters which have
   possibilities for future extensions. Therefore registries for these
   needs to be defined.

21.5.1 Transport Protocol Specification

   A registry for the parameter transport-protocol specification SHALL



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   be defined with the following rules:

        o Registering require an public available standards
          specification.

        o A contact person or organization with address and email.

        o A value definition that are following the ABNF syntax
          definition.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers the following values:

        o Use of the RTP [16] protocol for media transport in
          combination with the "RTP profile for audio and video
          conferences with minimal control" [2] over UDP. The usage is
          explained in RFC XXXX, appendix  B.1.

        o the same as RTP/AVP.

        o Use of the RTP [16] protocol for media transport in
          combination with the "Extended RTP Profile for RTCP-based
          Feedback (RTP/AVPF)" [19] over UDP. The usage is explained in
          RFC XXXX, appendix  B.1.

        o the same as RTP/AVPF.

        o Use of the RTP [16] protocol for media transport in
          combination with the "The Secure Real-time Transport Protocol
          (SRTP)"  [20] over UDP. The usage is explained in RFC XXXX,
          appendix  B.1.

        o the same as RTP/SAVP.

        o Use of the RTP [16] protocol for media transport in
          combination with the " [6] over UDP. The usage is explained in
          RFC XXXX, appendix  B.1.

        o the same as RTP/SAVPF.

        o Use of the RTP [16] protocol for media transport in
          combination with the "RTP profile for audio and video
          conferences with minimal control" [2] over TCP. The usage is
          explained in RFC XXXX, appendix B.2.2.

        o Use of the RTP [16] protocol for media transport in



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          combination with the "Extended RTP Profile for RTCP-based
          Feedback (RTP/AVPF)" [19] over TCP. The usage is explained in
          RFC XXXX, appendix B.2.2.

        o Use of the RTP [16] protocol for media transport in
          combination with the "The Secure Real-time Transport Protocol
          (SRTP)"  [20] over TCP. The usage is explained in RFC XXXX,
          appendix  B.2.2.

        o Use of the RTP [16] protocol for media transport in
          combination with the " [6] over TCP. The usage is explained in
          RFC XXXX, appendix B.2.2.

21.5.2 Transport modes

   A registry for the transport parameter mode SHALL be defined with the
   following rules:

        o Registering requires an IETF standard tracks document.

        o A contact person or organization with address and email.

        o A value definition that are following the ABNF token
          definition.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers 1 values:

        PLAY: See RFC XXXX.

21.5.3 Transport Parameters

   A registry for parameters that may be included in the Transport
   header SHALL be defined with the following rules:

        o Registering required a Open Standards document

        o A value definition that are following the ABNF token
          definition.

        o A describing text that explains how the registered value are
          used in RTSP.

   This specification registers all the transport parameters defined in
   Section 14.45.




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21.6 Cache Directive Extensions

   There exist a number of cache directives which can be sent in the
   Cache-Control header. A registry for this cache directives SHALL be
   defined with the following rules:

        o Registering requires an IETF standard tracks document.

        o A registration is required to contain a contact person.

        o Name of the directive and a definition of the value, if any.

        o Specification if it is an request or response directive.

        o A describing text that explains how the cache directive is
          used for RTSP controlled media streams.

   This specification registers the following values:

        no-cache:

        public:

        private:

        no-transform:

        only-if-cached:

        max-stale:

        min-fresh:

        must-revalidate:

        proxy-revalidate:

        max-age:

21.7 Accept-Credentials

   The security framework's TLS connection mechanism has two
   registerable entities.

21.7.1 Accept-Credentials policies

   In section 18.3.1 three policies for how to handle certificates.
   Further policies may be defined and SHALL be registered with IANA



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   using the following rules:

        o Registering requires an IETF standard tracks document.

        o A registration is required name a contact person.

        o Name of the policy.

        o A describing text that explains how the policy works for
          handling the certificates.

   This specification registers the following values:

        Any

        Proxy

        User

21.7.2 Accept-Credentials hash algorithms

   The Accept-Credentials header (See Section 14.2) allows for the usage
   of other algorithms for hashing the DER records of accepted entities.
   The registration of any future algorithm is expected to be extremely
   rare and could also be an interoperability problem. Therefore the
   bare for registering new algorithms is placed intentional high.

   Any registration of a new hash algorithm SHALL meet the following
   requirement:

        o Registration requires an IETF standard track document.

        o A definition of the algorithm and its identifier meeting the
          "token" ABNF requirement.

21.8 Range header formats

   The Range header allows for different range formats. New ones may be
   registered, but moderation should be applied as it makes
   interoperability more difficult. A registration SHALL fulfill the
   following requirements:

        o A publicly available standards document

        o A ABNF definition of the range format that fulfills the
          "range-ext" definition.

        o A Contact person for the registration



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        o Rules for how one handles the range when using a negative
          Scale.

21.9 URI Schemes

   This specification defines two URI schemes ("rtsp" and "rtsps") and
   reserves a third one ("rtspu"). Registrations are following RFC 4395
   [41].

21.9.1 The rtsp URI Scheme

        URI scheme name: rtsp

        Status: Permanent

        URI scheme syntax: See Section 19.2.1 of RFC XXXX.

        URI scheme semantics: The rtsp scheme is used to indicate
             resources accessible through the usage of the Real-time
             Streaming Protocol (RTSP). RTSP allows different operations
             on the resource identified by the URI, but the primary
             purpose is the streaming delivery of the resource to a
             client. However the operations that are currently defined
             are: Describing the resource for the purpose of configuring
             the receiving entity (DESCRIBE), configuring the delivery
             method and its addressing (SETUP), controlling the delivery
             (PLAY and PAUSE), reading or setting of resource related
             parameters (SET_PARAMETER and GET_PARAMETER, and
             termination of the session context created (TEARDOWN).

        Encoding considerations: IRIs in this scheme are defined and
             needs to be encoded as RTSP URIs when used within the RTSP
             protocol.  That encoding is done according to RFC 3987.

        Applications/protocols that use this URI scheme name:  RTSP 1.0
             (RFC 2326), RTSP 2.0 (RFC XXXX)

        Interoperability considerations: The change in URI syntax
             performed between RTSP 1.0 and 2.0 can create
             interoperability issues.

        Security considerations: All the security threats identified in
             Section 7 of RFC 3986 applies also to this scheme. They
             needs to be reviewed and considered in any implementation
             utilizing this scheme.

        Contact: Magnus Westerlund, magnus.westerlund@ericsson.com




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        Author/Change controller: IETF MMUSIC WG

        References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX,

21.9.2 The rtsps URI Scheme

        URI scheme name: rtsps

        Status: Permanent

        URI scheme syntax: See Section 19.2.1 of RFC XXXX.

        URI scheme semantics: The rtsps scheme is used to indicate
             resources accessible through the usage of the Real-time
             Streaming Protocol (RTSP) over TLS. RTSP allows different
             operations on the resource identified by the URI, but the
             primary purpose is the streaming delivery of the resource
             to a client. However the operations that are currently
             defined are: Describing the resource for the purpose of
             configuring the receiving entity (DESCRIBE), configuring
             the delivery method and its addressing (SETUP), controlling
             the delivery (PLAY and PAUSE), reading or setting of
             resource related parameters (SET_PARAMETER and
             GET_PARAMETER, and termination of the session context
             created (TEARDOWN).

        Encoding considerations: IRIs in this scheme are defined and
             needs to be encoded as RTSP URIs when used within the RTSP
             protocol.  That encoding is done according to RFC 3987.

        Applications/protocols that use this URI scheme name:  RTSP 1.0
             (RFC 2326), RTSP 2.0 (RFC XXXX)

        Interoperability considerations: The change in URI syntax
             performed between RTSP 1.0 and 2.0 can create
             interoperability issues.

        Security considerations: All the security threats identified in
             Section 7 of RFC 3986 applies also to this scheme. They
             needs to be reviewed and considered in any implementation
             utilizing this scheme.

        Contact: Magnus Westerlund, magnus.westerlund@ericsson.com

        Author/Change controller: IETF MMUSIC WG

        References: RFC 2326, RFC 3986, RFC 3987, RFC XXXX




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21.9.3 The rtspu URI Scheme

        URI scheme name: rtspu

        Status: Permanent

        URI scheme syntax: See Section 3.2 of RFC 2326.

        URI scheme semantics: The rtspu scheme is used to indicate
             resources accessible through the usage of the Real-time
             Streaming Protocol (RTSP) over unrelaible datagram
             transport. RTSP allows different operations on the resource
             identified by the URI, but the primary purpose is the
             streaming delivery of the resource to a client. However the
             operations that are currently defined are:  Describing the
             resource for the purpose of configuring the receiving
             entity (DESCRIBE), configuring the delivery method and its
             addressing (SETUP), controlling the delivery (PLAY and
             PAUSE), reading or setting of resource related parameters
             (SET_PARAMETER and GET_PARAMETER, and termination of the
             session context created (TEARDOWN).

        Encoding considerations: IRIs in this scheme are defined and
             needs to be encoded as RTSP URIs when used within the RTSP
             protocol.  That encoding is done according to RFC 3987.

        Applications/protocols that use this URI scheme name:  RTSP 1.0
             (RFC 2326)

        Interoperability considerations: The definition of the transport
             mechanism of RTSP over UDP has interoperability issues.
             That makes the usage of this scheme problematic.

        Security considerations: All the security threats identified in
             Section 7 of RFC 3986 applies also to this scheme. They
             needs to be reviewed and considered in any implementation
             utilizing this scheme.

        Contact: Magnus Westerlund, magnus.westerlund@ericsson.com

        Author/Change controller: IETF MMUSIC WG

        References: RFC 2326, RFC 3986, RFC 3987

21.10 SDP attributes

   This specification defines two SDP [1] attributes that it is
   requested that IANA register.



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   SDP Attribute ("att-field"):

        Attribute name:     range
        Long form:          Media Range Attribute
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition.

        Attribute name:     control
        Long form:          RTSP control URI
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             Absolute or Relative URIs.

        Attribute name:     etag
        Long form:          Entity Tag
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             See ABNF definition





A RTSP Protocol State Machine

   The RTSP session state machine describes the behavior of the protocol
   from RTSP session initialization through RTSP session termination.

   The State machine is defined on a per session basis which is uniquely
   identified by the RTSP session identifier. The session may contain
   one or more media streams depending on state. If a single media
   stream is part of the session it is in non-aggregated control. If two
   or more is part of the session it is in aggregated control.

   The below state machine is a normative description of the protocols
   behavior. However, in case of ambiguity with the earlier parts of
   this specification, the description in the earlier parts SHALL take
   precedence.



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A.1 States

   The state machine contains three states, described below. For each
   state there exist a table which shows which requests and events that
   is allowed and if they will result in a state change.

        Init: Initial state no session exist.

        Ready: Session is ready to start playing.

        Play: Session is playing, i.e. sending media stream data in the
             direction S -> C.

A.2 State variables

   This representation of the state machine needs more than its state to
   work. A small number of variables are also needed and is explained
   below.

        NRM: The number of media streams part of this session.

        RP: Resume point, the point in the presentation time line at
             which a request to continue will resume from. A time format
             for the variable is not mandated.

A.3 Abbreviations

   To make the state tables more compact a number of abbreviations are
   used, which are explained below.

        IFI: IF Implemented.

        md: Media

        PP: Pause Point, the point in the presentation time line at
             which the presentation was paused.

        Prs: Presentation, the complete multimedia presentation.

        RedP: Redirect Point, the point in the presentation time line at
             which a REDIRECT was specified to occur.

        SES: Session.

A.4 State Tables

   This section contains a table for each state. The table contains all
   the requests and events that this state is allowed to act on.  The



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   events which is method names are, unless noted, requests with the
   given method in the direction client to server (C -> S). In some
   cases there exist one or more requisite. The response column tells
   what type of response actions should be performed. Possible actions
   that is requested for an event includes: response codes, e.g. 200,
   headers that MUST be included in the response, setting of state
   variables, or setting of other session related parameters. The new
   state column tells which state the state machine changes to.

   The response to valid request meeting the requisites is normally a
   2xx (SUCCESS) unless other noted in the response column. The
   exceptions needs to be given a response according to the response
   column. If the request does not meet the requisite, is erroneous or
   some other type of error occur the appropriate response code MUST be
   sent. If the response code is a 4xx the session state is unchanged. A
   response code of 3rr will result in that the session is ended and its
   state is changed to Init. A response code of 304 results in no state
   change. However there exist restrictions to when a 3rr response may
   be used. A 5xx response SHALL not result in any change of the session
   state, except if the error is not possible to recover from. A
   unrecoverable error SHALL result the ending of the session. As it in
   the general case can't be determined if it was a unrecoverable error
   or not the client will be required to test. In the case that the next
   request after a 5xx is responded with 454 (Session Not Found) the
   client knows that the session has ended.

   The server will timeout the session after the period of time
   specified in the SETUP response, if no activity from the client is
   detected.  Therefore there exist a timeout event for all states
   except Init.

   In the case that NRM=1 the presentation URI is equal to the media URI
   or a specified presentation URI. For NRM>1 the presentation URI MUST
   be other than any of the medias that are part of the session. This
   applies to all states.


   The methods in Table 13 do not have any effect on the state machine
   or the state variables. However some methods do change other session
   related parameters, for example SET_PARAMETER which will set the
   parameter(s) specified in its body. Also all of these methods that
   allows Session header will also update the keep-alive timer for the
   session.


   The initial state of the state machine, see Table 14 can only be left
   by processing a correct SETUP request. As seen in the table the two
   state variables are also set by a correct request. This table also



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      Event         Prerequisite    Response
      ______________________________________________________________
      DESCRIBE      Needs REDIRECT  3rr, Redirect
      DESCRIBE                      200, Session description
      OPTIONS       Session ID      200, Reset session timeout timer
      OPTIONS                       200
      SET_PARAMETER Valid parameter 200, change value of parameter
      GET_PARAMETER Valid parameter 200, return value of parameter


   Table 13: None state-machine changing events


       Action           Requisite       New State  Response

_____________________________________________________________
       SETUP                              Ready    NRM=1, RP=0.0
       SETUP            Needs Redirect    Init     3rr Redirect
       S -> C:REDIRECT  No Session hdr    Init     Terminate all SES


   Table 14: State: Init

   shows that a correct SETUP can in some cases be redirected to another
   URI and/or server by a 3rr response.


   Action           Requisite          New State  Response
   _____________________________________________________________________
   SETUP            New URI              Ready    NRM+=1
   SETUP            Setten up URI        Ready    Change transport param
   TEARDOWN         Prs URI,             Init     No session hdr, NRM=0
   TEARDOWN         md URI,NRM=1         Init     No Session hdr, NRM=0
   TEARDOWN         md URI,NRM>1         Ready    Session hdr, NRM-=1
   PLAY             Prs URI, No range    Play     Play from RP
   PLAY             Prs URI, Range       Play     According to range
   PAUSE            Prs URI              Ready    Return PP
   S -> C:REDIRECT  Range hdr            Ready    Set RedP
   S -> C:REDIRECT  no range hdr         Init     Session is removed
   Timeout                               Init
   RedP reached                          Init     TEARDOWN of session


   Table 15: State: Ready


   In the Ready state, see Table 15, some of the actions are depending
   on the number of media streams (NRM) in the session, i.e. aggregated
   or non-aggregated control. A setup request in the ready state can
   either add one more media stream to the session or if the media


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   stream (same URI) already is part of the session change the transport
   parameters. TEARDOWN is depending on both the Request-URI and the
   number of media stream within the session. If the Request-URI is the
   presentations URI the whole session is torn down. If a media URI is
   used in the TEARDOWN request and more than one media exist in the
   session, the session will remain and a session header MUST be
   returned in the response. If only a single media stream remains in
   the session when performing a TEARDOWN with a media URI the session
   is removed. The number of media streams remaining after tearing down
   a media stream determines the new state.


   Action           Requisite          New State Response
   _____________________________________________________________________
   PAUSE            PrsURI               Ready   Set RP to present point
   PP reached                            Ready   RP = PP
   End of media     All media            Play    Set RP = End of media
   End of range                          Play    Set RP = End of range
   PLAY             Prs URI, No range    Play    Play from present point
   PLAY             Prs URI, Range       Play    According to range
   SETUP            New URI              Play    455
   SETUP            Setuped URI          Play    455
   SETUP            Setuped URI, IFI     Play    Change transport param.
   TEARDOWN         Prs URI              Init    No session hdr
   TEARDOWN         md URI,NRM=1         Init    No Session hdr, NRM=0
   TEARDOWN         md URI               Play    455
   S -> C:REDIRECT  Range hdr            Play    Set RedP
   S -> C:REDIRECT  no range hdr         Init    Session is removed
   RedP reached                          Init    TEARDOWN of session
   Timeout                               Init    Stop Media playout


   Table 16: State: Play


   The Play state table, see Table 16, is the largest. The table
   contains an number of requests that has presentation URI as a
   prerequisite on the Request-URI, this is due to the exclusion of
   non-aggregated stream control in sessions with more than one media
   stream.

   To avoid inconsistencies between the client and server, automatic
   state transitions are avoided. This can be seen at for example "End
   of media" event when all media has finished playing, the session
   still remain in Play state. An explicit PAUSE request MUST be sent to
   change the state to Ready. It may appear that there exist an
   automatic transitions in "RedP reached" and "PP reached", however
   they are requested and acknowledge before they take place. The time
   at which the transition will happen is known by looking at the range


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   header. If the client sends request close in time to these
   transitions it needs to be prepared for getting error message as the
   state may or may not have changed.

B Media Transport Alternatives

   This section defines how certain combinations of protocols, profiles
   and lower transports are used. This includes the usage of the
   Transport header's source and destination address parameters
   "src_addr" and "dest_addr".

B.1 RTP

   This section defines the interaction of RTSP with respect to the RTP
   protocol [16]. It also defines any necessary media transport
   signalling with regards to RTP.

   The available RTP profiles and lower layer transports are described
   below along with rules on signalling the available combinations.

B.1.1 AVP

   The usage of the "RTP Profile for Audio and Video Conferences with
   Minimal Control" [2] when using RTP for media transport over
   different lower layer transport protocols is defined below in regards
   to RTSP.

   One such case is defined within this document, the use of embedded
   (interleaved) binary data as defined in section 12.  The usage of
   this method is indicated by include the "interleaved" parameter.

   When using embedded binary data the "src_addr" and "dest_addr" SHALL
   NOT be used. This addressing and multiplexing is used as defined with
   use of channel numbers and the interleaved parameter.

B.1.2 AVP/UDP

   This part describes sending of RTP [16] over lower transport layer
   UDP [11] according to the profile "RTP Profile for Audio and Video
   Conferences with Minimal Control" defined in RFC 3551 [2]. This
   profiles requires one or two uni- or bi-directional UDP flows per
   media stream. The first UDP flow is for RTP and the second is for
   RTCP. Embedding of RTP data with the RTSP messages, in accordance
   with section 12, SHOULD NOT be performed when RTSP messages are
   transported over unreliable transport protocols, like UDP [11].

   The RTP/UDP and RTCP/UDP flows can be established using the Transport
   header's "src_addr", and "dest_addr" parameters.



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   In RTSP PLAY mode, the transmission of RTP packets from client to
   server is unspecified. The behavior in regards to such RTP packets
   MAY be defined in future.

   The "src_addr" and "dest_addr" parameters are used in the following
   way for media playback, i.e. Mode=PLAY:

        o The "src_addr" and "dest_addr" parameters MUST contain either
          1 or 2 address specifications.

        o Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
          contain either:

          - both an address and a port number, or

          - a port number without an address

        o The first address and port pair given in either of the
          parameters applies to the RTP stream. The second address and
          port pair if present applies to the RTCP stream.

        o The RTP/UDP packets from the server to the client SHALL be
          sent to the address and port given by first address and port
          pair of the "dest_addr" parameter.

        o The RTCP/UDP packets from the server to the client SHALL be
          sent to the address and port given by the second address and
          port pair of the "dest_addr" parameter. If no second pair is
          specified RTCP SHALL NOT be sent.

        o The RTCP/UDP packets from the client to the server SHALL be
          sent to the address and port given by the second address and
          port pair of the "src_addr" parameter. If no second pair is
          given RTCP SHALL NOT be sent.

        o The RTP/UDP packets from the client to the server SHALL be
          sent to the address and port given by the first address and
          port pair of the "src_addr" parameter.

        o RTP and RTCP Packets SHOULD be sent from the corresponding
          receiver port, i.e. RTCP packets from server should be sent
          from the "src_addr" parameters second address port pair.

B.1.3 AVPF/UDP

   The RTP profile "Extended RTP Profile for RTCP-based Feedback
   (RTP/AVPF)" [19] MAY be used as RTP profiles in session using RTP.
   All that is defined for AVP SHALL also apply for AVPF.



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   The usage of AVPF is indicated by the media initialization protocol
   used. In the case of SDP it is indicated by media lines (m=)
   containing the profile RTP/AVPF. That SDP MAY also contain further
   AVPF related SDP attributes configuring the AVPF session regarding
   reporting interval and feedback messages that shall be used that
   SHALL be followed.

B.1.4 SAVP/UDP

   The RTP profile "The Secure Real-time Transport Protocol (SRTP)"
   [20] is an RTP profile (SAVP) that MAY be used in RTSP sessions using
   RTP. All that is defined for AVP SHALL also apply for SAVP.

   The usage of SRTP requires that a security association is
   established. The RECOMMENDED mechanism for establishing that security
   association is to use MIKEY with RTSP as defined in RFC 4567  [21].

B.1.5 SAVPF/UDP

   The RTP profile "Extended Secure RTP Profile for RTCP-based Feedback
   (RTP/SAVPF)"  [6] is an RTP profile (SAVPF) that MAY be used in RTSP
   sessions using RTP. All that is defined for AVP SHALL also apply for
   SAVPF.

   The usage of SRTP requires that a security association is
   established. The RECOMMENDED mechanism for establishing that security
   association is to use MIKEY [22] with RTSP as defined in RFC 4567
   [21].

B.2 RTP over TCP

   Transport of RTP over TCP can be done in two ways, over independent
   TCP connections using RFC 4571  [23] or interleaved in the RTSP
   control connection. In both cases the protocol SHALL be "rtp" and the
   lower layer SHALL be TCP. The profile may be any of the above
   specified ones; AVP, AVPF, SAVP or SAVPF.

B.2.1 Interleaved RTP over TCP

   The use of embedded (interleaved) binary data transported on the RTSP
   connection is possible as specified in Section 12. When using this
   declared combination of interleaved binary data the RTSP messages
   MUST be transported over TCP. TLS may or may not be used.

   One should however consider that this will result that all media
   streams go through any proxy. Using independent TCP connections can
   avoid that issue.




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B.2.2 RTP over independent TCP

   In this Appendix, we describe the sending of RTP  [16] over lower
   transport layer TCP  [12] according to "Framing Real-time Transport
   Protocol (RTP) and RTP Control Protocol (RTCP) Packets over
   Connection-Oriented Transport" [23]. This Appendix adapts the
   guidelines for using RTP over TCP within SIP/SDP  [37] to work with
   RTSP.

   A client codes the support of RTP over independent TCP by specifying
   an RTP/AVP/TCP transport option without an interleaved parameter in
   the Transport line of a SETUP request. This transport option MUST
   include the "unicast" parameter.

   If the client wishes to use RTP with RTCP, two ports (or two
   address/port pairs) are specified by the dest_addr parameter. If the
   client wishes to use RTP without RTCP, one port (or one address/port
   pair) is specified by the dest_addr parameter.  Ordering rules of
   dest_addr ports follow the rules for RTP/AVP/UDP.

   If the client wishes to play the active role in initiating the TCP
   connection, it MAY set the "setup" parameter (See section 14.45) on
   the Transport line to be "active", or it MAY omit the setup
   parameter, as active is the default. If the client signals the active
   role, the ports for all dest_addr values MUST be set to 9 (the
   discard port).

   If the client wishes to play the passive role in TCP connection
   initiation, it MUST set the "setup" parameter on the Transport line
   to be "passive". If the client is able to assume the active or the
   passive role, it MUST set the "setup" parameter on the Transport line
   to be "actpass". In either case, the dest_addr port value for RTP
   MUST be set to the TCP port number on which the client is expecting
   to receive the RTP stream connection, and the dest_addr port value
   for RTCP MUST be set to the TCP port number on which the client is
   expecting to receive the RTCP stream connection.

   If upon receipt of a non-interleaved RTP/AVP/TCP SETUP request, a
   server decides to accept this requested option, the 2xx reply MUST
   contain a Transport option that specifies RTP/AVP/TCP (without using
   the interleaved parameter, and with using the unicast parameter). The
   dest_addr parameter value MUST be echoed from the parameter value in
   the client request unless the destination address (only port) was not
   provided in which can the server MAY include the source address of
   the RTSP TCP connection with the port number unchanged.

   In addition, the server reply MUST set the setup parameter on the
   Transport line, to indicate the role the server will play in the



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   connection setup. Permissible values are "active" (if a client set
   "setup" to "passive" or "actpass") and "passive" (if a client set
   "setup" to "active" or "actpass").

   If a server sets "setup" to "passive", the "src_addr" in the reply
   MUST indicate the ports the server is willing to receive an RTP
   connection and (if the client requested an RTCP connection by
   specifying two dest_addr ports or address/port pairs) and RTCP
   connection. If a server sets "setup" to "active", the ports specified
   in "src_addr" MUST be set to 9. The server MAY use the "ssrc"
   parameter, following the guidance in 14.45. Port ordering for
   src_addr follows the rules for RTP/AVP/UDP.

   For cases when servers have a public IP-address it is RECOMMENDED
   that the server take the passive role and the client the active role.
   This help in cases when the client is behind a NAT.

   After sending (receiving) a 2xx reply for a SETUP method for a non-
   interleaved RTP/AVP/TCP media stream, the active party SHOULD
   initiate the TCP connection as soon as possible. The client SHALL NOT
   send a PLAY request prior to the establishment of all the TCP
   connections negotiated using SETUP for the session. In case the
   server receives a PLAY request in a session that has not yet
   established all the TCP connections, it SHALL respond using the 464
   "Data Transport Not Ready Yet" (Section 13.4.16) error code.

   Once the PLAY request for a media resource transported over non-
   interleaved RTP/AVP/TCP occurs, media begins to flow from server to
   client over the RTP TCP connection, and RTCP packets flow
   bidirectionally over the RTCP TCP connection. As in the RTP/UDP case,
   client to server traffic on the TCP port is unspecified by this memo.
   The packets that travel on these connections SHALL be framed using
   the protocol defined in  [23], not by the framing defined for
   interleaving RTP over the RTSP control connection defined in Section
   12.

   A successful PAUSE request for a media being transported over
   RTP/AVP/TCP pauses the flow of packets over the connections, without
   closing the connections. A successful TEARDOWN request signals that
   the TCP connections for RTP and RTCP are to be closed as soon as
   possible.

   Subsequent SETUP requests on an already-SETUP RTP/AVP/TCP URI may be
   ambiguous in the following way: does the client wish to open up new
   TCP RTP and RTCP connections for the URI, or does the client wish to
   continue using the existing TCP RTP and RTCP connections? The client
   SHOULD use the "connection" parameter (defined in 14.45) on the
   Transport line to make its intention clear in the regard (by setting



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   "connection" to "new" if new connections are needed, and by setting
   "connection" to "existing" if the existing connections are to be
   used). After a 2xx reply for a SETUP request for a new connection,
   parties should close the pre-existing connections, after waiting a
   suitable period for any stray RTP or RTCP packets to arrive.

   Below, we rewrite part of the example media on demand example shown
   in 17.1 to use RTP/AVP/TCP non-interleaved:


      C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/2.0
            CSeq: 1
            User-Agent: PhonyClient/1.2

      M->C: RTSP/2.0 200 OK
            CSeq: 1
            Server: PhonyServer/1.0
            Date: 23 Jan 1997 15:35:06 GMT
            Content-Type: application/sdp
            Content-Length: 257
            Content-Base: rtsp://example.com/twister.3gp/
            Expires: 24 Jan 1997 15:35:06 GMT

            v=0
            o=- 2890844256 2890842807 IN IP4 192.0.2.5
            s=RTSP Session
            i=An Example of RTSP Session Usage
            e=adm@example.com
            a=control: *
            a=range: npt=0-0:10:34.10
            t=0 0
            m=audio 0 RTP/AVP 0
            a=control: trackID=1

      C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/2.0
            CSeq: 2
            User-Agent: PhonyClient/1.2
            Require: play.basic
            Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9"
                       setup=active;connection=new

      M->C: RTSP/2.0 200 OK
            CSeq: 2
            Server: PhonyServer/1.0
            Transport: RTP/AVP/TCP;unicast;dest_addr=":9"/":9";
                       src_addr="192.0.2.5:9000"/"192.0.2.5:9001"
                       setup=passive;connection=new;ssrc=93CB001E
            Session: 12345678



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            Expires: 24 Jan 1997 15:35:12 GMT
            Date: 23 Jan 1997 15:35:12 GMT
            Accept-Ranges: NPT

      C->M: TCP Connection Establishment

      C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/2.0
            CSeq: 4
            User-Agent: PhonyClient/1.2
            Range: npt=0-10, npt=30-
            Session: 12345678

      M->C: RTSP/2.0 200 OK
            CSeq: 4
            Server: PhonyServer/1.0
            Date: 23 Jan 1997 15:35:14 GMT
            Session: 12345678
            Range: npt=0-10, npt=30-623.10
            RTP-Info:  url="rtsp://example.com/twister.3gp/trackID=1";
               ssrc=4F312DD8:seq=54321;rtptime=2876889



B.2.3 Handling NPT Jumps in the RTP Media Layer

   RTSP allows media clients to control selected, non-contiguous
   sections of media presentations, rendering those streams with an RTP
   media layer[16]. Such control allows jumps to be created in NPT
   timeline of the RTSP session. For example, jumps in NPT can be caused
   by multiple ranges in the range specifier of a PLAY request or
   through a "seek" opertaion on an RTSP session which involves a PLAY,
   PAUSE, PLAY scenario where a new NPT is set for the session. The
   media layer rendering the RTP stream should not be affected by jumps
   in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be
   continuous and monotonic across jumps of NPT.


        We cannot assume that the RTSP client can communicate with
        the RTP media agent, as the two may be independent
        processes.  If the RTP timestamp shows the same gap as the
        NPT, the media agent will assume that there is a pause in
        the presentation. If the jump in NPT is large enough, the
        RTP timestamp may roll over and the media agent may believe
        later packets to be duplicates of packets just played out.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.



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      C->S: PLAY rtsp://xyz/fizzle RTSP/2.0
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15

      S->C: RTSP/2.0 200 OK
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15
        RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
                  ssrc=0D12F123:seq=0;rtptime=0



   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
       . . .
      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s



   Immediately after the end of the play range, the client follows up
   with a request to PLAY from a new NPT.


   C->S: PLAY rtsp://xyz/fizzle RTSP/2.0
         CSeq: 5
         Session: abcdefg
         Range: npt=18-20;

   S->C: RTSP/2.0 200 OK
         CSeq: 5
         Session: abcdefg
         Range: npt=18-20
         RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
                   ssrc=0D12F123:seq=50;rtptime=40100



   The ensuing RTP data stream is depicted below:

      S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
      S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
       . . .
      S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s



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   In this example, first, NPT 10 through 15 is played, then the client
   request the server to skip ahead and play NPT 18 through 20. The
   first segment is presented as RTP packets with sequence numbers 0
   through 49 and timestamp 0 through 39,200. The second segment
   consists of RTP packets with sequence number 50 through 69, with
   timestamps 40,100 through 55,200. While there is a gap in the NPT,
   there is no gap in the sequence number space of the RTP data stream.

   The RTP timestamp gap is present in the above example due to the time
   it takes to perform the second play request, in this case 12.5 ms
   (100/8000). To avoid this gap in playback due to the time it takes to
   perform RTSP requests, a PLAY request with multiple ranges needs to
   be specified. That would result in the following example:


      C->S: PLAY rtsp://xyz/fizzle RTSP/2.0
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15;npt=18-20

      S->C: RTSP/2.0 200 OK
        CSeq: 4
        Session: abcdefg
        Range: npt=10-15
        RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
                  ssrc=0D12F123:seq=0;rtptime=0



   The ensuing RTP data stream is depicted below:



      S -> C: RTP packet - seq = 0,  rtptime = 0,     NPT time = 10s
      S -> C: RTP packet - seq = 1,  rtptime = 800,   NPT time = 10.1s
       . . .
      S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
      S -> C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
      S -> C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
       . . .
      S -> C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s




B.2.4 Handling RTP Timestamps after PAUSE

   During a PAUSE / PLAY interaction in an RTSP session, the duration of



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   time for which the RTP transmission was halted MUST be reflected in
   the RTP timestamp of each RTP stream. The duration can be calculated
   for each RTP stream as the time elapsed from when the last RTP packet
   was sent before the PAUSE request was received and when the first RTP
   packet was sent after the subsequent PLAY request was received. The
   duration includes all latency incurred and processing time required
   to complete the request.


        The RTP RFC [16] states that: The RTP timestamp for each
        unit[packet] would be related to the wallclock time at
        which the unit becomes current on the virtual presentation
        timeline.

   In order to satisfy the requirements of [16], the RTP timestamp space
   needs to increase continuously with real time.  While this is not
   optimal for stored media, it is required for RTP and RTCP to function
   as intended. Using a continuous RTP timestamp space allows the same
   timestamp model for both stored and live media and allows better
   opportunity to integrate both types of media under a single control.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.


   C->S: PLAY rtsp://xyz/fizzle RTSP/2.0
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15;

   S->C: RTSP/2.0 200 OK
         CSeq: 4
         Session: abcdefg
         Range: npt=10-15
         RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
                   ssrc=0D12F123:seq=0;rtptime=0



   The ensuing RTP data stream is depicted below:


      S -> C: RTP packet - seq = 0, rtptime = 0,    NPT time = 10s
      S -> C: RTP packet - seq = 1, rtptime = 800,  NPT time = 10.1s
      S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
      S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s




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   The client then sends a PAUSE request:


   C->S: PAUSE rtsp://xyz/fizzle RTSP/2.0
         CSeq: 5
         Session: abdcdefg

   S->C: RTSP/2.0 200 OK
         CSeq: 5
         Session: abcdefg
         Range: npt=10.4-15



   20 seconds elapse and then the client sends a PLAY request. In
   addition the server requires 15 ms to process the request:


   C->S: PLAY rtsp://xyz/fizzle RTSP/2.0
         CSeq: 6
         Session: abcdefg

   S->C: RTSP/2.0 200 OK
         CSeq: 6
         Session: abcdefg
         Range: npt=10.4-15
         RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
                   ssrc=0D12F123:seq=4;rtptime=164400



   The ensuing RTP data stream is depicted below:

      S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
      S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
      S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s



   First, NPT 10 through 10.3 is played, then a PAUSE is received by the
   server. After 20 seconds a PLAY is received by the server which take
   15ms to process. The duration of time for which the session was
   paused is reflected in the RTP timestamp of the RTP packets sent
   after this PLAY request.

   A client can use the RTSP range header and RTP-Info header to map NPT
   time of a presentation with the RTP timestamp.




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   Note: In RFC 2326 [25], this matter was not clearly defined and was
   misunderstood commonly. However for RTSP 2.0 it is expected that this
   will be handled correctly and no exception handling will be required.

B.2.5 RTSP / RTP Integration

   For certain datatypes, tight integration between the RTSP layer and
   the RTP layer will be necessary. This by no means precludes the above
   restrictions. Combined RTSP/RTP media clients should use the RTP-Info
   field to determine whether incoming RTP packets were sent before or
   after a seek or before or after a PAUSE.

B.2.6 Scaling with RTP

   For scaling (see Section 14.39), RTP timestamps should correspond to
   the playback timing. For example, when playing video recorded at 30
   frames/second at a scale of two and speed (Section 14.40) of one, the
   server would drop every second frame to maintain and deliver video
   packets with the normal timestamp spacing of 3,000 per frame, but NPT
   would increase by 1/15 second for each video frame.


        Note: The above scaling puts requirements on the media
        codec or a media stream to support it. For example motion
        JPEG or other non-predictive video coding can easier handle
        the above example.

B.2.7 Maintaining NPT synchronization with RTP timestamps

   The client can maintain a correct display of NPT by noting the RTP
   timestamp value of the first packet arriving after repositioning.
   The sequence parameter of the RTP-Info (Section 14.38) header
   provides the first sequence number of the next segment.

B.2.8 Continuous Audio

   For continuous audio, the server SHOULD set the RTP marker bit at the
   beginning of serving a new PLAY request or at jumps in timeline. This
   allows the client to perform playout delay adaptation.

B.2.9 Multiple Sources in an RTP Session

   Note that more than one SSRC MAY be sent in the media stream. If it
   happens all sources are expected to be rendered simultaneously.

B.2.10 Usage of SSRCs and the RTCP BYE Message During an RTSP Session

   The RTCP BYE message indicates the end of use of a given SSRC. If all



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   sources leave an RTP session, it can, in most cases, be assumed to
   have ended. Therefore, a client or server SHALL NOT send a RTCP BYE
   message until it has finished using a SSRC. A server SHOULD keep
   using a SSRC until the RTP session is terminated. Prolonging the use
   of a SSRC allows the established synchronization context associated
   with that SSRC to be used to synchronize subsequent PLAY requests
   even if the PLAY response is late.

   An SSRC collision with the SSRC that transmits media does also have
   consequences, as it will force the media sender to change its SSRC in
   accordance with the RTP specification [16]. This will result in a
   loss of synchronization context, and require any receiver to wait for
   RTCP sender reports for all media requiring synchronization before
   being able to play out synchronized. Due to these reasons a client
   joining a session should take care to not select the same SSRC as the
   server. Any SSRC signalled in the Transport header SHOULD be avoided.
   A client detecting a collision prior to sending any RTP or RTCP
   messages can also select a new SSRC.

B.3 Future Additions

   It is the intention that any future protocol or profile regarding
   both for media delivery and lower transport should be easy to add to
   RTSP. This section provides the necessary steps that needs to be
   meet.

   The following things needs to be considered when adding a new
   protocol of profile for use with RTSP:

        o The protocol or profile needs to define a name tag
          representing it. This tag is required to be a ABNF "token" to
          be possible to use in the Transport header specification.

        o The useful combinations of protocol/profile/lower-layer needs
          to be defined and for each combination declare the necessary
          parameters to use in the Transport header.

        o For new media protocols the interaction with RTSP needs to be
          addressed. One important factor will be the media
          synchronization.

   See the IANA section (21) for information how to register new
   attributes.

C Use of SDP for RTSP Session Descriptions

   The Session Description Protocol (SDP, RFC 4566 [1]) may be used to
   describe streams or presentations in RTSP. This description is



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   typically returned in reply to a DESCRIBE request on an URI from a
   server to a client, or received via HTTP from a server to a client.

   This appendix describes how an SDP file determines the operation of
   an RTSP session. SDP as is provides no mechanism by which a client
   can distinguish, without human guidance, between several media
   streams to be rendered simultaneously and a set of alternatives
   (e.g., two audio streams spoken in different languages). However the
   SDP extension "Grouping of Media Lines in the Session Description
   Protocol (SDP)" [42] may provide such functionality depending on
   need. Also future grouping semantics may in the future be developed.

C.1 Definitions

   The terms "session-level", "media-level" and other key/attribute
   names and values used in this appendix are to be used as defined in
   SDP (RFC 4566 [1]):

C.1.1 Control URI

   The "a=control:" attribute is used to convey the control URI. This
   attribute is used both for the session and media descriptions. If
   used for individual media, it indicates the URI to be used for
   controlling that particular media stream. If found at the session
   level, the attribute indicates the URI for aggregate control
   (presentation URI). The session level URI SHALL be different from any
   media level URI. The presence of a session level control attribute
   SHALL be interpreted as support for aggregated control.  The control
   attribute SHALL be present on media level unless the presentation
   only contains a single media stream, in which case the attribute MAY
   only be present on the session level.

   ABNF for the attribute is defined in section 19.3.

   Example:

     a=control:rtsp://example.com/foo



   This attribute MAY contain either relative or absolute URIs,
   following the rules and conventions set out in RFC 3986 [8].
   Implementations SHALL look for a base URI in the following order:

        1.   the RTSP Content-Base field;

        2.   the RTSP Content-Location field;




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        3.   the RTSP Request-URI.

   If this attribute contains only an asterisk (*), then the URI SHALL
   be treated as if it were an empty embedded URI, and thus inherit the
   entire base URI.

   The URI handling for SDPs from container files need special
   consideration. For example lets assume that a container file has the
   URI: "rtsp://example.com/container.mp4". Lets further assume this URI
   is the base URI, and that there is a absolute media level URI:
   "rtsp://example.com/container.mp4/trackID=2". A relative media level
   URI that resolves in accordance with RFC 3986 [8] to the above given
   media URI is: "container.mp4/trackID=2". It is usually not desirable
   to need to include in or modify the SDP stored within the container
   file with the server local name of the container file. To avoid this,
   one can modify the base URI used to include a trailing slash, e.g.
   "rtsp://example.com/container.mp4/".  In this case the relative URI
   for the media will only need to be:  "trackID=2". However this will
   also mean that using "*" in the SDP will result in control URI
   including the trailing slash, i.e.
   "rtsp://example.com/container.mp4/".


        Note: The usage of TrackID in the above is not an
        standardized form, but one example out of several similar
        strings such as TrackID, Track_ID, StreamID that is used by
        different server vendors to indicate a particular piece of
        media inside a container file.

C.1.2 Media Streams

   The "m=" field is used to enumerate the streams. It is expected that
   all the specified streams will be rendered with appropriate
   synchronization. If the session is over multicast, the port number
   indicated SHOULD be used for reception. The client MAY try to
   override the destination port, through the Transport header. The
   servers MAY allow this, the response will indicate if allowed or not.
   If the session is unicast, the port number is the ones RECOMMENDED by
   the server to the client, about which receiver ports to use; the
   client MUST still include its receiver ports in its SETUP request.
   The client MAY ignore this recommendation.  If the server has no
   preference, it SHOULD set the port number value to zero.

   The "m=" lines contain information about what transport protocol,
   profile, and possibly lower-layer is to be used for the media stream.
   The combination of transport, profile and lower layer, like
   RTP/AVP/UDP needs to be defined for how to be used with RTSP.  The
   currently defined combinations are defined in section B, further



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   combinations MAY be specified.

   Usage of grouping of media lines [42] to determine which media lines
   should or should not be included in a RTSP session is unspecified.

   Example:

     m=audio 0 RTP/AVP 31



C.1.3 Payload Type(s)

   The payload type(s) are specified in the "m=" line. In case the
   payload type is a static payload type from RFC 3551 [2], no other
   information may be required. In case it is a dynamic payload type,
   the media attribute "rtpmap" is used to specify what the media is.
   The "encoding name" within the "rtpmap" attribute may be one of those
   specified in RFC 3551 (Sections 5 and 6), or an MIME type registered
   with IANA, or an experimental encoding as specified in SDP (RFC 4566
   [1]). Codec-specific parameters are not specified in this field, but
   rather in the "fmtp" attribute described below.

C.1.4 Format-Specific Parameters

   Format-specific parameters are conveyed using the "fmtp" media
   attribute. The syntax of the "fmtp" attribute is specific to the
   encoding(s) that the attribute refers to. Note that some of the
   format specific parameters may be specified outside of the fmtp
   parameters, like for example the "ptime" attribute for most audio
   encodings.

C.1.5 Directionality of media stream

   The SDP attributes "a=sendrecv", "a=recvonly" and "a=sendonly"
   provides instructions on which direction the media streams flow
   within a session. When using RTSP the SDP can be delivered to a
   client using either RTSP DESCRIBE or a number of RTSP external
   methods, like HTTP, FTP, and email. Based on this the SDP applies to
   how the RTSP client will see the complete session. Thus for media
   streams delivered from the RTSP server to the client would be given
   the "a=recvonly" attribute. A a=sendonly in a SDP provided to the
   client would indicate that a media stream would be sent from the
   client to the server. "a=sendrecv" would indicate media transmission
   occurs in both directions between client and server.

   The direction attributes are not commonly used in SDPs for RTSP, but
   may occur. To reflect this reality the following rules are defined.



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   "a=recvonly" in a SDP provided to the RTSP client SHALL indicate that
   media delivery will only occur in the direction from the server to
   the client. Thus an RTSP client shall initiate any RTSP session in
   the "PLAY" mode. In SDP provided to the RTSP client that lacks any of
   the directionality attributes (a=recvonly, a=sendonly, a=sendrecv)
   SHALL behave as if the "a=recvonly" attribute was received. Note that
   this overrules the normal default rule defined in SDP [1]. The usage
   of "a=sendonly" or "a=sendrecv" is not defined, nor is the
   interpretation of SDP by other entities than the RTSP client.

C.1.6 Range of Presentation

   The "a=range" attribute defines the total time range of the stored
   session or an individual media. Non-seekable live sessions can be
   indicated, while the length of live sessions can be deduced from the
   "t" and "r" SDP parameters.

   The attribute is both a session and a media level attribute. For
   presentations that contains media streams of the same durations, the
   range attribute SHOULD only be used at session-level. In case of
   different length the range attribute MUST be given at media level for
   all media, and SHOULD NOT be given at session level. If the attribute
   is present at both media level and session level the media level
   values SHALL be used.

   Note: Usually one will specify the same length for all media, even if
   there isn't media available for the full duration on all media.
   However that requires that the server accepts PLAY requests within
   that range.

   Servers SHALL take care to provide RTSP Range (see Section 14.34)
   values that are consistent with what is presented in the SDP for the
   content. There are no reason for non dynamic content, like media
   clips provided on demand to have inconsistent values. Inconsistent
   values between the SDP and the actual values for the content handled
   by the server is likely to generate some failure, like 457 "Invalid
   Range", in case the client uses PLAY requests with a Range header. In
   case the content is dynamic in length and it is infeasible to provide
   a correct value in the SDP the server is recommended to describe this
   as non-seekable content (see below). The server MAY override that
   property in the response to a PLAY request using the correct values
   in the Range header.

   The unit is specified first, followed by the value range. The units
   and their values are as defined in Section 3.4, 3.5 and 3.6 and MAY
   be extended with further formats. Any open ended range (start-), i.e.
   without stop range, is of unspecified duration and SHALL be
   considered as non-seekable content unless this property is



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   overridden. Multiple instances carrying different clock formats MAY
   be included at either session or media level.

   ABNF for the attribute is defined in section 19.3.

   Examples:

     a=range:npt=0-34.4368
     a=range:clock=19971113T2115-19971113T2203
     Non seekable stream of unknown duration:
     a=range:npt=0-



C.1.7 Time of Availability

   The "t=" field MUST contain suitable values for the start and stop
   times for both aggregate and non-aggregate stream control.  The
   server SHOULD indicate a stop time value for which it guarantees the
   description to be valid, and a start time that is equal to or before
   the time at which the DESCRIBE request was received. It MAY also
   indicate start and stop times of 0, meaning that the session is
   always available.

   For sessions that are of live type, i.e. specific start time, unknown
   stop time, likely unseekable, the "t=" and "r=" field SHOULD be used
   to indicate the start time of the event. The stop time SHOULD be
   given so that the live event will have ended at that time, while
   still not be unnecessary long into the future.

C.1.8 Connection Information

   In SDP, the "c=" field contains the destination address for the media
   stream. For on-demand unicast streams and some multicast streams, the
   destination address MAY be specified by the client via the SETUP
   request, thus overriding any specified address. To identify streams
   without a fixed destination address, where the client is required to
   specify a destination address, the "c=" field SHOULD be set to a null
   value. For addresses of type "IP4", this value SHALL be "0.0.0.0",
   and for type "IP6", this value SHALL be "0:0:0:0:0:0:0:0", i.e. the
   unspecified address according to RFC 3513 [24].

C.1.9 Entity Tag

   The optional "a=etag" attribute identifies a version of the session
   description. It is opaque to the client. SETUP requests may include
   this identifier in the If-Match field (see section 14.24) to only
   allow session establishment if this attribute value still corresponds



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   to that of the current description.  The attribute value is opaque
   and may contain any character allowed within SDP attribute values.

   ABNF for the attribute is defined in section 19.3.

   Example:

     a=etag:158bb3e7c7fd62ce67f12b533f06b83a




        One could argue that the "o=" field provides identical
        functionality. However, it does so in a manner that would
        put constraints on servers that need to support multiple
        session description types other than SDP for the same piece
        of media content.

C.2 Aggregate Control Not Available

   If a presentation does not support aggregate control no session level
   "a=control:" attribute is specified. For a SDP with multiple media
   sections specified, each section will have its own control URI
   specified via the "a=control:" attribute.

   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 192.0.2.56
   s=I came from a web page
   e=adm@example.com
   c=IN IP4 0.0.0.0
   t=0 0
   m=video 8002 RTP/AVP 31
   a=control:rtsp://audio.com/movie.aud
   m=audio 8004 RTP/AVP 3
   a=control:rtsp://video.com/movie.vid



   Note that the position of the control URI in the description implies
   that the client establishes separate RTSP control sessions to the
   servers audio.com and video.com

   It is recommended that an SDP file contains the complete media
   initialization information even if it is delivered to the media
   client through non-RTSP means. This is necessary as there is no
   mechanism to indicate that the client should request more detailed



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   media stream information via DESCRIBE.

C.3 Aggregate Control Available

   In this scenario, the server has multiple streams that can be
   controlled as a whole. In this case, there are both a media-level
   "a=control:" attributes, which are used to specify the stream URIs,
   and a session-level "a=control:" attribute which is used as the
   Request-URI for aggregate control. If the media-level URI is
   relative, it is resolved to absolute URIs according to Section C.1.1
   above.

   Example:


   C->M: DESCRIBE rtsp://example.com/movie RTSP/2.0
         CSeq: 1

   M->C: RTSP/2.0 200 OK
         CSeq: 1
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Base: rtsp://example.com/movie/
         Content-Length: 228

         v=0
         o=- 2890844256 2890842807 IN IP4 192.0.2.211
         s=I contain
         i=<more info>
         e=adm@example.com
         c=IN IP4 0.0.0.0
         t=0 0
         a=control:*
         m=video 8002 RTP/AVP 31
         a=control:trackID=1
         m=audio 8004 RTP/AVP 3
         a=control:trackID=2



   In this example, the client is required to establish a single RTSP
   session to the server, and uses the URIs
   rtsp://example.com/movie/trackID=1 and
   rtsp://example.com/movie/trackID=2 to set up the video and audio
   streams, respectively. The URI rtsp://example.com/movie/ , which is
   resolved from the "*", controls the whole presentation (movie).

   A client is not required to issues SETUP requests for all streams



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   within an aggregate object. Servers should allow the client to ask
   for only a subset of the streams.

C.4 RTSP external SDP delivery

   There are some considerations that needs to be made when the session
   description is delivered to client outside of RTSP, for example in
   HTTP or email.

   First of all the SDP needs to contain absolute URIs, relative will in
   most cases not work as the delivery will not correctly forward the
   base URI. And as SDP might be temporarily stored on file system
   before being loaded into an RTSP capable client, thus if possible to
   transport the base URI it still would need to be merged into the
   file.

   The writing of the SDP session availability information, i.e. "t="
   and "r=", needs to be carefully considered. When the SDP is fetched
   by the DESCRIBE method it is with very high probability that the it
   is valid. However the same are much less certain for SDPs distributed
   using other methods. Therefore the publisher of the SDP should take
   care to follow the recommendations about availability in the SDP
   specification [1].

D Minimal RTSP implementation

   This section defines the minimal implementation requirements for RTSP
   agents.

D.1 Minimal Core Implementation

   The minimal core implementation is what is required to negotiate the
   usage of any other features. A minimal core implementation is not
   supporting any other feature set will be useless as the minimal
   implementation doesn't deliver any service. All feature sets SHALL
   include the minimal core.

   A minimal core implementation SHALL support the following
   functionalities:

        o Establishing a connection between RTSP agents using TCP.

        o Implement the reception and response to the OPTIONS method.

        o Implement the handling of all headers mandatory or conditional
          in regards to the usage of the OPTIONS method. See tables 9
          and 10. This include at least the capability to ignore unknown
          headers.



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        o Implement the headers related to capability negotiation and
          exchange:

          - Require

          - Supported

          - Proxy-Require

          - Proxy-Supported

          - Unsupported

D.2 Recommended Core Implementation

   A RTSP Agent is also RECOMMENDED to support the following:

        o RTSP basic and digest authentication: The 401 response, the
          WWW-Authenticate and Authorization headers, and both Basic and
          Digest authentication methods as defined by [10].

        o Secure RTSP message transport as specified by section D.4.

D.3 The Basic Playback Feature Support

   This section defines what is required to be supported for clients,
   proxies and servers to be supporting the "play.basic" feature-tag.

D.3.1 Client

   A play.basic supporting client SHALL implement the following:

        o The RTSP methods as required by Table 7.

        o All the RTSP headers that are required required or conditional
          in requests or responses to method required to be supported
          according to Tables 9, 10, 11, and 12 and in addition the
          following headers:

          - Content-Base

          - Content-Encoding and at least the Identity method.

          - Content-Location

          - Location

          - Range and the npt time format)



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          - RTP-Info

        o Handling of all Status code categories.

        o Media delivery using RTP/AVP over UDP.

   A play.basic supporting client is also RECOMMENDED to support the
   following:

        o Expires header

        o From header

D.3.2 Server

   A play.basic supporting server SHALL implement the following:

        o The RTSP methods as required by Table 7.

        o Reception and responding to all headers specified in Section
          14. The implementation of functionality provided by all these
          header with the following exceptions:

          - Scale

          - Speed

          - Blocksize

        o Media delivery using RTP/AVP over UDP.

   A play.basic supporting Server is also RECOMMENDED to support the
   following:

D.3.3 Proxy

   A play.basic supporting proxy SHALL implement the following:

        o At least passing through all the methods listed in Table 7.

        o The handling of all RTSP headers that are required to be
          handled by the server and clients supporting "play.basic" and
          in addition the following headers:

          - Cache-Control

          - Expires




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          - Via

D.4 Secure Transport

   Any Client, Proxy or Server supporting secure transport of RTSP
   messages and usage of the "rtsps" URI scheme SHALL implement; The
   Accept-Credentials and Connection-Credentials headers; TLS over TCP.

E Requirements for Unreliable Transport of RTSP messages

   This section provides any one intending to define how to transport of
   RTSP messages over a unreliable transport protocol with some
   information learned by the attempt in RFC 2326 [25]. RFC 2326 define
   both an URI scheme and some basic functionality for transport of RTSP
   messages over UDP, however it was not sufficient for reliable usage
   and successful interoperability.

   The RTSP scheme defined for unreliable transport of RTSP messages was
   "rtspu". It has been reserved by this specification as at least one
   commercial implementation exist, thus avoiding any collisions in the
   name space.

   The following considerations should exist for operation of RTSP over
   an unreliable transport protocol:

        o Request shall be acknowledged by the receiver. If there is no
          acknowledgement, the sender may resend the same message after
          a timeout of one round-trip time (RTT). Any retransmissions
          due to lack of acknowledgement must carry the same sequence
          number as the original request.

        o The round-trip time can be estimated as in TCP (RFC 1123)
          [43], with an initial round-trip value of 500 ms.  An
          implementation may cache the last RTT measurement as the
          initial value for future connections.

        o If RTSP is used over a small-RTT LAN, standard procedures for
          optimizing initial TCP round trip estimates, such as those
          used in T/TCP (RFC 1644) [44], can be beneficial.

        o The Timestamp header (Section 14.44) is used to avoid the
          retransmission ambiguity problem [45] and obviates the need
          for Karn's algorithm.

        o The registered default port for RTSP over UDP for the server
          is 554.

        o RTSP messages can be carried over any lower-layer transport



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          protocol that is 8-bit clean.

        o RTSP messages are vulnerable to bit errors and should not be
          subjected to them.

        o Source authentication, or at least validation that RTSP
          messages comes from the same entity becomes extremely
          important, as session hijacking may be substantially easier
          for RTSP message transport using an unreliable protocol like
          UDP than for TCP.

   There exist two RTSP headers thats primarily are intended for being
   used by the unreliable handling of RTSP messages and which will be
   maintained:

        CSeq See section 14.19

        Timestamp See section 14.44

F Backwards Compatibility Considerations

   This section contains notes on issues about backwards compatibility
   with clients or servers being implemented according to RFC 2326 [25].
   Note that there exist no requirement to implement RTSP 1.0, in fact
   we recommend against it as it is difficult to do in an interoperable
   way.

   A server implementing RTSP/2.0 MUST include a RTSP-Version of
   RTSP/2.0 in all responses to requests containing RTSP-Version
   RTSP/2.0. If a server receives a RTSP/1.0 request, it MAY respond
   with a RTSP/1.0 response if it chooses to support RFC 2326. If the
   server chooses not to support RFC 2326, it SHOULD respond with a 505
   (RTSP Version not supported) status code. A server MUST NOT respond
   to a RTSP-Version RTSP/1.0 request with a RTSP-Version RTSP/2.0
   response.

   Clients implementing RTSP/2.0 MAY use an OPTIONS request with a
   RTSP-Version of 2.0 to determine whether a server supports RTSP/2.0.
   If the server responds with either a RTSP-Version of 1.0 or a status
   code of 505 (RTSP Version not supported), the client will have to use
   RTSP/1.0 requests if it chooses to support RFC 2326.

   In RFC 2326, receivers were advised to be prepared to also interpret
   CR and LF by themselves as line terminators in addition to CRLF. If a
   server or client wishes to support RFC 2326, it should treat a CR or
   LF by itself as a CRLF.

F.1 Play Request in Play mode



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   The behavior in the server when a Play is received in Play mode has
   changed (Section 11.4). In RFC 2326, the new PLAY request would be
   queued until the current Play completed. Any new PLAY request now
   take effect immediately replacing the previous request.

F.2 Using Persistent Connections

   Some server implementations of RFC 2326 maintain a one-to-one
   relationship between a connection and an RTSP session. Such
   implementations require clients to use a persistent connection to
   communicate with the server and when a client closes its connection,
   the server may remove the RTSP session. This is worth noting if a
   RTSP 2.0 client also supporting 1.0 connects to a 1.0 server.

G Open Issues

   This section contains a list of open issues that still needs to be
   resolved. However also any open issues in the bug tracker at
   http://rtspspec.sourceforge.net should also be considered.

        1.   Should the SMPTE range format be updated to support the 50
             and 60 frames per second modes?

        2.   Should we define a recommended format for error message
             bodies?

        3.   Today there is no recommended or required format for 300
             response entities containing URI lists. Should one be
             defined?

        4.   Should the dest_addr parameter in the Transport header in
             responses include the destination used by the server?

        5.   Should a IPv6 multicast scope parameter for the Transport
             header be defined? This would be similar to TTL.

        6.   The Expires header (Section 14.22 contains the below
             paragraph:

             Expires header field with a date value of some time in the
             future on a media stream that otherwise would by default be
             non- cacheable indicates that the media stream is
             cacheable, unless indicated otherwise by a Cache-Control
             header field (Section 14.10).

             Is there any purpose for this in RTSP, or could we remove
             this statement and instead rely on the Cache-Control
             header?



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        7.   Should proxies strip out the credentials for themselves
             when forwarding messages with Accept-Credentials?

        8.   Is Session ID combined with TLS a sufficient mechanism to
             prevent hijacking?

        9.   Move to start TLS mechanism like the one defined in RFC
             2817?

        10.  Look into the GRID communities proxy-certs and see how this
             relates to the current TLS proxy solution.

        11.  Resolve Eric Rescorlas security comments on the Proxy TLS
             solution:

             1.   There doesn't seem to be any way to communicate your
                  cipher suite preferences.

             2.   I don't see how certificate-based client
                  authentication works. Is it supposed to?

             3.   You need to provide the entire cert chain in
                  Connection-Credentials, not just the certificate.

        12.  Consider to switch to SHA256 instead of SHA1 for the digest
             over the DER encoded certs.

        13.  Resolve the following Stephen Farrel issue:  "C. I don't
             understand how the client-side proxies can be expected to
             know enough about proxies existing toward the server. If
             they don't then I'm not sure how they can be expected to
             make any decision that's better than would be the case were
             policy to be dealt with solely on a hop-by-hop basis. Maybe
             I'm missing something that can provide that information?"

        14.  Resolve the following Stephen Farrel issue:  "D. The "User"
             policy model is that a client presents acceptable name/URIs
             and digests to the proxy. TLS doesn't really provide a way
             for that proxy, as a client, to ask the server for the
             "right" certificate, so I suspect there's a gap here
             that'll be hard to fill. (If the client imposed a
             constraint as to the root-CA that had to be used then
             that'd map to the next TLS connection, but maybe it'd be
             too coarse-grained?)"

H Changes

   Compared to RTSP 1.0 (RFC 2326), the below changes has been made when



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   defining RTSP 2.0. Note that this list does not reflect minor changes
   in wording or correction of typographical errors.

        o The Transport header has been changed in the following way:

          - The ABNF has been changed to define that extensions are
            possible, and that unknown extension parameters are to be
            ignored.

          - To prevent backwards compatibility issues, any extension or
            new parameter requires the usage of a feature-tag combined
            with the Require header.

          - Syntax unclarities with the Mode parameter has been
            resolved.

          - Syntax error with ";" for multicast and unicast has been
            resolved.

          - Two new addressing parameters has been defined, src_addr and
            dest_addr. These replaces the parameters "port",
            "client_port", "server_port", "destination", "source".

          - Support for IPv6 explicit addresses in all address fields
            has been included.

          - To handle URI definitions that contain ";" or "," a quoted
            URI format has been introduced and is required.

          - Defined IANA registries for the transport headers
            parameters, transport-protocol, profile, lower-transport,
            and mode.

          - The transport headers interleaved parameter's text was made
            more strict and use formal requirements levels. It was also
            clarified that the interleaved channels are symmetric and
            that it is the server that sets the channel numbers.

          - It has been clarified that the client can't request of the
            server to use a certain RTP SSRC, using a request with the
            transport parameter SSRC.

          - Syntax definition for SSRC has been clarified to require
            8HEX.  It has also been extend to allow multiple values for
            clients supporting this version.

          - Clarified the text on the transport headers "dest_addr"
            parameters regarding what security precautions the server is



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            required to perform.

        o The Range formats has been changed in the following way:

          - The NPT format has been given a initial NPT identifier that
            must now be used.

          - All formats now support initial open ended formats of type
            "npt=-10".

        o RTSP message handling has been changed in the following way:

          - RTSP messages now uses URIs rather then URLs.

          - It has been clarified that a 4xx message due to missing CSeq
            header shall be returned without a CSeq header.

          - Rules for how to handle timing out RTSP messages has been
            added.

        o The HTTP references has been updated to RFC 2616 and RFC 2617.
          This has resulted in that the Public, and the Content-Base
          header needed to be defined in the RTSP specification. Known
          effects on RTSP due to HTTP clarifications:

          - Content-Encoding header can include encoding of type
            "identity".

        o The state machine section has completely been rewritten. It
          includes now more details and are also more clear about the
          model used.

        o A IANA section has been included with contains a number of
          registries and their rules. This will allow us to use IANA to
          keep track of RTSP extensions.

        o Than transport of RTSP messages has seen the following
          changes:

          - The use of UDP for RTSP message transport has been
            deprecated due to missing interest and to broken
            specification.

          - The rules for how TCP connections is to be handled has been
            clarified. Now it is made clear that servers should not
            close the TCP connection unless they have been unused for
            significant time.




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          - Strong recommendations why server and clients should use
            persistent connections has also been added.

          - There is now a requirement on the servers to handle non-
            persistent connections as this provides fault tolerance.

          - Added wording on the usage of Connection:Close for RTSP.

          - specified usage of TLS for RTSP messages, including a scheme
            to approve a proxies TLS connection to the next hop.

        o The following header related changes have been made:

          - Accept-Ranges response header is added. This header
            clarifies which range formats that can be used for a
            resource.

          - Changed the Range header to allow multiple ranges for
            creating editing list.

          - Fixed the missing definitions for the Cache-Control header.
            Also added to the syntax definition the missing delta-
            seconds for max-stale and min-fresh parameters.

          - Put requirement on CSeq header that the value is increased
            by one for each new RTSP request. A Recommendation to start
            at 1 has also been added.

          - Added requirement that the Date header must be used for all
            messages with entity and the Server should always include
            it.

          - Removed possibility of using Range header with Scale header
            to indicate when it is to be activated, since it can't work
            as defined. Also added rule that lack of Scale header in
            response indicates lack of support for the header. Feature-
            tags for scaled playback has been defined.

          - The Speed header must now be responded to indicate support
            and the actual speed going to be used. A feature-tag is
            defined. Notes on congestion control was also added.

          - The Supported header was borrowed from SIP to help with the
            feature negotiation in RTSP.

          - Clarified that the Timestamp header can be used to resolve
            retransmission ambiguities.




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          - The Session header text has been expanded with a explanation
            on keep alive and which methods to use. SET_PARAMETER is now
            recommended to use if only keep-alive within RTSP is
            desired.

          - It has been clarified how the Range header formats is used
            to indicate pause points in the PAUSE response.

          - Clarified that RTP-Info URIs that are relative, uses the
            Request-URI as base URI. Also clarified that used URI must
            be that one that was used in the SETUP request. They are now
            also required to be quoted. The header also expresses the
            SSRC for the provided RTP timestamp and sequence number
            values.

          - Added text that requires the Range to always be present in
            PLAY responses. Clarified what should be sent in case of
            live streams.

          - The headers table has been updated using a structured
            borrowed from SIP. Those tables carries much more
            information and should provide a good overview of the
            available headers.

          - It has been is clarified that any message with a message
            body is required to have a Content-Length header. This was
            the case in RFC 2326 but could be misinterpreted.

          - To resolve functionality around ETag. The ETag and If-None-
            Match header has been added from HTTP with necessary
            clarification in regards to RTSP operation.

          - Imported the Public header from HTTP RFC 2068 [39] since it
            has been removed from HTTP due to lack of use. Public is
            used quite frequently in RTSP.

          - Clarified rules for populating the Public header so that it
            is an intersection of the capabilities of all the RTSP
            agents in a chain.

        o The Protocol Syntax has been changed in the following way:

          - All BNF definitions are updated according to the rules
            defined in RFC 4234 [4] and has been gathered in a separate
            section 19.

          - The BNF for the User-Agent and Server headers has been
            corrected so now only the description is in the HTTP



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            specification.

          - Some definitions in the introduction regarding the RTSP
            session has been changed.

          - The protocol has been made fully IPv6 capable. Certain of
            the functionality, like using explicit IPv6 addresses in
            fields requires that the protocol support this updated
            specification.

          - Added a fragment part to the RTSP URI. This seem to be
            indicated by the note below the definition however it was
            not part of the BNF.

          - The CHAR rule has been changed to exclude NULL.

        o The Status codes has been changed in the following way:

          - The use of status code 303 "See Other" has been deprecated
            as it does not make sense to use in RTSP.

          - When sending response 451 and 458 the response body should
            contain the offending parameters.

          - Clarification on when a 3rr redirect status code can be
            received has been added. This includes receiving 3rr as a
            result of request within a established session. This
            provides clarification to a previous unspecified behavior.

          - Removed the 201 (Created) and 250 (Low On Storage Space)
            status codes as they are only relevant to recording, which
            is deprecated.

        o The following functionality has been deprecated from the
          protocol:

          - The use of Queued Play.

          - The use of PLAY method for keep-alive in play state.

          - The RECORD and ANNOUNCE methods and all related
            functionality. Some of the syntax has been removed.

          - The possibility to use timed execution of methods with the
            time parameter in the Range header.

          - The description on how rtspu works is not part of the core
            specification and will require external description. Only



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            that it exist is defined here and some requirements for the
            the transport is provided.

        o The following changes has been made in relation to methods:

          - The OPTIONS method has been clarified with regards to the
            use of the Public and Allow headers.

          - The RECORD and ANNOUNCE methods are removed as they are
            lacking implementation and not considered necessary in the
            core specification. Any work on these methods should be done
            as a extension document to RTSP.

          - Added text clarifying the usage of SET_PARAMETER for keep-
            alive and usage without any body.

          - PLAY method is now allowed to be pipelined with the
            pipelining of one or more SETUP requests following the
            initial that generates the session for aggregated control.

        o Wrote a new section about how to setup different media
          transport alternatives and their profiles, and lower layer
          protocols. This resulted that the appendix on RTP interaction
          was moved there instead in the part describing RTP. The
          section also includes guidelines what to think of when writing
          usage guidelines for new protocols and profiles.

        o Setup and usage of independent TCP connections for transport
          of RTP has been specified.

        o Added a new section describing the available mechanisms to
          determine if functionality is supported, called "Capability
          Handling". Renamed option-tags to feature-tags.

        o Added a contributors section with people who has contribute
          actual text to the specification.

        o Added a section Use Cases that describes the major use cases
          for RTSP.

        o Clarified the usage of a=range and how to indicate live
          content that are not seekable with this header.

        o Text specifying the special behavior of PLAY for live content.


H.1 Changes needing to be updated




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 o The minimal implementation specification has been changed:

   - Required Timestamp, Via, and Unsupported headers for a minimal
     server implementation.

   - Recommended that Cache-Control, Expires and Date headers be
     supported by server implementations.

I Author Addresses

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   electronic mail: schulzrinne@cs.columbia.edu

   Anup Rao
   Cisco
   USA
   electronic mail: anrao@cisco.com

   Rob Lanphier
   Seattle, WA, USA
   electronic mail: robla@robla.net

   Magnus Westerlund
   Ericsson AB, EAB/TVA/A
   Torshamsgatan 23
   SE-164 80 STOCKHOLM
   SWEDEN
   electronic mail: magnus.westerlund@ericsson.com

   Aravind Narasimhan
   Overture Computing Corp.,
   East Windsor, NJ 08520
   USA
   electronic mail: aravind.narasimhan@gmail.com

J Contributors

   The following people have made written contributions that were
   included in the specification:

        o Tom Marshall contributed text on the usage of 3rr status
          codes.




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        o Thomas Zheng contributed text on the usage of the Range in
          PLAY responses.

        o Sean Sheedy contributed text on the timeout behavior of RTSP
          messages and connections, and the 463 status code.

        o Fredrik Lindholm contributed text about the RTSP security
          framework.

        o John Lazzaro contributed the text for RTP over Independent
          TCP.

   The following people have provided detailed comments on updated
   versions of this specification:

        o Stephan Wenger

K Acknowledgements

   This draft is based on the functionality of the original RTSP draft
   submitted in October 1996. It also borrows format and descriptions
   from HTTP/1.1.

   This document has benefited greatly from the comments of all those
   participating in the MMUSIC-WG. In addition to those already
   mentioned, the following individuals have contributed to this
   specification:

   Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
   Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
   Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
   Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
   John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,
   Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
   Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
   Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
   Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,
   Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
   Chesire, David Walker, Geetha Srikantan, Stephan Wenger, Pekka Pessi,
   Jae-Hwan Kim, Holger Schmidt, Stephen Farrell and Mela Martti.

L Normative References

   [1] M. Handley, V. Jacobson, and C. Perkins, "Sdp: Session
   description protocol," RFC 4566, Internet Engineering Task Force,
   July 2006.

   [2] H. Schulzrinne and S. Casner, "RTP profile for audio and video



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   conferences with minimal control," RFC 3551, Internet Engineering
   Task Force, July 2003.

   [3] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, L. Masinter, P.
   Leach, and T. Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1,"
   RFC 2616, Internet Engineering Task Force, June 1999.

   [4] D. Crocker, "Augmented bnf for syntax specifications: Abnf," RFC
   4234, Internet Engineering Task Force, Oct. 2005.

   [5] S. Bradner, "Key words for use in rfcs to indicate requirement
   levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.

   [6] e. a. J. Ott, "Extended secure rtp profile for rtcp-based
   feedback (rtp/savpf)," internet draft, Internet Engineering Task
   Force, Oct. 2006.  Work in progress.

   [7] M. Duerst and M. Suignard, "Internationalized resource
   identifiers (iris)," RFC 3987, Internet Engineering Task Force, Jan.
   2005.

   [8] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
   identifier (URI): generic syntax," RFC 3986, Internet Engineering
   Task Force, Jan.  2005.

   [9] T. Dierks and E. Rescorla, "The transport layer security (tls)
   protocol version 1.1," RFC 4346, Internet Engineering Task Force,
   Apr. 2006.

   [10] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. Leach,
   A. Luotonen, and L. Stewart, "HTTP authentication: Basic and digest
   access authentication," RFC 2617, Internet Engineering Task Force,
   June 1999.

   [11] J. Postel, "User datagram protocol," RFC 768, Internet
   Engineering Task Force, Aug. 1980.

   [12] J. Postel, "Transmission control protocol," RFC 793, Internet
   Engineering Task Force, Sept. 1981.

   [13] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
   3629, Internet Engineering Task Force, Nov. 2003.

   [14] R. Housley, W. Polk, W. Ford, and D. Solo, "Internet X.509
   public key infrastructure certificate and certificate revocation list
   (CRL) profile," RFC 3280, Internet Engineering Task Force, Apr. 2002.

   [15] NIST, "Fips pub 180-1:secure hash standard," tech. rep.,



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   National Institute of Standards and Technology, Apr. 1995.

   [16] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
   a transport protocol for real-time applications," RFC 3550, Internet
   Engineering Task Force, July 2003.

   [17] E. Rescorla, "HTTP over TLS," RFC 2818, Internet Engineering
   Task Force, May 2000.

   [18] T. Narten and H. Alvestrand, "Guidelines for writing an IANA
   considerations section in rfcs," RFC 2434, Internet Engineering Task
   Force, Oct. 1998.

   [19] J. O. et al., "Extended rtp profile for real-time transport
   control protocol (rtcp)-based feedback (rtp/avpf)," RFC 4585,
   Internet Engineering Task Force, July 2006.

   [20] M. Baugher, D. McGrew, M. Naslund, E. Carrara, and K. Norrman,
   "The secure real-time transport protocol (SRTP)," RFC 3711, Internet
   Engineering Task Force, Mar. 2004.

   [21] J. A. et al., "Key management extensions for session description
   protocol (sdp) and real time streaming protocol (rtsp)," RFC 4567,
   Internet Engineering Task Force, July 2006.

   [22] J. Arkko, E. Carrara, F. Lindholm, M. Naslund, and K. Norrman,
   "MIKEY:  multimedia internet keying," RFC 3830, Internet Engineering
   Task Force, Aug.  2004.

   [23] J. Lazzaro, "Framing real-time transport protocol (rtp) and rtp
   control protocol (rtcp) packets over connection-oriented transport,"
   RFC 4571, Internet Engineering Task Force, July 2006.

   [24] R. Hinden and S. Deering, "Internet protocol version 6 (ipv6)
   addressing architecture," RFC 3513, Internet Engineering Task Force,
   Apr. 2003.

M Informative References

   [25] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [26] F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
   "Internationalization of the hypertext markup language," RFC 2070,
   Internet Engineering Task Force, Jan.  1997.

   [27] ISO/IEC, "Information technology -- generic coding of moving



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   pictures and associated audio information: Systems," Published
   standard ISO 13818-1:2000, International Organization for
   Standardization ISO/IEC, Geneva, Switzerland, Oct. 2000.

   [28] H. Schulzrinne, "A comprehensive multimedia control architecture
   for the Internet," in Proc. International Workshop on Network and
   Operating System Support for Digital Audio and Video (NOSSDAV), (St.
   Louis, Missouri), May 1997.

   [29] International Telecommunication Union, "Visual telephone systems
   and equipment for local area networks which provide a non-guaranteed
   quality of service," Recommendation H.323, Telecommunication
   Standardization Sector of ITU, Geneva, Switzerland, May 1996.

   [30] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
   RFC 1961, Internet Engineering Task Force, June 1996.

   [31] J. Miller, P. Resnick, and D. Singer, "Rating services and
   rating systems (and their machine readable descriptions),"
   Recommendation REC-PICS-services-961031, W3C (World Wide Web
   Consortium), Boston, Massachusetts, Oct. 1996.

   [32] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
   distribution label syntax and communication protocols,"
   Recommendation REC-PICS-labels-961031, W3C (World Wide Web
   Consortium), Boston, Massachusetts, Oct. 1996.

   [33] D. Mills, "Network time protocol (version 3) specification,
   implementation," RFC 1305, Internet Engineering Task Force, Mar.
   1992.

   [34] ISO/IEC, "Information technology -- generic coding of moving
   pictures and associated audio informaiton -- part 6: extension for
   digital storage media and control," Draft International Standard ISO
   13818-6, International Organization for Standardization ISO/IEC
   JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.

   [35] ISO/IEC, "Data elements and interchange formats -- information
   interchange -- representation of dates and times," Published standard
   ISO 8601, International Organization for Standardization ISO/IEC,
   Geneva, Switzerland, Dec. 2000.

   [36] S. Josefsson and I. Ed., "The base16, base32, and base64 data
   encodings," RFC 3548, Internet Engineering Task Force, July 2003.

   [37] D. Yon and G. Camarillo, "Tcp-based media transport in the
   session description protocol (sdp)," RFC 4145, Internet Engineering
   Task Force, Sept. 2005.



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   [38] M. Westerlund and T. Zeng, "How to make real-time streaming
   protocol (rtsp) traverse network address translators (nat) and
   interact with firewalls.," internet draft, Internet Engineering Task
   Force, Oct. 2005.  Work in progress.

   [39] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-
   Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
   Engineering Task Force, Jan. 1997.

   [40] Third Generation Partnership Project (3GPP), "Transparent end-
   to-end packet-switched streaming service (pss); protocols and
   codecs," Technical Specification 26.234, Third Generation Partnership
   Project (3GPP), Dec. 2002.

   [41] T. Hansen, T. Hardie, and L. Masinter, "Guidelines and
   registration procedures for new uri schemes," RFC 4395, Internet
   Engineering Task Force, Feb. 2006.

   [42] G. Camarillo, G. Eriksson, J. Holler, and H. Schulzrinne,
   "Grouping of media lines in the session description protocol (SDP),"
   RFC 3388, Internet Engineering Task Force, Dec. 2002.

   [43] "Requirements for internet hosts - application and support," RFC
   1123, Internet Engineering Task Force, Oct. 1989.

   [44] R. Braden, "T/TCP -- TCP extensions for transactions functional
   specification," RFC 1644, Internet Engineering Task Force, July 1994.

   [45] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
   Reading, Massachusetts: Addison-Wesley, 1994.


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