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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
                                                             Columbia U.
                                                                  A. Rao
                                                                   Cisco
                                                             R. Lanphier
                                                            RealNetworks
                                                           M. Westerlund
                                                                Ericsson


draft-ietf-mmusic-rfc2326bis-03.txt
March 3, 2003
Expires: September, 2003


                  Real Time Streaming Protocol (RTSP)

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

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

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference mate-
   rial 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

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.

Abstract

   This memorandum is a revision of RFC 2326, which is currently a Pro-
   posed Standard.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over the delivery of data with real-time proper-
   ties. RTSP provides an extensible framework to enable controlled, on-
   demand delivery of real-time data, such as audio and video. Sources



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   of data can include both live data feeds and stored clips. This pro-
   tocol is intended to control multiple data delivery 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 1889).














































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1 Introduction

1.1 The Update of the RTSP Specification

   This is the draft to an update of the RTSP which currently is a pro-
   posed standard defined in  [21]. During the years since RTSP was pub-
   lished many flaws has been found. This draft tries to address these.
   The work is not yet completed to get all known issues resolved.

   The goal is to progress RTSP to draft standard. If that is possible
   without first publishing it as a proposed standard is not yet deter-
   mined, as it depends on the changes necessary to make the protocol
   work.

   See the list of changes in chapter  F to see what has been addressed.
   The currently open issues are listed in chapter  E.

   There is currently a list of reported bugs available at "http://rtsp-
   spec.sourceforge.net". This list should be taken into account when
   reading this specification. A lot of these bugs are addressed but not
   yet all. Please comment on unresolved ones to give your view.

   Another way of giving input on this work is to send e-mail to the
   MMUSIC WG's mailing list mmusic@ietf.org and the authors.

   Take special notice of the following:

     + The example section  15 has not yet been revised as the changes
       to protocol has not been completed.

     + The BNF chapter  16 has neither been compiled completely.

   All of the contents of RFC 2326 is not longer part of this draft.  In |
   an attempt to prevent the draft from becoming to thick for its own    |
   good, the specification has been reduced and split. The content of    |
   this draft is the core specification of the protocol.  It contains    |
   the basic idea behind RTSP, the basic and general functionality nec-  |
   essary to establish on-demand a play-back session, and the protocol   |
   extension mechanisms. This allow us too keep this draft as short as   |
   possible, it is however still a rather thick document.                |

   Any other functionality will be published as extension documents.  So |
   far there exist two proposals:                                        |

     + NAT and FW traversal mechanisms for RTSP are described in a docu- |
       ment called "How to make Real-Time Streaming Protocol (RTSP) tra- |
       verse Network Address Translators (NAT) and interact with Fire-   |
       walls." [33].                                                     |



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     + The MUTE extension [34] contains a proposal on how to add the     |
       possibility to MUTE and UNMUTE media streams in a aggregated      |
       media session without affecting the time-line of the playback.    |
       Unfortunately the draft has expired in IETF's repository.         |

   There has been discussion about the following extensions to RTSP,     |
   they have however so far not become concrete proposals:               |

     + Transport security for RTSP messages (rtsps).                     |

     + Unreliable transport of RTSP messages (rtspu).                    |

     + The Record functionality.                                         |

     + A text body type with suitable syntax for basic parameters to be  |
       used in SET_PARAMETER, and GET_PARAMETER. Including IANA registry |
       within the defined name space.                                    |

     + An RTSP MIB.                                                      |


1.2 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls
   either a single or several time-synchronized streams of continuous
   media such as audio and video. It does not typically deliver the con-
   tinuous streams itself, although interleaving of the continuous media
   stream with the control stream is possible (see Section 11.11). In
   other words, RTSP acts as a "network remote control" for multimedia
   servers.

   The set of streams to be controlled is defined by a presentation
   description. This memorandum does not define a format for a presenta-
   tion description.

   There is no necessity for a notion of an RTSP connection; instead, a
   server maintains a session labeled by an identifier. An RTSP session
   is in normally not tied to a transport-level connection such as a TCP
   connection. During an RTSP session, an RTSP client may open and close
   many reliable transport connections to the server to issue RTSP
   requests. Alternatively, it may use a connectionless transport proto-
   col such as UDP.

   The streams controlled by RTSP may use RTP [1], but the operation of
   RTSP does not depend on the transport mechanism used to carry contin-
   uous media.





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   The protocol is intentionally similar in syntax and operation to
   HTTP/1.1 [26] so that extension mechanisms to HTTP can in most cases
   also be added to RTSP. However, RTSP differs in a number of important
   aspects from HTTP:

     + RTSP introduces a number of new methods and has a different pro-
       tocol identifier.

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

     + Both an RTSP server and client can issue requests.

     + Data is usually carried out-of-band by a different protocol.
       Session descriptions is one possible exception.

     + RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
       consistent with current HTML internationalization efforts [3].

     + The Request-URI always contains the absolute URI. Because of
       backward compatibility with a historical blunder, HTTP/1.1 [26]
       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 request a pre-
          sentation description via 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
          for security reasons.


     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 distributed     |
          teaching applications. Several parties in the conference may   |
          take turns "pushing the remote control buttons".

     Addition of media to an existing presentation: Particularly for
          live presentations, it is useful if the server can tell the
          client about additional media becoming available.



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   RTSP requests may be handled by proxies, tunnels and caches as in
   HTTP/1.1 [26].

1.3 Requirements

   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 [4].

1.4 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [26]. Terms
   not listed here are defined as in HTTP/1.1.

     Aggregate control: The control of the multiple streams using a sin-
          gle timeline by the server. For audio/video feeds, this means
          that the client may issue a single play or pause message to
          control both the audio and video feeds.

     Aggregate control URI: The URI that represents the whole aggregate.
          Normally specified in the session description.

     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 comprise a presentation when played together. RTSP
          servers may offer aggregate control on these files, though the
          concept of a container file is not embedded in the protocol.

     Continuous media: Data where there is a timing relationship between
          source and sink; that is, the sink must reproduce the timing
          relationship that existed at the source. The most common exam-
          ples of continuous media are audio and motion video. Continu-
          ous media can be real-time (interactive), 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 metainformation in the form of
          entity-header fields and content in the form of an entity-
          body, as described in Section 8.




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

     Media initialization: Datatype/codec specific initialization.  This
          includes such things as clockrates, color tables, etc. Any
          transport-independent information which is required by a
          client for playback of a media stream occurs in the media ini-
          tialization 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 presenta- |
          tion may originate from different media servers. A media       |
          server may reside on the same or a different host as the web   |
          server the presentation is invoked from.

     Media server indirection: Redirection of a media client to a dif-
          ferent 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 applica-
          tion group. When using RTP, a stream consists of all RTP and
          RTCP packets created by a source within an RTP session. This
          is equivalent to the definition of a DSM-CC stream([5]).

     Message: The basic unit of RTSP communication, consisting of a
          structured sequence of octets matching the syntax defined in
          Section 16 and transmitted via a connection or a connection-
          less protocol.

     Non-Aggregated Control: Control of a single media stream.  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, using a presentation description as
          defined below. In most cases in the RTSP context, this implies
          aggregate control of those streams, but does not have to.

     Presentation description: A presentation description contains
          information about one or more media streams within a presenta-
          tion, such as the set of encodings, network addresses and



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          information about the content. Other IETF protocols such as
          SDP (RFC 2327 [24]) use the term "session" for a live presen-
          tation. The presentation description may take several differ-
          ent formats, including but not limited to the session descrip-
          tion 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.


     RTSP session: A state established on a RTSP server by a client with |
          an SETUP request. The RTSP session exist until it either time- |
          outs or is explicitly removed by a TEARDOWN request.  The ses- |
          sion contains state about which media resources that can be    |
          played and their transport.

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

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 trans-
          port level (TLS, RFC 2246 [27]) or within the protocol itself.
          All HTTP authentication mechanisms such as basic (RFC 2616
          [26]) and digest authentication (RFC 2069 [6]) are directly
          applicable.

     Transport-independent: RTSP may use either an unreliable datagram
          protocol (UDP) (RFC 768 [7]), a reliable datagram protocol
          (RDP, RFC 1151, not widely used [8]) or a reliable stream pro-
          tocol such as TCP (RFC 793 [9]) as it implements application-
          level reliability.

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



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     Control of recording devices: The protocol can control both record-
          ing and playback devices, as well as devices that can alter-
          nate between the two modes ("VCR").

     Separation of stream control and conference initiation: Stream con-
          trol is divorced from inviting a media server to a conference.
          In particular, SIP [10] or H.323 [28] may be used to invite a
          server to a conference.

     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 presenta-
          tion description must contain at least one RTSP URI.

     Proxy and firewall friendly: The protocol should be readily handled
          by both application and transport-layer (SOCKS [11]) fire-
          walls. A firewall may need to understand the SETUP method to
          open a "hole" for the UDP 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
          [12,13]) for associating labels with content. However, RTSP
          does not just add methods to HTTP since the controlling con-
          tinuous media requires server state in most cases.

     Appropriate server control: If a client can start a stream, it must
          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.

     Capability negotiation: If basic features are disabled, there must
          be some clean mechanism for the client to determine which
          methods are not going to be implemented. This allows clients
          to present the appropriate user interface. For example, if
          seeking is not allowed, the user interface must be able to
          disallow moving a sliding position indicator.


     An earlier requirement in RTSP was multi-client capability.
     However, it was determined that a better approach was to make



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     sure that the protocol is easily extensible to the multi-
     client scenario. Stream identifiers can be used by several
     control streams, so that "passing the remote" would be possi-
     ble. The protocol would not address how several clients nego-
     tiate access; this is left to either a "social protocol" or
     some other floor control mechanism.

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:

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

     + Some servers may not support setting stream parameters and thus
       not support GET_PARAMETER and SET_PARAMETER.

   A server SHOULD implement all header fields described in Section 13.

   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 [26],
   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 mag-
   nitude of changes supported:

     + Existing methods can be extended with new parameters, as long as
       these parameters can be safely ignored by the recipient. (This is
       equivalent to adding new parameters to an HTML tag.) If the
       client needs negative acknowledgement when a method extension is
       not supported, a tag corresponding to the extension may be added
       in the Require: field (see Section 13.32).

     + New methods can be added. If the recipient of the message does
       not understand the request, it responds with error code 501 (Not
       Implemented) and the sender should not attempt to use this method
       again.  A client may also use the OPTIONS method to inquire about
       methods supported by the server. The server SHOULD list the meth-
       ods it supports using the Public response header.

     + A new version of the protocol can be defined, allowing almost all
       aspects (except the position of the protocol version number) to
       change.





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

1.7 Overall Operation

   Each presentation and media stream may be identified by an RTSP URL.
   The overall presentation and the properties of the media the presen-
   tation is made up 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 main-
   tains 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, lan-
   guage, 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 iden-
   tified by an RTSP URL, 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 exam-
   ple, 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 distin-
   guished:

     Unicast: The media is transmitted to the source of the RTSP
          request, with the port number chosen by the client. Alterna-
          tively, the media is transmitted on the same reliable stream
          as RTSP.

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





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     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 descrip-
          tion, established by means outside the scope of this specifi-
          cation.

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
   occur on a TCP connection while the data flows 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 con-
   trolled by RTSP requests issued sequentially on different TCP connec-
   tions. 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.

   Many methods in RTSP do not contribute to state. However, the follow- |
   ing 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.

     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 13.37) 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 inter-
   act with HTTP in that the initial contact with streaming content is
   often to be made through a web page. The current protocol specifica-
   tion 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
   roundtrips in web-browser-based scenarios, yet also allows for



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   standalone 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 not stateless; they may set parame-
   ters 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.

   While most real-time media will use RTP as a transport protocol, RTSP
   is not tied to RTP.

   RTSP assumes the existence of a presentation description format that
   can express both static and temporal properties of a presentation
   containing several media streams.

2 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 [26]).

   All the mechanisms specified in this document are described in both
   prose and an augmented Backus-Naur form (BNF) similar to that used in
   [H2.1]. It is described in detail in RFC 2234 [14], with the differ-
   ence that this RTSP specification maintains the "#" notation for
   comma-separated lists from [H2.1].

   In this draft, we use indented and smaller-type paragraphs to provide
   background and motivation. This is intended to give readers who were
   not involved with the formulation of the specification an understand-
   ing of why things are the way that they are in RTSP.

   b

3 Protocol Parameters

3.1 RTSP Version





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   HTTP Specification Section [H3.1] applies, with HTTP replaced by
   RTSP. This specification defines version 1.0 of RTSP.

3.2 RTSP URL

   The "rtsp", "rtsps" and "rtspu" schemes are used to refer to network
   resources via the RTSP protocol. This section defines the scheme-spe-
   cific syntax and semantics for RTSP URLs.


   rtsp_URL  =  ( "rtsp:" / "rtspu:" / "rtsps:" )
                "//" host [ ":" port ] [ abs_path ] [ "#" fragment ]
   host      =  As defined by RFC 2732 [30]
   abs_path  =  As defined by RFC 2396 [22]
   port      =  *DIGIT



     Note that fragment and query identifiers do not have a well-
     defined meaning at this time, with the interpretation left to
     the RTSP server.

   The scheme rtsp requires that commands are issued via a reliable pro- |
   tocol (within the Internet, TCP), while the scheme rtspu identifies   |
   an unreliable protocol (within the Internet, UDP). The scheme rtsps   |
   identifies a reliable transport using TLS [27]. The rtspu and rtsps   |
   is not defined in this specification and if for future extensions of  |
   the protocol.

   If the port is empty or not given, port 554 is assumed.  The seman-
   tics are that the identified resource can be controlled by RTSP at
   the server listening for TCP (scheme "rtsp") connections or UDP
   (scheme "rtspu") packets on that port of host, and the Request-URI
   for the resource is rtsp_URL.

   The use of IP addresses in URLs SHOULD be avoided whenever possible
   (see RFC 1924 [16]). Note: Using qualified domain names in any URL is
   one requirement for making it possible for RFC 2326 implementations
   of RTSP to use IPv6. This specification is updated to allow for lit-
   eral IPv6 addresses in RTSP URLs using the host specification in RFC
   2732 [30].

   A presentation or a stream is identified by a textual media identi-
   fier, using the character set and escape conventions [H3.2] of URLs
   (RFC 2396 [22]). URLs 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 individ-
   ual stream within the presentation. Note that some request methods



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   can only be applied to streams, not presentations and vice versa.

   For example, the RTSP URL:

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


identifies 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 URL:

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


identifies the presentation "twister", which may be composed of audio
and video streams.


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

   The path components of the RTSP URL 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 URL.

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 charac- |
   ters long to make guessing it more difficult. (See Section 17.)


   session-id  =  8*( ALPHA / DIGIT / safe )


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



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   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 time". For 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 value is zero, it may be omitted. Subframes are
   measured in one-hundredth of a frame.


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


   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



3.5 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position rela-
   tive to the beginning of the presentation, not to be confused with
   the Network Time Protocol (NTP). 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 content.

   NPT is defined as in DSM-CC: "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),



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   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."  [5]

   npt-range       =  ["npt" "="] npt-range-spec
                      ; implementations SHOULD use npt= prefix, but SHOULD
                      ; be prepared to interoperate with RFC 2326
                      ; implementations which don't use it
   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


   Examples:

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




     The syntax conforms to ISO 8601. The npt-sec notation is opti-
     mized for automatic generation, the ntp-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 timestamps, using UTC (GMT).
   Fractions of a second may be indicated.


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





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

     19961108T143720.25Z



3.7 Feature-tags

   Feature-tags are unique identifiers used to designate new features in
   RTSP. These tags are used in in Require (Section 13.32), Proxy-
   Require (Section 13.27), Unsupported (Section 13.41), and Supported
   (Section 13.38) header fields.

   Syntax:

   feature-tag  =  token


   The creator of a new RTSP feature-tag should either prefix the fea-
   ture-tag with a reverse domain name (e.g., "com.foo.mynewfeature" is
   an apt name for a feature whose inventor can be reached at
   "foo.com"), or register the new feature-tag with the Internet
   Assigned Numbers Authority (IANA), see IANA Section  18.

4 RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [18]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by them-
   selves as line terminators.


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

   The 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 2279 [18])





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   RTSP messages can be carried over any lower-layer transport protocol
   that is 8-bit clean. RTSP messages are vulnerable to bit errors and
   SHOULD NOT be subjected to them.

   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]

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 consists of only CRLF.)

     2.   If a Content-Length header field (section 13.14) 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.

   Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
   transfer coding(see [H3.6.1]) and requires the presence of the Con-
   tent-Length header field.


     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 chun-
     ked transfer encoding unnecessary.





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5 General Header Fields

   See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade,
   and Warning headers are not defined. RTSP further defines the CSeq,
   and Timestamp:


   general-header  =  Cache-Control  ; Section 13.9
                   /  Connection     ; Section 13.10
                   /  CSeq           ; Section 13.17
                   /  Date           ; Section 13.18
                   /  Timestamp      ; Section 13.39
                   /  Via            ; Section 13.44


6 Request

   A request message from a client to a server or vice versa includes,
   within the first line of that message, the method to be applied to
   the resource, the identifier of the resource, and the protocol ver-
   sion in use.


   Request  =   Request-Line      ; Section 6.1
            *(  general-header    ; Section 5
            /   request-header    ; Section 6.2
            /   entity-header )   ; Section 8.1
                CRLF
                [ message-body ]  ; Section 4.3


6.1 Request Line


   Request-Line  =  Method SP Request-URI SP RTSP-Version CRLF



   Method  =  "DESCRIBE"        ; Section 11.2
           /  "GET_PARAMETER"   ; Section 11.7
           /  "OPTIONS"         ; Section 11.1
           /  "PAUSE"           ; Section 11.5
           /  "PLAY"            ; Section 11.4
           /  "PING"            ; Section 11.10
           /  "REDIRECT"        ; Section 11.9
           /  "SETUP"           ; Section 11.3
           /  "SET_PARAMETER"   ; Section 11.8




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           /  "TEARDOWN"        ; Section 11.6
           /  extension-method



   extension-method  =  token
   Request-URI       =  "*" / absolute_URI
   RTSP-Version      =  "RTSP" "/" 1*DIGIT "." 1*DIGIT


6.2 Request Header Fields


   request-header  =  Accept             ; Section 13.1
                   /  Accept-Encoding    ; Section 13.2
                   /  Accept-Language    ; Section 13.3
                   /  Authorization      ; Section 13.6
                   /  Bandwidth          ; Section 13.7
                   /  Blocksize          ; Section 13.8
                   /  From               ; Section 13.20
                   /  If-Modified-Since  ; Section 13.23
                   /  Proxy-Require      ; Section 13.27
                   /  Range              ; Section 13.29
                   /  Referer            ; Section 13.30
                   /  Require            ; Section 13.32
                   /  Scale              ; Section 13.34
                   /  Session            ; Section 13.37
                   /  Speed              ; Section 13.35
                   /  Supported          ; Section 13.38
                   /  Transport          ; Section 13.40
                   /  User-Agent         ; Section 13.42


   Note that in contrast to HTTP/1.1 [26], RTSP requests always contain
   the absolute URL (that is, including the scheme, host and port)
   rather than just the absolute path.


     HTTP/1.1 requires servers to understand the absolute URL, 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.

   The asterisk "*" in the Request-URI means that the request does not
   apply to a particular resource, but to the server or proxy itself,
   and is only allowed when the method used does not necessarily apply
   to a resource.




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   One example would be:


     OPTIONS * RTSP/1.0


Which will determine the capabilities of the server or the proxy that
first receives the request. If one needs to address the server explic-
itly one needs to put in a absolute URL with the servers address.


     OPTIONS rtsp://example.com RTSP/1.0



7 Response

   [H6] applies except that HTTP-Version is replaced by RTSP-Version.
   Also, RTSP defines additional status codes and does not define some
   HTTP codes. The valid response codes and the methods they can be used
   with are defined in Table 1.

   After receiving and interpreting a request message, the recipient
   responds with an RTSP response message.


   Response  =   Status-Line       ; Section 7.1
             *(  general-header    ; Section 5
             /   response-header   ; Section 7.1.2
             /   entity-header )   ; Section 8.1
                 CRLF
                 [ message-body ]  ; Section 4.3


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.


   Status-Line  =  RTSP-Version SP Status-Code SP Reason-Phrase CRLF


7.1.1 Status Code and Reason Phrase




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   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 12. 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
   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:

     + 1xx: Informational - Request received, continuing process

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

     + 3rr: Redirection - Further action must be taken in order to com-
       plete the request

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

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

   The individual values of the numeric status codes defined for
   RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
   presented below. 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 [26] status codes and
   adds RTSP-specific status codes starting at x50 to avoid conflicts
   with newly defined HTTP status codes.



     Status-Code  =  "100"          ; Continue
                  /  "200"          ; OK
                  /  "201"          ; Created
                  /  "250"          ; Low on Storage Space
                  /  "300"          ; Multiple Choices
                  /  "301"          ; Moved Permanently
                  /  "302"          ; Moved Temporarily
                  /  "303"          ; See Other
                  /  "304"          ; Not Modified
                  /  "305"          ; Use Proxy
                  /  "350"          ; Going Away




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                  /  "351"          ; Load Balancing
                  /  "400"          ; Bad Request
                  /  "401"          ; Unauthorized
                  /  "402"          ; Payment Required
                  /  "403"          ; Forbidden
                  /  "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
                  /  "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, excluding CR, LF>


   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



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   x00 status code of that class, with the exception that an unrecog-
   nized 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.

7.1.2 Response Header Fields

   The response-header fields allow the request recipient to pass addi-
   tional information about the response which cannot be placed in the
   Status-Line. These header fields give information about the server
   and about further access to the resource identified by the Request-
   URI.


   response-header  =  Accept-Ranges       ; Section
   13.4
                    /  Location            ; Section 13.25
                    /  Proxy-Authenticate  ; Section 13.26
                    /  Public              ; Section 13.28
                    /  Range               ; Section 13.29
                    /  Retry-After         ; Section 13.31
                    /  RTP-Info            ; Section 13.33
                    /  Scale               ; Section 13.34
                    /  Session             ; Section 13.37
                    /  Server              ; Section 13.36
                    /  Speed               ; Section 13.35
                    /  Transport           ; Section 13.40
                    /  Unsupported         ; Section 13.41
                    /  Vary                ; Section 13.43
                    /  WWW-Authenticate    ; Section 13.45


   Response-header field names can be extended reliably only in combina-
   tion with a change in the protocol version. However, new or experi-
   mental 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 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



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        Code  reason
        --------------------------------------------------------
        100   Continue                          all
        --------------------------------------------------------
        200   OK                                all
        201   Created                           RECORD
        250   Low on Storage Space              RECORD
        --------------------------------------------------------
        300   Multiple Choices                  all
        301   Moved Permanently                 all
        302   Found                             all
        303   See Other                         all
        305   Use Proxy                         all
        350   Going Away                        all
        351   Load Balancing                    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
        --------------------------------------------------------
        500   Internal Server Error             all
        501   Not Implemented                   all
        502   Bad Gateway                       all



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        503   Service Unavailable               all
        504   Gateway Timeout                   all
        505   RTSP Version Not Supported        all
        551   Option not support                all


   Table 1: Status codes and their usage with RTSP methods

   responses will only include the entity-headers.

   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 optional meta-information about the
   entity-body or, if no body is present, about the resource identified
   by the request.


   entity-header     =  Allow             ; Section 13.5
                     /  Content-Base      ; Section 13.11
                     /  Content-Encoding  ; Section 13.12
                     /  Content-Language  ; Section 13.13
                     /  Content-Length    ; Section 13.14
                     /  Content-Location  ; Section 13.15
                     /  Content-Type      ; Section 13.16
                     /  Expires           ; Section 13.19
                     /  Last-Modified     ; Section 13.24
                     /  extension-header
   extension-header  =  message-header


   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 a RTSP message with an entity body
   MUST include a Content-Type header.

9 Connections

   RTSP requests can be transmitted in several different ways:





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     + persistent transport connections used for several request-
       response transactions;

     + one connection per request/response transaction;

     + connectionless mode.

   The type of transport connection is defined by the RTSP URI (Section
   3.2). For the scheme "rtsp", a connection is assumed, while the
   scheme "rtspu" calls for RTSP requests to be sent without setting up
   a connection.

   Unlike HTTP, RTSP allows the media server to send requests to the
   media client. However, this is only supported for persistent connec-
   tions, as the media server otherwise has no reliable way of reaching
   the client.  Also, this is the only way that requests from media
   server to client are likely to traverse firewalls.

9.1 Pipelining

   A client that supports persistent connections or connectionless mode
   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 same order that the requests were received.

9.2 Reliability and Acknowledgements

   Requests are acknowledged by the receiver unless they are sent to a
   multicast group. If there is no acknowledgement, the sender may
   resend the same message after a timeout of one round-trip time (RTT).
   The round-trip time is estimated as in TCP (RFC 1123) [15], with an
   initial round-trip value of 500 ms. An implementation MAY cache the
   last RTT measurement as the initial value for future connections.

   If a reliable transport protocol is used to carry RTSP, requests MUST
   NOT be retransmitted; the RTSP application MUST instead rely on the
   underlying transport to provide reliability.


     If both the underlying reliable transport such as TCP and the
     RTSP application retransmit requests, it is possible that each
     packet loss results in two retransmissions. The receiver can-
     not typically take advantage of the application-layer retrans-
     mission since the transport stack will not deliver the appli-
     cation-layer retransmission before the first attempt has
     reached the receiver. If the packet loss is caused by conges-
     tion, multiple retransmissions at different layers will exac-
     erbate the congestion.



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   If RTSP is used over a small-RTT LAN, standard procedures for opti-
   mizing initial TCP round trip estimates, such as those used in T/TCP
   (RFC 1644) [19], can be beneficial.

   The Timestamp header (Section 13.39) is used to avoid the retransmis-
   sion ambiguity problem [20] and obviates the need for Karn's algo-
   rithm.

   Each request carries a sequence number in the CSeq header (Section
   13.17), which MUST be incremented by one for each distinct request
   transmitted. If a request is repeated because of lack of acknowledge-
   ment, the request MUST carry the original sequence number (i.e., the
   sequence number is not incremented).

   Systems implementing RTSP MUST support carrying RTSP over TCP and MAY
   support UDP. The default port for the RTSP server is 554 for both UDP
   and TCP.

   A number of RTSP packets destined for the same control end point may
   be packed into a single lower-layer PDU or encapsulated into a TCP
   stream. RTSP data MAY be interleaved with RTP and RTCP packets.
   Unlike HTTP, an RTSP message MUST contain a Content-Length header
   field whenever that message contains a payload. Otherwise, an RTSP
   packet is terminated with an empty line immediately following the
   last message header.

9.3 The usage of connections


   TCP can be used for both persistent connections and for one message   |
   exchange per connection, as presented above. This section gives fur-  |
   ther rules and recommendations on how to handle these connections so  |
   maximum interoperability and flexibility can be achieved.             |

   A server SHALL handle both persistent connections and one             |
   request/response transaction per connection. A persistent connection  |
   MAY be used for all transactions between the server and client,       |
   including messages to multiple RTSP sessions. However the persistent  |
   connection MAY also be closed after a few message exchanges, e.g. the |
   initial setup and play command in a session. Later when the client    |
   wishes to send a new request, e.g.  pause, to the session a new con-  |
   nection is opened. This connection may either be for a single message |
   exchange or can be kept open for several messages, i.e. persistent.   |

   A major motivation for allowing non-persistent connections are that   |
   they ensure fault tolerance. A server and client supporting non-per-  |
   sistent connection can survive a loss of a TCP connection, e.g. due   |
   to a NAT timeout. When the it is discovered that the TCP connection   |



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   has been lost one sets up a new one.                                  |

   The client MAY close the connection at any time when no outstanding   |
   request/response transactions exist. The server SHOULD NOT close the  |
   connection unless at least one RTSP session timeout period has passed |
   without data traffic. A server MUST NOT initiate a close of a connec- |
   tion directly after responding to a TEARDOWN request for the whole    |
   session.                                                              |

   The client 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 is RECOMMENDED to use only a single connec-    |
   tion.                                                                 |

   Older services which was implemented according to RFC 2326 sometimes  |
   requires the client to use persistent connection. The client closing  |
   the connection may result in that the server removes the session. To  |
   achieve interoperability with old servers any client is strongly REC- |
   OMMENDED to use persistent connections.                               |

   A Client is also strongly RECOMMENDED to use persistent connections   |
   as it allows the server to send request to the client.  In cases      |
   where no connection exist between the server and the client, this may |
   cause the server to be forced to drop the RTSP session without noti-  |
   fying the client why,due to the lack of signalling channel. An exam-  |
   ple of such a case is when the server desires to send a REDIRECT      |
   request for a RTSP session to the client.                             |

   If a service requires the use of persistent connection an feature-tag |
   is specified for usage in the Require and Proxy-Require headers.      |


   con.persistent                                                           ||


   A server implemented according to this specification MUST respond     |
   that it supports the "play.basic" feature-tag above. A client MAY     |
   send a request including the Supported header in a request to deter-  |
   mine support of non-persistent connections. A server supporting non-  |
   persistent connections will return the "play.basic" feature-tag in    |
   its response. If the client receives the feature-tag in the response, |
   it can be certain that the server handles non-persistent connections.

9.4 Use of IPv6

   This specification has been updated so that it supports IPv6.  How-
   ever this support was not present in RFC 2326 therefore some interop-
   erability issues exist. A RFC 2326 implementation can support IPv6 as



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   long as no explicit IPv6 addresses are used within RTSP messages.
   This require that any RTSP URL pointing at a IPv6 host must use fully
   qualified domain name and not a IPv6 address.  Further the Transport
   header must not use the parameters source and destination.

   Implementations according to this specification MUST understand IPv6
   addresses in URLs, and headers. By this requirement the feature-tag
   "play.basic" can be used to determine that a server or client is
   capable of handling IPv6 within RTSP.

10 Capability Handling

   This chapter describes the capability handling mechanism available in
   RTSP which allows RTSP to be extended. Extensions too 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 these two cases.

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

   It is not necessary to use OPTIONS to discover support of a method,
   it is possible to simple try it. If the receiver of the request does
   not support the method it will respond with an error code indicating
   the the method are 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.

   To handle functionality additions that are not new methods feature-
   tags are defined. Each feature-tag represents a certain block of
   functionality. The amount of functionality that a feature-tag repre-
   sents can vary significant. A simple feature-tag can simple represent
   the functionality a single header gives.  Another feature-tag is
   "play.basic" which represents the minimal playback implementation
   according to the updated specification.

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

     Supported: The supported header are used to determine the complete
          set of functionality that both client and server has. The



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          intended usage is to determine before one needs to use a func-
          tionality that it is supported. If can be used in any method
          however OPTIONS is the most suitable as one at the same time
          determines all methods that are implemented. When sending a
          request the requestor declares all its capabilities by includ-
          ing all supported feature-tags. The results in that the
          receiver learns the requestors feature support. The receiver
          then includes its set of features in the response.

     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 must
          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 understood are 551 (Option Not Supported). Also the fea-
          tures that are not understood 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 551 error response to tell
          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 sit-
          uations as it knows which features that was not supported.

11 Method Definitions

   The method token indicates the method to be performed on the resource
   identified by the Request-URI case-sensitive. New methods may be
   defined in the future. Method names may not start with a $ character
   (decimal 24) and must be a token as defined by the ABNF. Methods are
   summarized in Table 2.


   Notes on Table 2: PAUSE is recommended, but not required in that a
   fully functional server can be built that does not support this
   method, for example, for live feeds. If a server does not support a



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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     recommended    recommended
    PING           C->S, S->C  P,S     recommended    optional
    PLAY           C->S        P,S     required       required
    REDIRECT       S->C        P,S     optional       optional
    SETUP          C->S        S       required       required
    SET_PARAMETER  C->S, S->C  P,S     optional       optional
    TEARDOWN       C->S        P,S     required       required


   Table 2: Overview of RTSP methods, their direction, and what  objects
   (P:  presentation, S: stream) they operate on. Legend: R=Responde to,
   Sd=Send, Opt: Optional, Req: Required, Rec: Recommended

   particular method, it MUST return 501 (Not Implemented) and a client
   SHOULD not try this method again for this server.

11.1 OPTIONS

   The behavior is equivalent to that described in [H9.2]. An OPTIONS
   request may be issued at any time, e.g., if the client is about to
   try a nonstandard request. It does not influence the session state.
   The Public header MUST be included in responses to indicate which
   methods that are supported by the server. To specify which methods
   that are possible to use for the specified resource, the Allow MAY be
   used. By including in the OPTIONS request a Supported header, the
   requester can determine which features the other part supports.

   The request URI determines which scope the OPTIONS request has.  By
   giving the URI of a certain media the capabilities regarding this
   media will be responded. By using the "*" URI the request regards the
   next hop only, while having a URL with only the host address regards
   the server without any media relevance.

   Example:


     C->S:  OPTIONS * RTSP/1.0
            CSeq: 1
            User-Agent: PhonyClient/1.2
            Require:
            Proxy-Require: gzipped-messages
            Supported: play-basic



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     S->C:  RTSP/1.0 200 OK
            CSeq: 1
            Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
            Supported: play-basic, implicit-play, gzipped-messages
            Server: PhonyServer/1.0



   Note that some of the feature-tags in Require and Proxy-Require are
   necessarily fictional features (one would hope that we would not pur-
   posefully overlook a truly useful feature just so that we could have
   a strong example in this section).

11.2 DESCRIBE

   The DESCRIBE method retrieves the description of a presentation or
   media object identified by the request URL from a server. It may use
   the Accept header to specify the description formats that the client
   understands. The server responds with a description of the requested
   resource. The DESCRIBE reply-response pair constitutes the media ini-
   tialization phase of RTSP.

   Example:


     C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
           CSeq: 312
           User-Agent: PhonyClient 1.2
           Accept: application/sdp, application/rtsl, application/mheg

     S->C: RTSP/1.0 200 OK
           CSeq: 312
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Content-Type: application/sdp
           Content-Length: 376

           v=0
           o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
           s=SDP Seminar
           i=A Seminar on the session description protocol
           u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
           e=mjh@isi.edu (Mark Handley)
           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



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           m=application 32416 UDP WB
           a=orient:portrait



   The DESCRIBE response MUST contain all media initialization informa-
   tion for the resource(s) that it describes. If a media client obtains
   a presentation description from a source other than DESCRIBE and that
   description contains a complete set of media initialization parame-
   ters, the client SHOULD use those parameters and not then request a
   description for the same media via RTSP.

   Additionally, servers SHOULD NOT use the DESCRIBE response as a means
   of media indirection.


     By forcing a DESCRIBE response to contain all media initial-
     ization for the set of streams that it describes, and discour-
     aging use of DESCRIBE for media indirection, we avoid looping
     problems that might result from other approaches.

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

     + via RTSP's DESCRIBE method;

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

     + via the command line or standard input (thus working as a browser
       helper application launched with an SDP file or other media ini-
       tialization format).

   It is RECOMMENDED that minimal servers support the DESCRIBE method,
   and highly recommended that minimal clients support the ability to
   act as a "helper application" that accepts a media initialization
   file from standard input, command line, and/or other means that are
   appropriate to the operating environment of the client.

11.3 SETUP

   The SETUP request for a URI specifies the transport mechanism to be
   used for the streamed media. A client can 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
   this, it MUST respond with error 455 (Method Not Valid In This
   State).



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   A server MAY allow a client to do SETUP while in playing state to add
   additional media streams. If not supported the server shall responde
   with error 455 (Method Not Allowed In This State). If supported the
   added media shall then start to play in sync with the already playing
   media. To be able to sync the media with the already playing streams
   the SETUP response MUST include a RTP-Info header with the timestamp
   value, and a Range header with the corresponding normal play time. To
   indicate support for this optional feature the feature-tag:
   "setup.playing" is defined.

   For the benefit of any intervening firewalls, a client must indicate
   the transport parameters even if it has no influence over these
   parameters, for example, where the server advertises a fixed multi-
   cast address.


     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 ini-
     tialization.

   The Transport header specifies the transport parameters acceptable to
   the client for data transmission; the response will contain the
   transport parameters selected by the server.


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Transport: RTP/AVP;unicast;client_port=4588-4589

     S->C: RTSP/1.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Server: PhonyServer 1.0
           Session: 47112344
           Transport: RTP/AVP;unicast;
             client_port=4588-4589;server_port=6256-6257



   The server generates session identifiers in response to SETUP
   requests. If a SETUP request to a server includes a session identi-
   fier, the server MUST bundle this setup request into the existing
   session (aggregated session) or return error 459 (Aggregate Operation
   Not Allowed) (see Section 12.4.11).





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   To control an aggregated session an aggregated control URI MUST be
   used. The aggregated control URI MUST be different from any of the
   media control URIs included in the aggregate. The aggregated URI
   SHOULD be specified by session description, as no general rule exist
   to derive it from the included media's.

   A session will exist until it is torn down by a TEARDOWN request or
   times out. The server MAY remove a session that have had no liveness
   signs from the client in the specified timeout time. The default
   timeout time is 60 seconds, the server MAY set this to another value,
   by in the SETUP response include a timeout value in the session
   header. For further discussion see chapter  13.37. Signs of client
   liveness are:

     + RTCP sender or receiver reports from the client in any of the RTP
       sessions part of the RTSP session.

     + Any RTSP request which includes a Session header with the ses-
       sion's ID.

11.4 PLAY

   The PLAY method tells the server to start sending data via the mecha-
   nism specified in SETUP. A client MUST NOT issue a PLAY request until
   any outstanding SETUP requests have been acknowledged as successful.

   In an aggregated session the PLAY request MUST contain an aggregated
   control URL. 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 must use separate RTSP ses-
   sions for each media.

   The PLAY request positions 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 is reached. To allow for precise composi-
   tion multiple ranges MAY be specified.  The range values are valid if
   all given ranges are part of any media. 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, imme-
   diately following, seconds 20 to 25, and finally seconds 30 through
   the end.


     C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
           CSeq: 835



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           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 starts playing a
   stream from the beginning unless the stream has been paused. If a
   stream has been paused via PAUSE, stream delivery resumes at the
   pause point.

   The Range header may also contain a time parameter. This parameter
   specifies a time in UTC at which the playback should start. If the
   message is received after the specified time, playback is started
   immediately. The time parameter may be used to aid in synchronization
   of streams obtained from different sources. Note: The usage of time
   has two problems. First, at the time requested the RTSP state machine
   may not accept the request. The client will not get any notification
   of the failure. Secondly, the server has difficulties to produce the
   synchronization information for the RTP-Info header ahead of the
   actually play-out. Due to these reasons it is RECOMMENDED that a
   client not issues more than one timed request and no request without
   timing , until it is performed. The server SHALL in responses to
   timed PLAY request give in the RTP-Info header, the sequence number
   of the next RTP packet that will be send for that media, the RTP
   timestamp value corresponding to the activation time of the request.
   Unless the session is in paused state and not plays a single media
   packet the RTP sequence number will be in error. The RTP timestamp
   should be correct unless another timestamp rate has been used in
   between the issuing of the request and activation.

   Server MUST include a "Range" header in any PLAY response. The        |
   response MUST use the same format as the request's range header con-  |
   tained. If no Range header was in the request, the NPT time format    |
   SHOULD be used unless the client showed support for other formats.    |
   For a session with live media streams the Range header MUST also be   |
   given, containing a valid time indication. It is RECOMMENDED that     |
   either "npt=now-" or a absolute time value (clock) for the corre-     |
   sponding time is given, i.e.  "clock=20030213T143205Z-". The UTC      |
   clock format SHOULD only be used if client has shown support for it.

   For a on-demand stream, the server MUST reply with the actual range
   that will be played back. This may differ from the requested range if
   alignment of the requested range to valid frame boundaries is
   required for the media source. If no range is specified in the
   request, the start position SHALL still be returned in the reply. The



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   unit of the range in the reply is the same as that in the request. If
   the medias part of an aggregate has different lengths the PLAY
   request and any Range SHALL be performed as long it is valid for the
   longest media.  Media will be sent whenever it is available for the
   given play-out point.

   After playing the desired range, the presentation is NOT automati-
   cally paused, media deliver simply stops. A PAUSE request MUST be
   issued before another PLAY request can issued. Note: This is one
   change resulting in a non-operability with RFC 2326 implementations.
   A client not issuing a PAUSE request before a new PLAY will be stuck
   in PLAYING state. A client desiring to play the media from the begin-
   ning MUST send a PLAY request with a Range header pointing at the
   beginning, e.g. npt=0-.

   The following example plays the whole presentation starting at SMPTE
   time code 0:10:20 until the end of the clip. The playback is to start
   at 15:36 on 23 Jan 1997. 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/1.0
         CSeq: 833
         Session: 12345678
         Range: smpte=0:10:20-;time=19970123T153600Z

   S->C: RTSP/1.0 200 OK
         CSeq: 833
         Date: 23 Jan 1997 15:35:06 GMT
         Server: PhonyServer 1.0
         Range: smpte=0:10:22-;time=19970123T153600Z
         RTP-Info:url=rtsp://example.com/twister.en;
            seq=14783;rtptime=2345962545



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


     C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: clock=19961108T142300Z-19961108T143520Z

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:06 GMT



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           Server:PhonyServer 1.0
           Range: clock=19961108T142300Z-19961108T143520Z
           RTP-Info:url=rtsp://example.com/meeting.en;
              seq=53745;rtptime=484589019




   A media server only supporting playback MUST support the npt format
   and MAY support the clock and smpte formats.

   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 queued play functionality described in RFC 2326 [21] is removed
   and multiple ranges can be used to achieve a similar performance. If
   a server receives a PLAY request while in the PLAY state, the server
   SHALL responde using the error code 455 (Method Not Valid In This
   State). This will signal the client that queued play are not sup-
   ported.

   The use of PLAY for keep-alive signaling, i.e. PLAY request without a
   range header, has also been decapitated.  Instead a client can use,
   PING, SET_PARAMETER or OPTIONS for keep alive. A server receiving a
   PLAY keep alive SHALL respond with the 455 error code.

   When playing live media, indicated by the Accept-Ranges header the    |
   session are in a live state. This live state will put some restric-   |
   tions on the action available for a client. A PLAY request without a  |
   Range header will start media deliver at the current point in the     |
   live presentation, i.e. now. Any seeking in the media will be impos-  |
   sible. The only allowed usage of the Range header is npt=now-, and    |
   certain clock units.  The usage of npt=now- is unnecessary as it has  |
   the exact same meaning as a request without Range header. The clock   |
   format can be used to specify start and stop times for media delivery |
   in a live session.

11.5 PAUSE

   The PAUSE request causes the stream delivery to be interrupted        |
   (halted) temporarily. A PAUSE request MUST be done with the aggre-    |
   gated 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    |



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   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/1.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76



   The PAUSE request may contain a Range header specifying when the
   stream or presentation is to be halted. We refer to this point as the
   "pause point". The header MUST contain a single value, expressed as
   the beginning value an open range. For example, the following clip
   will be played from 10 seconds through 21 seconds of the clip's nor-
   mal play time, under the assumption that the PAUSE request reaches
   the server within 11 seconds of the PLAY request. Note that some
   lines has been broken in an non-correct way to fit the page:


     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678
           Range: npt=10-30

     S->C: RTSP/1.0 200 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;
                   seq=5712;rtptime=934207921,
                   url=rtsp://example.com/fizzle/videotrack;
                   seq=57654;rtptime=2792482193
           Session: 12345678

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0



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           CSeq: 835
           Session: 12345678
           Range: npt=21-

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



   The pause request becomes effective the first time the server is      |
   encountering the time point specified in any of the multiple ranges.  |
   If the Range header specifies a time outside any range from the PLAY  |
   request, the error 457 (Invalid Range) SHALL be returned. If a media  |
   unit (such as an audio or video frame) starts presentation at exactly |
   the pause point, it is not played. If the Range header is missing,    |
   stream delivery is interrupted immediately on receipt of the message  |
   and the pause point is set to the current normal play time. However,  |
   the pause point in the media stream MUST be maintained. A subsequent  |
   PLAY request without Range header resumes from the pause point and    |
   play until media end.

   The actual 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 simple included the Range header from the PAUSE
   response.

   For example, if the server have a play request for ranges 10 to 15
   and 20 to 29 pending and then receives a pause request for NPT 21, it
   would start playing the second range and stop at NPT 21. If the pause
   request is for NPT 12 and the server is playing at NPT 13 serving the
   first play request, the server stops immediately. If the pause
   request is for NPT 16, the server returns a 457 error message. To
   prevent that the second range is played and the server stops after
   completing the first range, a PAUSE request for 20 must be issued.

   As another example, if a server has received requests to play ranges
   10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
   request for NPT=14 would take effect while the server plays the first
   range, with the second range effectively being ignored, assuming the
   PAUSE request arrives before the server has started playing the sec-
   ond, overlapping range. Regardless of when the PAUSE request arrives,
   it sets the pause point to 14.



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   If the server has already sent data beyond the time specified in the
   the PAUSE request Range header, a PLAY without range would still
   resume at that point in time, specified by the pause's range header,
   as it is assumed that the client has discarded data after that point.
   This ensures continuous pause/play cycling without gaps.

11.6 TEARDOWN

   The TEARDOWN request stops the stream delivery for the given URI,
   freeing the resources associated with it. If the URI is the aggre-
   gated control URI for this presentation, any RTSP session identifier
   associated with the session is no longer valid. The use of "*" as URI
   in TEARDOWN will also result in that the session is removed indepen-
   dent of the number of medias that was part of it. If the URI in the
   request was for a media within an aggregated session that media is
   removed from the aggregate. However the session and any other media
   stream yet not torn down remains, and any valid request, e.g. PLAY or
   SETUP, can be issued. As an optional feature a server MAY keep the
   session in case the last remaining media is torn down with a TEARDOWN
   request with an URI equal to the media URI. To Indicate what has been
   performed, a server that after any TEARDOWN request, still has a
   valid session MUST in the response return a session header.

   A server MAY choose to allow TEARDOWN of individual media while in
   PLAY state. When this is not allowed the response SHALL be 455
   (Method Not Valid In This State). If a server implements TEARDOWN and
   SETUP in PLAY state it MUST signal this using the "setup.playing"
   feature-tag.

   Example:


     C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 892
           Session: 12345678

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



11.7 GET_PARAMETER

   The GET_PARAMETER request retrieves the value of a parameter of 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 sessions context. The content of the reply and response is left



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   to the implementation.  GET_PARAMETER with no entity body may be used
   to test client or server liveness ("ping").

   Example:


     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 431
           Content-Type: text/parameters
           Session: 12345678
           Content-Length: 15

           packets_received
           jitter

     C->S: RTSP/1.0 200 OK
           CSeq: 431
           Content-Length: 46
           Content-Type: text/parameters

           packets_received: 10
           jitter: 0.3838




     The "text/parameters" section is only an example type for
     parameter. This method is intentionally loosely defined with
     the intention that the reply content and response content will
     be defined after further experimentation.

11.8 SET_PARAMETER

   This method requests to set the value of a parameter for a presenta-
   tion or stream specified by the URI.

   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 can locate a parameter error 451
   (Parameter Not Understood) SHALL be used.  In the case a parameter is
   not allowed to change the error code 458 (Parameter Is Read-Only).
   The response body SHOULD contain only the parameters that has errors.
   Otherwise no body SHALL be returned.




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   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/1.0
           CSeq: 421
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff

     S->C: RTSP/1.0 451 Parameter Not Understood
           CSeq: 421
           Content-length: 10
           Content-type: text/parameters

           barparam




     The "text/parameters" section is only an example type for
     parameter. This method is intentionally loosely defined with
     the intention that the reply content and response content will
     be defined after further experimentation.

11.9 REDIRECT

   A redirect request informs the client that it MUST connect to another |
   server location. The REDIRECT request MAY contain the header Loca-    |
   tion, which indicates that the client should issue requests for that  |
   URL. If the Location URL only contains a host address the client      |
   shall connect to the given host, while using the path from the URL on |
   the current server.                                                   |




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   If a REDIRECT request contains a Session header, it is end-to-end and |
   applies only to the given session. If there are proxies in the        |
   request chain, they SHOULD NOT disconnect the control channel unless  |
   there are no remaining sessions.                                      |

   If a REDIRECT request does not contain a Session header, it is next-  |
   hop and applies to the control connection. The Location header SHOULD |
   only contain a host address. If there are proxies in the request      |
   chain, they SHOULD do all of the following: (1) respond to the REDI-  |
   RECT request, (2) disconnect the control channel from the requestor,  |
   (3) reconnect to the given host address, and (4) pass the request to  |
   each applicable client (typically those clients with an active ses-   |
   sion or unanswered request from the requestor). Note that the proxy   |
   is responsible for accepting the REDIRECT response from its clients   |
   and these responses MUST NOT be passed on to either the requesting or |
   the destination server.

   The redirect request MAY contain the header Range, which indicates
   when the redirection takes effect. If the Range contains a "time="
   value that is the wall clock time that the redirection MUST at the
   latest take place. When the "time=" parameter is present the range
   value MUST be ignored. However the range entered MUST be syntactical
   correct and SHALL point at the beginning of any on-demand content. If
   no time parameter is part of the Range header then redirection SHALL
   take place when the media playout from the server reaches the given
   time. The range value MUST be a single value in the open ended form,
   e.g. npt=59-.

   If the client wants to continue to send or receive media for this
   resource, the client MUST issue a TEARDOWN request for the current
   session. A new session must be established with the designated host.
   A client SHOULD issue a new DESCRIBE request with the URL given in
   the Location header, unless the URL only contains a host address. In
   the cases the Location only contains a host address the client MAY
   assume that the media on the server it is redirected to is identical.
   Identical media means that all media configuration information from
   the old session still is valid except for the host address. In the
   case of absolute URLs in the location header the media redirected to
   can be either identical, slightly different or totally different.
   This is the reason why a new DESCRIBE request SHOULD be issued.

   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/1.0
           CSeq: 732
           Location: rtsp://bigserver.com:8001



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           Range: clock=19960213T143205Z-
           Session: uZ3ci0K+Ld-M



11.10 PING

   This method is a bi-directional mechanism for server or client live-
   ness checking. It has no side effects. The issuer of the request MUST
   include a session header with the session ID of the session that is
   being checked for liveness.

   Prior to using this method, an OPTIONS method is RECOMMENDED to be
   issued in the direction which the PING method would be used. This
   method MUST NOT be used if support is not indicated by the Public
   header. Note: That an 501 (Not Implemented) response means that the
   keep-alive timer has not been updated.

   When a proxy is in use, PING with a * indicates a single-hop liveness
   check, whereas PING with a URL including an host address indicates an
   end-to-end liveness check.

   Example:

     C->S: PING * RTSP/1.0
           CSeq: 123
           Session:12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 123
           Session:12345678



11.11 Embedded (Interleaved) Binary Data

   Certain firewall designs and other circumstances may force a server
   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, fol-
   lowed by the length of the encapsulated binary data as a binary, two-
   byte integer in network byte order. The stream data follows immedi-
   ately afterwards, without a CRLF, but including the upper-layer



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   protocol headers. Each $ block contains exactly one upper-layer pro-
   tocol 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 13.40).

   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 13.40. 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 direc-
   tions.


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


     C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
           CSeq: 2
           Transport: RTP/AVP/TCP;unicast;interleaved=0-1

     S->C: RTSP/1.0 200 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://foo.com/bar.file RTSP/1.0



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           CSeq: 3
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 3
           Session: 12345678
           Date: 05 Jun 1997 18:59:15 GMT
           RTP-Info: url=rtsp://foo.com/bar.file;
             seq=232433;rtptime=972948234

     S->C: $000{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $000{2 byte length}{"length" bytes data, w/RTP header}
     S->C: $001{2 byte length}{"length" bytes  RTCP packet}



12 Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused. Status codes
   that have the same meaning are not repeated here. See Table 1 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.

12.1 Success 1xx

12.1.1 100 Continue

   See, [H10.1.1].

12.2 Success 2xx

12.2.1 250 Low on Storage Space

   The server returns this warning after receiving a RECORD request that
   it may not be able to fulfill completely due to insufficient storage
   space. If possible, the server should use the Range header to indi-
   cate what time period it may still be able to record. Since other
   processes on the server may be consuming storage space simultane-
   ously, a client should take this only as an estimate.

12.3 Redirection 3xx

   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.





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   See [H10.3] for definition of status code 300 to 305. However com-
   ments are given for some to how they apply to RTSP. Further a couple
   of new status codes are defined.

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

12.3.1 300 Multiple Choices

12.3.2 301 Moved Permanently

   The request resource are moved permanently and resides now at the URI
   given by the location header. The user client SHOULD redirect auto-
   matically to the given URI.

12.3.3 302 Found

   The requested resource reside temporarily at the URI given by the
   Location header. The Location header MUST be included. Is intended to
   be used for many types of temporary redirects, e.g.  load balancing.
   It is RECOMMENDED that one set the reason phrase to something more
   meaningful than "Found" in these cases.

12.3.4 303 See Other

   This status code SHALL NOT be used in RTSP. However as it was allowed
   to use in RFC 2326 it is possible that such response will be
   received.

12.3.5 304 Not Modified

   If the client has performed a conditional DESCRIBE or SETUP (see
   12.23) 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:

     + Date

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

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



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   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 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.

12.3.6 305 Use Proxy

   See [H10.3.6].

12.4 Client Error 4xx

12.4.1 400 Bad Request

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

12.4.2 405 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.

12.4.3 451 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).

12.4.4 452 reserved

   This error code was removed from RFC 2326 [21] and is obsolete.

12.4.5 453 Not Enough Bandwidth

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

12.4.6 454 Session Not Found





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

12.4.7 455 Method Not Valid in This State

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

12.4.8 456 Header Field Not Valid for Resource

   The server could not act on a required request header. For example,
   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 SHOULD be returned to inform the client of which
   format(s) that are allowed.

12.4.9 457 Invalid Range

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

12.4.10 458 Parameter Is Read-Only

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

12.4.11 459 Aggregate Operation Not Allowed

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

12.4.12 460 Only Aggregate Operation Allowed

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

12.4.13 461 Unsupported Transport

   The Transport field did not contain a supported transport specifica-
   tion.

12.4.14 462 Destination Unreachable




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   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 Destination param-
   eter in the Transport field.

12.5 Server Error 5xx

12.5.1 551 Option not supported

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

13 Header Field Definitions


             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
             PING          C->S, S->C P,S    PNG
             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 3: 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 general syntax for header fields is covered in Section 4.2 This
   section lists the full set of header fields along with notes on syn-
   tax, meaning, and usage.  Throughout this section, we use [HX.Y] to
   refer to Section X.Y of the current HTTP/1.1 specification RFC 2616
   [26].  Examples of each header field are given.

   Information about header fields in relation to methods and proxy pro-
   cessing is summarized in Table 4 and Table 5.





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   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 all requests and responses.

   The "proxy" column describes the operations a proxy may perform on a
   header field:

     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.

     r: A proxy must 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 3:

     c: Conditional; requirements on the header field depend on the con-
          text 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 required if the message body is not empty.
          See sections 13.14, 13.16 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, and a Client/Server MAY ignore the header
   field if present in the request or response (The exception to this



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   rule is the Require header field discussed in 13.32). 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 appli-
   cable" 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.

   A Client/Server SHOULD ignore extension header parameters that are
   not understood.

   The From, Location, and RTP-Info header fields contain a URI. If the
   URI contains a comma, or semicolon, the URI MUST be enclosed in dou-
   ble 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.



13.1 Accept

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

     The "level" parameter for presentation descriptions is prop-
     erly defined as part of the MIME type registration, not here.

   See [H14.1] for syntax.

   Example of use:

     Accept: application/rtsl q=1.0, application/sdp;level=2



13.2 Accept-Encoding

   See [H14.3]

13.3 Accept-Language

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



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   Header              Where  Proxy DES OPT SETUP PLAY PAUSE TRD
   --------------------------------------------------------------
   Accept                R           o   -    -    -     -   -
   Accept-Encoding       R      r    o   -    -    -     -   -
   Accept-Language       R      r    o   -    -    -     -   -
   Accept-Ranges         r      r    -   -    o    -     -   -
   Accept-Ranges        456     r    -   -    -    o     o   -
   Allow                 r           -   o    -    -     -   -
   Allow                405          -   -    -    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    -     -   -
   Connection                        o   o    o    o     o   o
   Content-Base          r           o   -    -    -     -   -
   Content-Base         4xx          o   o    o    o     o   o
   Content-Encoding      R      r    -   -    -    -     -   -
   Content-Encoding      r      r    o   -    -    -     -   -
   Content-Encoding     4xx     r    o   o    o    o     o   o
   Content-Language      R      r    -   -    -    -     -   -
   Content-Language      r      r    o   -    -    -     -   -
   Content-Language     4xx     r    o   o    o    o     o   o
   Content-Length        r      r    *   -    -    -     -   -
   Content-Length       4xx     r    *   *    *    *     *   *
   Content-Location      r           o   -    -    -     -   -
   Content-Location     4xx          o   o    o    o     o   o
   Content-Type          r           *   -    -    -     -   -
   Content-Type         4xx          *   *    *    *     *   *
   CSeq                 Rc           m   m    m    m     m   m
   Date                        am    o   o    o    o     o   o
   Expires               r      r    o   -    -    -     -   -
   From                  R      r    o   o    o    o     o   o
   Host                              o   o    o    o     o   o
   If-Match              R      r    -   -    o    -     -   -
   If-Modified-Since     R      r    o   -    o    -     -   -
   Last-Modified         r      r    o   -    -    -     -   -
   Location             3rr          o   o    o    o     o   o
   Proxy-Authenticate   407    amr   m   m    m    m     m   m
   Proxy-Require         R     ar    o   o    o    o     o   o
   Public                r    admr   -  m*    -    -     -   -
   Public               501   admr  m*  m*   m*    m*   m*   m*
   Range                 R           -   -    -    o     o   -
   Range                 r           -   -    c    m*    -   -
   Referer               R           o   o    o    o     o   o
   Require               R           o   o    o    o     o   o



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   Retry-After        3rr,503        o   o    o    -     -   -
   RTP-Info              r           -   -    o    m     -   -
   Scale                             -   -    -    o     -   -
   Session               R           -   o    o    m     m   m
   Session               r           -   c    m    m     m   o
   Server                R           -   o    -    -     -   -
   Server                r           o   o    o    o     o   o
   Speed                             -   -    -    o     -   -
   Supported             R           o   o    o    o     o   o
   Supported             r           c   c    c    c     c   c
   Timestamp             R           o   o    o    o     o   o
   Timestamp             c           m   m    m    m     m   m
   Transport                         -   -    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  4: Overview of RTSP header fields related to methods DESCRIBE,
   OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.


13.4 Accept-Ranges

   The Accept-Ranges response-header field allows the server to indicate
   its acceptance of range requests and possible formats for a resource: |

   Accept-Ranges      =  "Accept-Ranges" ":" acceptable-ranges              ||
   acceptable-ranges  =  1#range-unit / "none"                              ||
   range-unit         =  NPT / SMPTE / UTC / LIVE / extension-format        ||
   extension-format   =  token                                              ||



   This header has the same syntax as [H14.5]. However new range-units
   are defined and byte-ranges SHALL NOT be used.  Inclusion of any of
   the three time formats indicates acceptance by the server for PLAY
   and PAUSE requests with this format.  Inclusion of the "LIVE" tag
   indicates that the resource has LIVE properties. The headers value is
   valid for the resource specified by the URI in the request, this
   response corresponds to.




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   Header              Where  Proxy GPR SPR RDR PNG
   -----------------------------------------------------
   Allow                405          -   -   -   -
   Authorization         R           o   o   o   o
   Bandwidth             R           -   o   -   -
   Blocksize             R           -   o   -   -
   Connection                        o   o   o   -
   Content-Base          R           o   o   -   -
   Content-Base          r           o   o   -   -
   Content-Base         4xx          o   o   o   -
   Content-Encoding      R      r    o   o   -   -
   Content-Encoding      r      r    o   o   -   -
   Content-Encoding     4xx     r    o   o   o   -
   Content-Language      R      r    o   o   -   -
   Content-Language      r      r    o   o   -   -
   Content-Language     4xx     r    o   o   o   -
   Content-Length        R      r    *   *   -   -
   Content-Length        r      r    *   *   -   -
   Content-Length       4xx     r    *   *   *   -
   Content-Location      R           o   o   -   -
   Content-Location      r           o   o   -   -
   Content-Location     4xx          o   o   o   -
   Content-Type          R           *   *   -   -
   Content-Type          r           *   *   -   -
   Content-Type         4xx          *   *   *   -
   CSeq                 Rc           m   m   m   m
   Date                        am    o   o   o   o
   From                  R      r    o   o   o   o
   Host                              o   o   o   o
   Last-Modified         R      r    -   -   -   -
   Last-Modified         r      r    o   -   -   -
   Location             3rr          o   o   o   o
   Location              R           -   -   m   -
   Proxy-Authenticate   407    amr   m   m   m   m
   Proxy-Require         R     ar    o   o   o   o
   Public               501   admr  m*  m*  m*  m*
   Range                 R           -   -   o   -
   Referer               R           o   o   o   -
   Require               R           o   o   o   o
   Retry-After        3rr,503        o   o   -   -
   Scale                             -   -   -   -
   Session               R           o   o   o   m
   Session               r           c   c   o   m
   Server                R           o   o   o   o
   Server                r           o   o   -   o



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   Supported             R           o   o   o   o
   Supported             r           c   c   c   c
   Timestamp             R           o   o   o   o
   Timestamp             c           m   m   m   m
   Unsupported           r           c   c   c   c
   User-Agent            R          m*  m*   -  m*
   User-Agent            r           -   -  m*   -
   Vary                  r           c   c   -   -
   Via                   R     amr   o   o   o   o
   Via                   c     dr    m   m   m   m
   WWW-Authenticate     401          m   m   m   m
   -----------------------------------------------------
   Header              Where  Proxy GPR SPR RDR PNG



   Table 5: Overview of RTSP header fields related to methods GET_PARAM-
   ETER, SET_PARAMETER,REDIRECT, and PING.


   A server is RECOMMENDED to use this header in SETUP responses to
   indicate to the client which range time formats the media supports.
   The header SHOULD also be included in "456" responses which is a
   result of use of unsupported range formats.

13.5 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.

   Example of use:

     Allow: SETUP, PLAY, SET_PARAMETER



13.6 Authorization

   See [H14.8]

13.7 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



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   during an RTSP session, e.g., due to modem retraining.


   Bandwidth  =  "Bandwidth" ":" 1*DIGIT


   Example:

     Bandwidth: 4000



13.8 Blocksize

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

   (400) if the value is syntactically invalid.


   Blocksize  =  "Blocksize" ":" 1*DIGIT


13.9 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 appli-
   cation, 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-direc-
   tive 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, but rather of the stream identified by the
   SETUP request. Responses to RTSP requests are not cacheable, except
   for responses to DESCRIBE.




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   Cache-Control             =  "Cache-Control" ":" 1#cache-directive
   cache-directive           =  cache-request-directive
                            /   cache-response-directive
   cache-request-directive   =  "no-cache"
                            /   "max-stale" ["=" delta-seconds]
                            /   "min-fresh" "=" delta-seconds
                            /   "only-if-cached"
                            /   cache-extension
   cache-response-directive  =  "public"
                            /   "private"
                            /   "no-cache"
                            /   "no-transform"
                            /   "must-revalidate"
                            /   "proxy-revalidate"
                            /   "max-age" "=" delta-seconds
                            /   cache-extension
   cache-extension           =  token [ "=" ( token / quoted-string ) ]
   delta-seconds             =  1*DIGIT


     no-cache: Indicates that the media stream MUST NOT be cached any-
          where. 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 opera-
          tional problems may occur, however, when these transformations
          have been applied to streams intended for certain kinds of
          applications. For example, applications for medical imaging,
          scientific data analysis and those using end-to-end authenti-
          cation all depend on receiving a stream that is bit-for-bit
          identical to the original entity-body. 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.





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     only-if-cached: In some cases, such as times of extremely poor net-
          work 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 con-
          straints of the request, or respond with a 504 (Gateway Time-
          out) 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 must 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 mean-
          ing as the must-revalidate directive, except that it does not
          apply to non-shared user agent caches. It can be 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



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          client has supplied its own validator in the request, the sup-
          plied 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 vali-
          dated 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.

13.10 Connection

   See [H14.10]. The use of the connection option "close" in RTSP mes-
   sages 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 shall have the choice of continue using
   a connection indefinitely as long as it sends valid messages.

13.11 Content-Base

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


   Content-Base  =  "Content-Base" ":" absoluteURI


   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.

13.12 Content-Encoding

   See [H14.11]

13.13 Content-Language




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   See [H14.12]

13.14 Content-Length

   The Content-Length general-header field contains the length of the
   content of the method (i.e. after the double CRLF following the last
   header). Unlike HTTP, it MUST be included in all messages that carry
   content 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].

13.15 Content-Location

   See [H14.14]

13.16 Content-Type

   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.

13.17 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 num-
   ber. 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. 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.


   CSeq  =  "Cseq" ":" 1*DIGIT


13.18 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.





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13.19 Expires

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

     DESCRIBE response: The Expires header indicates a date and time
          after which the description 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 14 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:


   Expires  =  "Expires" ":" HTTP-date


   An example of its use is


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



   RTSP/1.0 clients and caches MUST treat other invalid date formats,
   especially including the value "0", as having occurred in the past
   (i.e., already expired).

   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/1.0 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 13.9).



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13.20 From

   See [H14.22].

13.21 Host

   The Host HTTP request header field [H14.23] is not needed for RTSP.
   It should be silently ignored if sent.

13.22 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.

   The identifier is an opaque identifier, and thus is not specific to
   any particular session description language.

13.23 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 will be returned without any message-body.


   If-Modified-Since  =  "If-Modified-Since" ":" HTTP-date


   An example of the field is:


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



13.24 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



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   DESCRIBE, the header field indicates the last modification date and
   time of the description, for SETUP that of the media stream.

13.25 Location

   See [H14.30].

13.26 Proxy-Authenticate

   See [H14.33].

13.27 Proxy-Require

   The Proxy-Require request-header field is used to indicate proxy-sen-
   sitive 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 Unsup-
   ported header. Servers should treat this field identically to the
   Require field, i.e. the Proxy-Require requirements does also apply to
   the server.

   See Section 13.32 for more details on the mechanics of this message
   and a usage example.



        Proxy-Require  =  "Proxy-Require" ":" 1#feature-tag

   Example of use:

      Proxy-Require: play.basic, con.persistent



13.28 Public

   The Public response-header field lists the set of methods supported
   by the server. The purpose of this field is strictly to inform the
   recipient of the capabilities of the server regarding unusual meth-
   ods. 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.


        Public  =  "Public" ":" 1#method

   Example of use:




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      Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN



   This header field applies only to the server directly connected to
   the client (i.e., the nearest neighbor in a chain of connections).
   If the response passes through a proxy, the proxy MUST either remove
   the Public header field or replace it with one applicable to its own
   capabilities.

13.29 Range

   The Range request and response header field specifies a range of
   time. The range can be specified in a number of units.  This specifi-
   cation defines the smpte (Section 3.4), npt (Section 3.5), and clock
   (Section 3.6) range units. Within RTSP, byte ranges [H14.35.1] are
   not meaningful and MUST NOT be used. The header may also contain a
   time parameter in UTC, specifying the time at which the operation is
   to be made effective. Servers supporting the Range header MUST under-
   stand the NPT range format and SHOULD understand the SMPTE range for-
   mat. The Range response header indicates what range of time is actu-
   ally being played. If the Range header is given in a time format that
   is not understood, the recipient should return 501 (Not Implemented).

   Ranges are half-open intervals, including the lower point, but
   excluding the upper 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. As an 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.


   Range             =  "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ]
   ranges-specifier  =  npt-range / utc-range / smpte-range


   Example:

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




     The notation is similar to that used for the HTTP/1.1 [26]
     byte-range header. It allows clients to select an excerpt from



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     the media object, and to play from a given point to the end as
     well as from the current location to a given point. The start
     of playback can be scheduled for any time in the future,
     although a server may refuse to keep server resources for
     extended idle periods.

13.30 Referer

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

13.31 Retry-After

   See [H14.37].

13.32 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  13.38) 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 Sup-
   ported). This header does not apply to proxies, for the same func-
   tionality in respect to proxies see, header Proxy-Require (Section
   13.27).


     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.


   Require  =  "Require" ":" feature-tag *("," feature-tag)


   Example:

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Require: funky-feature



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           Funky-Parameter: funkystuff

   S->C:   RTSP/1.0 551 Option not supported
           CSeq: 302
           Unsupported: funky-feature

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 303

   S->C:   RTSP/1.0 200 OK
           CSeq: 303



   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 trans-
   mitted with every exchange.

   Proxies and other intermediary devices SHOULD ignore features that
   are not understood in this field. If a particular extension requires
   that intermediate devices support it, the extension should be tagged
   in the Proxy-Require field instead (see Section 13.27).

13.33 RTP-Info

   The RTP-Info response-header field is used to set RTP-specific param-
   eters in the PLAY response. For streams using RTP as transport proto-
   col the RTP-Info header SHALL be part of a 200 response to PLAY.

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

     seq: Indicates the sequence number of the first packet of the
          stream. 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.

     rtptime: Indicates the RTP timestamp corresponding to the time
          value in the Range response header. (Note: For aggregate con-
          trol, a particular stream may not actually generate a packet
          for the Range time value returned or implied. Thus, there is
          no guarantee that the packet with the sequence number



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          indicated by seq actually has the timestamp indicated by rtp-
          time.) The client uses this value to calculate the mapping of
          RTP time to NPT.


          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 times-
          tamps to NPT. Furthermore, in order to ensure that this
          information is available at the necessary time (immedi-
          ately at startup or after a seek), and that it is deliv-
          ered reliably, this mapping is placed in the RTSP control
          channel.

          In order to compensate for drift for long, uninterrupted pre-
          sentations, 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.

   Syntax:

   RTP-Info        =  "RTP-Info" ":" 1#rtsp-info-spec
   rtsp-info-spec  =  stream-url 1*parameter
   stream-url      =  quoted-url / unquoted-url
   unquoted-url    =  "url" "=" safe-url
                  /   ";" "mode" = <"> 1#Method <">
   quoted-url      =  "url" "=" <"> needquote-url <">
   safe-url        =  url
   needquote-url   =  url //That contains ; or ,
   url             =  ( absoluteURI / relativeURI )
   parameter       =  ";" "seq" "=" 1*DIGIT
                  /   ";" "rtptime" "=" 1*DIGIT


   Additional constraint: safe-url MUST NOT contain the semicolon (";")
   or comma (",") characters. The quoted-url form SHOULD only be used
   when a URL does not meet the safe-url constraint, in order to ensure
   compatibility with implementations conformant to RFC 2326 [21].

   absoluteURI and relativeURI are defined in RFC 2396 [22] with RFC
   2732 [30] applied.

   Example:

   RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
             url=rtsp://foo.com/bar.avi/streamid=1;seq=30211





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13.34 Scale

   A scale value of 1 indicates normal play at the normal forward view-
   ing 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.

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


   Scale  =  "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]


   Example of playing in reverse at 3.5 times normal rate:


     Scale: -3.5



13.35 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.  Implementa-
   tion 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



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   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" feature-tag.


   Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]


   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 presen-
   tation at a higher or lower rate is necessary. Implementors should
   keep in mind that bandwidth for the session may be negotiated before-
   hand (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 public best-effort networks the
   sender is responsible for performing congestion control of the
   stream. This MAY result in that the communicated speed is impossible
   to maintain.

13.36 Server

   See [H14.38], however the header syntax is here corrected.


   Server  =  "Server" ":" ( product / comment ) *(SP (product / comment))


13.37 Session

   The Session request-header and response-header field identifies an
   RTSP session started by the media server in a SETUP response and con-
   cluded by TEARDOWN on the presentation URL. The session identifier is
   chosen by the media server (see Section 3.3) and MUST be returned in
   the SETUP response. Once a client receives a Session identifier, it
   MUST return it for any request related to that session.


   Session  =  "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]




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   The timeout parameter is only allowed in a response header.  The
   server uses it to indicate to the client how long the server is pre-
   pared to wait between RTSP commands or other signs of life before
   closing the session due to lack of activity (see Section A).  The
   timeout is measured in seconds, with a default of 60 seconds (1
   minute).

   The mechanisms for showing liveness of the client is, any RTSP mes-
   sage with a Session header, or a RTCP message. It is RECOMMENDED that
   a client does not wait to the last second of the timeout before try-
   ing to send a liveness message. Even for RTSP messages using reliable
   protocols, such as TCP, the message may take some time to arrive
   safely at the receiver. To show liveness between RTSP request with
   other effects, the following mechanisms can be used, in descending
   order of preference:

     RTCP: Is used to report transport statistics and SHALL also work as
          keep alive. The server can determine the client by used net-
          work 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 gives lower statistical guarantees to reach the
          server. However that probability is so little that it can be
          ignored in most cases. For example, a session with 60 seconds
          timeout and enough bitrate assigned to the RTCP messages, so
          the client sends a message on average every 5 seconds.  That
          session have for a network with 5 % packet loss the probabil-
          ity to not get a liveness sign over to the server in the time-
          out interval is 2.4*E-16. In sessions with shorter timeout
          times, or much higher packet loss, or small RTCP bandwidths
          SHOULD use any of the mechanisms below.

     PING: The use of the PING method is the best of the RTSP based
          methods.  It has no other effects than updating the timeout
          timer. In that way it will be a minimal message, that also
          does not cause any extra processing for the server. The down-
          side is that it may not be implemented. A client SHOULD use a
          OPTIONS request to verify support of the PING at the server.
          It is possible to detect support by sending a PING to the
          server. If a 200 (OK) message is received the server supports
          it. In case a 501 (Not Implemented) is received it does not
          support PING and there is no meaning in continue trying. Also
          the reception of a error message will also mean that the live-
          ness timer is not updated.

     SET_PARAMETER: When using SET_PARAMETER for keep alive, no body
          SHOULD be included. This method is basically as good as PING,
          however the implementation support of the method is today lim-
          ited. The same considerations as for PING apply regarding



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          checking of support in server and proxies.

     OPTIONS: This method does also work. However it causes the server
          to perform unnecessary processing and result in bigger
          responses than necessary for the task. The reason for this is
          that the Public is always included creating overhead.

   Note that a session identifier identifies an RTSP session across
   transport sessions or connections. Control messages for more than one
   RTSP URL may be sent within a single RTSP session. Hence, it is pos-
   sible that clients use the same session for controlling many streams
   constituting a presentation, as long as all the streams come from the
   same server. (See example in Section 15). However, multiple "user"
   sessions for the same URL from the same client MUST use different
   session identifiers.

     The session identifier is needed to distinguish several deliv-
     ery requests for the same URL coming from the same client.

   The response 454 (Session Not Found) is returned if the session iden-
   tifier is invalid.

13.38 Supported

   The Supported header field enumerates all the extensions supported by
   the client or server. When offered in a request, the receiver MUST
   respond with its corresponding Supported header.

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


        Supported  =  "Supported" ":" [feature-tag *("," feature-tag)]      ||

   Example:                                                              |

     C->S:  OPTIONS rtsp://example.com/ RTSP/1.0                         |
            Supported: foo, bar, blech                                   |

     S->C:  RTSP/1.0 200 OK                                              |
            Supported: bar, blech, baz                                   |



13.39 Timestamp

   The Timestamp general-header field describes when the client sent the
   request to the server. The value of the timestamp is of significance



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   only to the client and may use any timescale. The server 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 client to compute the round-trip time to the server so
   that it can adjust the timeout value for retransmissions. It also
   resolves retransmission ambiguities for unreliable transport of RTSP.


   Timestamp  =  "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
   delay      =  *(DIGIT) [ "." *(DIGIT) ]


13.40 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 des-
   tination 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 Transport header field MAY also be used to change certain trans-
   port parameters. A server MAY refuse to change parameters of an
   existing stream.

   The server MAY return a Transport response-header field in the
   response to indicate the values actually chosen.

   A Transport request header field MAY contain a list of transport
   options acceptable to the client, in the form of multiple transport-
   spec entries. In that case, the server MUST return the single option
   (transport-spec) which was actually chosen.

   A transport-spec transport option may only contain one of any given
   parameter within it. Parameters may be given in any order.  Addition-
   ally, it may only contain the unicast or multicast transport parame-
   ter.


     The Transport header field is restricted to describing a sin-
     gle 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.




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   The syntax for the transport specifier is

   transport/profile/lower-transport.


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

   Below are the configuration parameters associated with transport:

   General parameters:

     unicast / multicast: This parameter is a mutually exclusive indica-
          tion 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 trans-
          mission MUST indicate such capability by including two full
          transport-specs with separate parameters for each.

     destination: The address of the stream recipient to which a stream
          will be sent. The client originating the RTSP request may
          specify the destination address of the stream recipient with
          the destination parameter. When the destination field is spec-
          ified, the recipient may be a different party than the origi-
          nator of the request. To avoid becoming the unwitting perpe-
          trator of a remote-controlled denial-of-service attack, a
          server SHOULD authenticate the client originating the request
          and SHOULD log such attempts before allowing the client to
          direct a media stream to a recipient address not chosen by the
          server. While, this is particularly important if RTSP commands
          are issued via UDP, implementations cannot rely on TCP as
          reliable means of client identification by itself either.

          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 recipi-
          ent 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 communi-
          cate 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.

          IPv6 addresses are RECOMMENDED to be given as fully qualified
          domain to make it backwards compatible with RFC 2326



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          implementations.

     source: If the source address for the stream is different than can
          be derived from the RTSP endpoint address (the server in play-
          back), the source address SHOULD be specified. To maintain
          backwards compatibility with RFC 2326, any IPv6 host's address
          must be given as a fully qualified domain name.


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

     layers: The number of multicast layers to be used for this media
          stream. The layers are sent to consecutive addresses starting
          at the destination address.


     dest_addresses: A general destination address parameter that can    |
          contain one or more address and port pair. For each combina-   |
          tion of Protocol/Profile/Lower Transport the interpretation of |
          the address or addresses needs to be defined. The client or    |
          server SHALL NOT use this parameter unless both client and     |
          server has shown support. This parameter MUST be supported by  |
          client and servers that implements this specification. Support |
          is indicated by the use of the feature-tag "play.basic". This  |
          parameter SHALL NOT be used in the same transport specifica-   |
          tion as any of the parameters "destination", "source", "port", |
          "client_port", and "server_port".                              |

          The same security consideration that are given for the "Desti- |
          nation" parameter does also applies to this parameter. This    |
          parameter can be used for redirecting traffic to recipient not |
          desiring the media traffic.                                    |

     src_addresses: A General source address parameter that can contain  |
          one or more address and port pair. For each combination of     |
          Protocol/Profile/Lower Transport the interpretation of the     |
          address or addresses needs to be defined. The client or server |
          SHALL NOT use this parameter unless both client and server has |
          shown support. This parameter MUST be supported by client and  |
          servers that implements this specification. Support is indi-   |
          cated by the use the feature-tag "play.basic". This parameter  |
          SHALL NOT be used in the same transport specification as any   |
          of the parameters "destination", "source", "port",             |
          "client_port", and "server_port".                              |




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          The address or addresses indicated in the src_addresses param- |
          eter SHOULD be used both for sending and receiving of the      |
          media streams data packet. The main reasons are two: First by  |
          sending from the indicated ports the source address will be    |
          known by the receiver of the packet. Secondly, in the presence |
          of NATs some traversal mechanism requires either knowledge     |
          from which address and port a packet flow is coming, or having |
          the possibility to send data to the sender port.

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

     append: The append parameter was used together with RECORD and is
          now deprecated.

     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 11.11. 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 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.

   RTP-specific:

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





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     port: This parameter provides the RTP/RTCP port pair for a multi-
          cast session. It is should be specified as a range, e.g.,
          port=3456-3457

     client_port: This parameter provides the unicast RTP/RTCP port pair
          on the client where media data and control information is to
          be sent. It is specified as a range, e.g., port=3456-3457

     server_port: This parameter provides the unicast RTP/RTCP port pair
          on the server where media data and control information is to
          be sent. It is specified as a range, e.g., port=3456-3457

     ssrc: The ssrc parameter indicates the RTP SSRC [23] value that
          should be (request) or will be (response) used by the media
          server. This parameter is only valid for unicast transmission.
          It identifies the synchronization source to be associated with
          the media stream, and is expressed as an eight digit hexideci-
          mal value. In cases that a sender will use multiple SSRCs it
          SHOULD NOT use this parameter.

     client_ssrc: The client_ssrc parameter indicates the RTP SSRC [23]
          value that will be used by the client. This parameter is only
          valid for unicast transmission. It identifies the synchroniza-
          tion source to be associated with the media stream, and is
          expressed as an eight digit hexidecimal value. In cases that a
          client will use multiple SSRCs it SHOULD NOT use this parame-
          ter.

Transport                =  "Transport" ":" 1#transport-spec             ||
transport-spec           =  transport-id *parameter                      ||
transport-id             =  transport-protocol "/" profile ["/" lower-transport]||
                            ; no LWS is allowed inside transport-id      ||
transport-protocol       =  "RTP" / token                                ||
profile                  =  "AVP" / token                                ||
lower-transport          =  "TCP" / "UDP" / token                        ||
parameter                =  ";" ( "unicast" / "multicast" )              ||
                        /   ";" "source" "=" host                        ||
                        /   ";" "destination" [ "=" host ]               ||
                        /   ";" "interleaved" "=" channel [ "-" channel ]||
                        /   ";" "append"                                 ||
                        /   ";" "ttl" "=" ttl                            ||
                        /   ";" "layers" "=" 1*DIGIT                     ||
                        /   ";" "port" "=" port-spec                     ||
                        /   ";" "client_port" "=" port-spec              ||
                        /   ";" "server_port" "=" port-spec              ||
                        /   ";" "ssrc" "=" ssrc                          ||
                        /   ";" "client_ssrc" "=" ssrc                   ||




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                        /   ";" "mode" "=" mode-spec                     ||
                        /   ";" "dest_addresses" "=" addr-list           ||
                        /   ";" "src_addresses" "=" addr-list            ||
                        /   ";" trn-parameter-extension                  ||
port-spec                =  port [ "-" port ]                            ||
trn-parameter-extension  =  par-name "=" trn-par-value                   ||
par-name                 =  token                                        ||
trn-par-value            =  *unreserved                                  ||
ttl                      =  1*3(DIGIT)                                   ||
ssrc                     =  8*8(HEX)                                     ||
channel                  =  1*3(DIGIT)                                   ||
mode-spec                =  <"> 1#mode <"> / mode                        ||
mode                     =  "PLAY" / "RECORD" / token                    ||
addr-list                =  host-port *("/" host-port)                   ||
host-port                =  host [":" port]                              ||
host                     =  see chapter  16                              ||
port                     =  see chapter  16                              ||


   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/1.0
           CSeq: 302
           Transport: RTP/AVP;multicast;mode="PLAY",
               RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"

     S->C: RTSP/1.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Session: 47112344
           Transport: RTP/AVP;unicast;client_port=3456-3457;
               server_port=6256-6257;mode="PLAY"



13.41 Unsupported

   The Unsupported response-header field lists the features not sup-
   ported by the server. In the case where the feature was specified via
   the Proxy-Require field (Section 13.27), if there is a proxy on the



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   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 13.32 for a usage example.


   Unsupported  =  "Unsupported" ":" feature-tag *("," feature-tag)


13.42 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.


   User-Agent           =  "User-Agent" ":" ( product / comment ) 0*(SP
   (product / comment)


13.43 Vary

   See [H14.44]

13.44 Via

   See [H14.45].

13.45 WWW-Authenticate

   See [H14.47].

14 Caching

   In HTTP, response-request pairs are cached. RTSP differs signifi-     |
   cantly in that respect. Responses are not cacheable, with the excep-  |
   tion 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.) How-    |
   ever, it is desirable for the continuous media data, typically deliv- |
   ered 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 issu-
   ing a SETUP or DESCRIBE request, respectively, and comparing the



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   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 possi-
   bly making a local copy for later reuse. The exact behavior allowed
   to the cache is given by the cache-response directives described in
   Section 13.9. A cache MUST answer any DESCRIBE requests if it is cur-
   rently 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.

   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 addi-
   tional 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, 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.

15 Examples

   The following examples refer to stream description formats that are
   not standards, such as RTSL. The following examples are not to be
   used as a reference for those formats.

15.1 Media on Demand (Unicast)

   Client C requests a movie from 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 presen-
   tation and all its streams, including the codecs that are available,
   dynamic RTP payload types, the protocol stack, and content informa-
   tion 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.



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

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

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         s=RTSP Session
         e=adm@example.com
         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/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057

   A->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001

   C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 1
         User-Agent: PhonyClient/1.2
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059

   V->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
                    server_port=5002-5003

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 23456789



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         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://video.example.com/twister/video;
       seq=12312232;rtptime=78712811

   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 12345678
         Range: smpte=0:10:00-

   A->C: RTSP/1.0 200 OK
         CSeq: 2
         User-Agent: PhonyClient/1.2
         Session: 12345678
         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
       seq=876655;rtptime=1032181

   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 12345678

   A->C: RTSP/1.0 200 OK
         CSeq: 3

   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 3
         User-Agent: PhonyClient/1.2
         Session: 23456789

   V->C: RTSP/1.0 200 OK
         CSeq: 3



   Even though the audio and video track are on two different servers,
   and may start at slightly different times and may drift with respect
   to each other, the client can synchronize the two using standard RTP
   methods, in particular the time scale contained in the RTCP sender
   reports.

15.2 Streaming of 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 streams.



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

   The following is an example of using a single RTSP session to control
   multiple streams. It also illustrates the use of aggregate URLs.

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


   C->M: DESCRIBE rtsp://example.com/twister RTSP/1.0
         CSeq: 1

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Date: 23 Jan 1997 15:35:06 GMT
         Content-Type: application/sdp
         Content-Length: 164

         v=0
         o=- 2890844256 2890842807 IN IP4 172.16.2.93
         s=RTSP Session
         i=An Example of RTSP Session Usage
         e=adm@example.com
         a=control:rtsp://example.com/twister
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://example.com/twister/audio
         m=video 0 RTP/AVP 26
         a=control:rtsp://example.com/twister/video

   C->M: SETUP rtsp://example.com/twister/audio RTSP/1.0
         CSeq: 2



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         Transport: RTP/AVP;unicast;client_port=8000-8001

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;unicast;client_port=8000-8001;
                    server_port=9000-9001
         Session: 12345678

   C->M: SETUP rtsp://example.com/twister/video RTSP/1.0
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003;
                    server_port=9004-9005
         Session: 12345678

   C->M: PLAY rtsp://example.com/twister RTSP/1.0
         CSeq: 4
         Range: npt=0-
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 4
         Session: 12345678
         Range: npt=0-
         RTP-Info: url=rtsp://example.com/twister/video;
       seq=12345;rtptime=3450012,
       url=rtsp://example.com/twister/audio;
       seq=54321;rtptime=2876889

   C->M: PAUSE rtsp://example.com/twister/video RTSP/1.0
         CSeq: 5
         Session: 12345678

   M->C: RTSP/1.0 460 Only aggregate operation allowed
         CSeq: 5

   C->M: PAUSE rtsp://example.com/twister RTSP/1.0
         CSeq: 6
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 6
         Session: 12345678




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   C->M: SETUP rtsp://example.com/twister RTSP/1.0
         CSeq: 7
         Transport: RTP/AVP;unicast;client_port=10000
         Session: 12345678

   M->C: RTSP/1.0 459 Aggregate operation not allowed
         CSeq: 7




   In the first instance of failure, the client tries to pause one
   stream (in this case video) of the presentation. This is not allowed
   as this session is set up for aggregated control. In the second
   instance, the aggregate URL may not be used for SETUP and one control
   message is required per stream to set up transport parameters.

     This keeps the syntax of the Transport header simple and
     allows easy parsing of transport information by firewalls.

15.3 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 SHOULD use the rules set forth in the session descrip-
   tion for request URLs, rather than assuming that a consistent URL may
   always be used throughout. Here's 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/1.0
             Accept: application/x-rtsp-mh, application/sdp
             CSeq: 1

       S->C  RTSP/1.0 200 OK
             CSeq: 1
             Content-base: rtsp://foo.com/test.wav/
             Content-type: application/sdp
             Content-length: 48

             v=0
             o=- 872653257 872653257 IN IP4 172.16.2.187
             s=mu-law wave file
             i=audio test
             t=0 0
             m=audio 0 RTP/AVP 0
             a=control:streamid=0




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       C->S  SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
             Transport: RTP/AVP/UDP;unicast;
                        client_port=6970-6971;mode="PLAY"
             CSeq: 2

       S->C  RTSP/1.0 200 OK
             Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
                        server_port=6970-6971;mode="PLAY"
             CSeq: 2
             Session: 2034820394

       C->S  PLAY rtsp://foo.com/test.wav RTSP/1.0
             CSeq: 3
             Session: 2034820394

       S->C  RTSP/1.0 200 OK
             CSeq: 3
             Session: 2034820394
             Range: npt=0-600
             RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
               seq=981888;rtptime=3781123



   Note the different URL in the SETUP command, and then the switch back
   to the aggregate URL 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.

   In this special case, it is recommended that servers be forgiving of
   implementations that send:


       C->S  PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
             CSeq: 3



   In the worst case, servers should send back:


       S->C  RTSP/1.0 460 Only aggregate operation allowed
             CSeq: 3







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   One would also hope that server implementations are also forgiving of
   the following:


       C->S  SETUP rtsp://foo.com/test.wav RTSP/1.0
             Transport: rtp/avp/udp;client_port=6970-6971;mode="PLAY"
             CSeq: 2



   Since there is only a single stream in this file, it's not ambiguous
   what this means.

15.4 Live Media Presentation Using Multicast

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


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

   W->C: HTTP/1.1 200 OK
         Content-Type: application/x-rtsl

         <session>
           <track src="rtsp://live.example.com/concert/audio">
         </session>

   C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 1

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 44

         v=0
         o=- 2890844526 2890842807 IN IP4 192.16.24.202
         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

   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;multicast



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   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;multicast;destination=224.2.0.1;
                    port=3456-3457;ttl=16
         Session: 0456804596

   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 3
         Session: 0456804596

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 0456804596
         Range:npt=now-



16 Syntax

   The RTSP syntax is described in an augmented Backus-Naur form (BNF)
   as defined in RFC 2234 [14]. Also the "#" rule from RFC 2616 [26] is
   also defined and used in this syntax description.




16.1 Base Syntax


   OCTET           =  <any 8-bit sequence of data>
   CHAR            =  <any US-ASCII character (octets 0 - 127)>
   UPALPHA         =  <any US-ASCII uppercase letter "A".."Z">
   LOALPHA         =  <any US-ASCII lowercase letter "a".."z">
   ALPHA           =  UPALPHA / LOALPHA
   DIGIT           =  <any US-ASCII digit "0".."9">
   CTL             =  <any US-ASCII control character
                      (octets 0 - 31) and DEL (127)>
   CR              =  <US-ASCII CR, carriage return (13)>
   LF              =  <US-ASCII LF, linefeed (10)>
   SP              =  <US-ASCII SP, space (32)>
   HT              =  <US-ASCII HT, horizontal-tab (9)>
   <">             =  <US-ASCII double-quote mark (34)>
   BACKSLASH       =  <US-ASCII backslash (92)>
   CRLF            =  CR LF
   LWS             =  [CRLF] 1*( SP / HT )
   TEXT            =  <any OCTET except CTLs>
   tspecials       =  "(" / ")" / "<" / ">" / "@"




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                  /   "," / ";" / ":" / BACKSLASH / <">
                  /   "/" / "[" / "]" / "?" / "="
                  /   "{" / "}" / SP / HT
   token           =  1*<any CHAR except CTLs or tspecials>
   quoted-string   =  ( <"> *(qdtext) <"> )
   qdtext          =  <any TEXT except <">>
   quoted-pair     =  BACKSLASH CHAR
   message-header  =  field-name ":" [ field-value ] CRLF
   field-name      =  token
   field-value     =  *( field-content / LWS )
   field-content   =  <the OCTETs making up the field-value and
                     consisting
                     of either *TEXT or combinations of token, tspecials,
                     and quoted-string>
   safe            =  "$" / "-" / "_" / "." / "+"
   extra           =  "!" / "*" / "'" / "(" / ")" / ","
   hex             =  DIGIT / "A" / "B" / "C" / "D" / "E" / "F" /
                      "a" / "b" / "c" / "d" / "e" / "f"
   escape          =  "%" hex hex
   reserved        =  ";" / "/" / "?" / ":" / "@" / "&" / "="
   unreserved      =  alpha / digit / safe / extra
   xchar           =  unreserved / reserved / escape


16.2 RTSP Protocol Definition

16.2.1 Message Syntax



   generRTSP-message=  st=rtRequest / Response ; RTSP/1.0 messages
                       *(message-header CRLF)
                       CRLF
                       [ message-body ]
   start-line       =  Request-Line / Status-Line




        Request  g=nerRequest-Line ; Sec;iSection 6.1
             /   request-header    ; Section 6.2
             /   entity-header )   ; Section 8.1
                 CRLF
                 [ message-body ]  ; Section 4.3
   Response  =   Status-Line       ; Section 7.1
             *(  general-header    ; Section 5
             /   response-header   ; Section 7.1.2




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             /   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
           /  "PING"            ; Section 11.10
           /  "REDIRECT"        ; Section 11.9
           /  "SETUP"           ; Section 11.3
           /  "SET_PARAMETER"   ; Section 11.8
           /  "TEARDOWN"        ; Section 11.6
           /  extension-method



   extension-method  =  token
   Request-URI       =  "*" / absolute_URI
   RTSP-Version      =  "RTSP" "/" 1*DIGIT "." 1*DIGIT




     Status-Code  =  "100"           ; Continue
                  /  "200"           ; OK
                  /  "201"           ; Created
                  /  "250"           ; Low on Storage Space
                  /  "300"           ; Multiple Choices
                  /  "301"           ; Moved Permanently
                  /  "302"           ; Moved Temporarily
                  /  "303"           ; See Other
                  /  "304"           ; Not Modified
                  /  "305"           ; Use Proxy
                  /  "400"           ; Bad Request
                  /  "401"           ; Unauthorized
                  /  "402"           ; Payment Required
                  /  "403"           ; Forbidden
                  /  "404"           ; Not Found




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                  /  "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
                  /  "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, excluding CR, LF>



   general-header  =  Cache-Control      ; Section 13.9
                   /  Connection         ; Section 13.10
                   /  CSeq               ; Section 13.17
                   /  Date               ; Section 13.18
                   /  Timestamp          ; Section 13.39
                   /  Via                ; Section 13.44
   request-header  =  Accept             ; Section 13.1
                   /  Accept-Encoding    ; Section 13.2
                   /  Accept-Language    ; Section 13.3




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                   /  Authorization      ; Section 13.6
                   /  Bandwidth          ; Section 13.7
                   /  Blocksize          ; Section 13.8
                   /  From               ; Section 13.20
                   /  If-Modified-Since  ; Section 13.23
                   /  Proxy-Require      ; Section 13.27
                   /  Range              ; Section 13.29
                   /  Referer            ; Section 13.30
                   /  Require            ; Section 13.32
                   /  Scale              ; Section 13.34
                   /  Session            ; Section 13.37
                   /  Speed              ; Section 13.35
                   /  Supported          ; Section 13.38
                   /  Transport          ; Section 13.40
                   /  User-Agent         ; Section 13.42



   response-header  =  Accept-Ranges       ; Section 13.4
                    /  Location            ; Section 13.25
                    /  Proxy-Authenticate  ; Section 13.26
                    /  Public              ; Section 13.28
                    /  Range               ; Section 13.29
                    /  Retry-After         ; Section 13.31
                    /  RTP-Info            ; Section 13.33
                    /  Scale               ; Section 13.34
                    /  Session             ; Section 13.37
                    /  Server              ; Section 13.36
                    /  Speed               ; Section 13.35
                    /  Transport           ; Section 13.40
                    /  Unsupported         ; Section 13.41
                    /  Vary                ; Section 13.43
                    /  WWW-Authenticate    ; Section 13.45



   rtsp_URL          =  ( "rtsp:" / "rtspu:" / "rtsps" )
                        "//" host [ ":" port ] [ abs_path ] [ "#" fragment ]
   host              =  As defined by RFC 2732 [30]
   abs_path          =  As defined by RFC 2396 [22]
   port              =  *DIGIT
   smpte-range       =  smpte-type "=" smpte-range-spec
   smpte-range-spec  =  ( smpte-time "-" [ smpte-time ] ) / ( "-" smpte-time )
   smpte-type        =  "smpte" / "smpte-30-drop" / "smpte-25"
                        ; other timecodes may be added
   smpte-time        =  1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
                        [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]




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   npt-range       =  ["npt" "="] npt-range-spec
                      ; implementations SHOULD use npt= prefix, but SHOULD
                      ; be prepared to interoperate with RFC 2326
                      ; implementations which don't use it
   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
   utc-range-spec  =  ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
   utc-time        =  utc-date "T" utc-time "Z"
   utc-date        =  8DIGIT ; < YYYYMMDD >
   utc-time        =  6DIGIT [ "." fraction ]; < HHMMSS.fraction >
   fraction        =  1*DIGIT



   feature-tag  =  token


16.2.2 Header Syntax


   Transport                =  "Transport" ":" 1#transport-spec
   transport-spec           =  transport-id *parameter
   transport-id             =  transport-protocol "/" profile ["/" lower-transport]
                               ; no LWS is allowed inside transport-id
   transport-protocol       =  "RTP" / token
   profile                  =  "AVP" / token
   lower-transport          =  "TCP" / "UDP" / token
   parameter                =  ";" ( "unicast" / "multicast" )
                           /   ";" "source" "=" host
                           /   ";" "destination" [ "=" host ]
                           /   ";" "interleaved" "=" channel [ "-" channel ]
                           /   ";" "append"
                           /   ";" "ttl" "=" ttl
                           /   ";" "layers" "=" 1*DIGIT
                           /   ";" "port" "=" port-spec
                           /   ";" "client_port" "=" port-spec
                           /   ";" "server_port" "=" port-spec
                           /   ";" "ssrc" "=" ssrc
                           /   ";" "client_ssrc" "=" ssrc
                           /   ";" "mode" "=" mode-spec
                           /   ";" "dest_addresses" "=" addr-list




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                           /   ";" "src_addresses" "=" addr-list
                           /   ";" trn-parameter-extension
   port-spec                =  port [ "-" port ]
   trn-parameter-extension  =  par-name "=" trn-par-value
   par-name                 =  token
   trn-par-value            =  *unreserved
   ttl                      =  1*3(DIGIT)
   ssrc                     =  8*8(HEX)
   channel                  =  1*3(DIGIT)
   mode-spec                =  <"> 1#mode <"> / mode
   mode                     =  "PLAY" / "RECORD" / token
   addr-list                =  host-port *("/" host-port)
   host-port                =  host [":" port]
   host                     =  see chapter  16
   port                     =  see chapter  16


17 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:

     Authentication Mechanisms: RTSP and HTTP share common authentica-
          tion schemes, and thus should follow the same prescriptions
          with regards to authentication . See chapter 15.1 of [2] for
          client authentication issues, and chapter 15.2 of [2] for
          issues regarding support for multiple authentication mecha-
          nisms. Also see [H15.6].

     Abuse of Server Log Information: RTSP and HTTP servers will presum-
          ably 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 sensi-
          tive than that normally transmitted via HTTP. Therefore, all
          of the precautions regarding the protection 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 URLs are opaque
          handles that do not necessarily have file system semantics, it
          is anticipated that many implementations will translate por-
          tions of the request URLs directly to file system calls. In



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          such cases, file systems SHOULD follow the precautions out-
          lined in [H15.5], such as checking for ".." in path compo-
          nents.

     Personal Information: RTSP clients are often privy to the same
          information that HTTP clients are (user name, location, etc.)
          and thus should be equally. See [H15.1] for further recommen-
          dations.

     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 fol-
          lowed.

     DNS Spoofing: Presumably, given the longer connection times typi-
          cally 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 one another, then it must
          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 invali-
          date resources over which they have no authority. ([H15.4])

   In addition to the recommendations in the current HTTP specification
   (RFC 2616 [26], as of this writing) and also of the previous RFC2068
   [2], 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.

          The attacker may initiate traffic flows to one or more IP
          addresses 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
          SHOULD only allow client-specified destinations for RTSP-ini-
          tiated traffic flows if the server has verified the client's
          identity, either against a database of known users using RTSP
          authentication mechanisms (preferably digest authentication or
          stronger), or other secure means.



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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.

     Authentication: Servers SHOULD implement both basic and digest [6]
          authentication. In environments requiring tighter security for
          the control messages, transport layer mechanisms such as TLS
          (RFC 2246 [27]) SHOULD be used.

     Stream issues: RTSP only provides for stream control. Stream deliv-
          ery 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.

18 IANA Considerations

   This section set up a number of registers for RTSP that should be
   maintained by IANA. For each registry there is a description on what
   it shall contain, what specification is needed when adding a entry
   with IANA, and finally the entries that this document needs to regis-
   ter. See also the section 1.6 "Extending RTSP".

   The sections describing how to register an item uses some of the
   requirements level described in RFC 2434 [29], namely " First Come,
   First Served", "Specification Required", and "Standards Action".

   A registration request to IANA MUST contain the following informa-
   tion:

     + A name of the item to register according to the rules specified
       by the intended registry.

     + Indication of who has change control over the feature (for exam-
       ple, IETF, ISO, ITU-T, other international standardization bod-
       ies, a consortium or a particular company or group of companies);



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     + A reference to a further description, if available, for example
       (in order of preference) an RFC, a published standard, a pub-
       lished paper, a patent filing, a technical report, documented
       source code or a computer manual;

     + For proprietary features, contact information (postal and email
       address);

18.1 Feature-tags

18.1.1 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 con-
   tains named entries representing certain functionality.

   The usage of feature-tags is explained in section 10 and 11.1.

18.1.2 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.  Any propri-
   etary feature SHALL have as the first part of the name a vendor tag,
   which identifies the organization.

18.1.3 Registered entries

   The following feature-tags are in this specification defined and
   hereby registered. The change control belongs to the Authors and the
   IETF MMUSIC WG.

     play.basic: The minimal implementation for playback operations
          according to section D.

     play.scale: Support of scale operations for media playback.

     play.speed: Support of the speed functionality for playback.

     setup.playing: The use of SETUP and TEARDOWN in play state.

     con.persistent: Support and use of persistent connections, see
          chapter  9.3.




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18.2 RTSP Methods

18.2.1 Description

   What a method is, is described in section 11.  Extending the protocol
   with new methods allow for totally new functionality.

18.2.2 Registering New Methods with IANA

   A new method MUST be registered through an IETF standard track docu-
   ment. The reason is that new methods may radically change the proto-
   cols behavior and purpose.

   A specification for a new RTSP method MUST consist of the following
   items:

     + A method name which follows the BNF rules for methods.

     + 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.

     + A list or table specifying which of the registered headers that
       are allowed to use with the method in request or/and response.

     + Describe how the method relates to network proxies.

18.2.3 Registered Entries

   This specification, RFCXXXX, registers 10 methods: DESCRIBE,
   GET_PARAMETER, OPTIONS, PAUSE, PING, PLAY, REDIRECT, SETUP,
   SET_PARAMETER, and TEARDOWN.

18.3 RTSP Status Codes

18.3.1 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.

18.3.2 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 fol-
   lowing:




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     + The requested number.

     + A description what the status code means and the expected behav-
       ior of the sender and receiver of the code.

18.3.3 Registered Entries

   RFCXXX, registers the numbered status code defined in the BNF entry
   "Status-Code" except "extension-code" in section 7.1.1.

18.4 RTSP Headers

18.4.1 Description

   By specifying new headers a method(s) can be enhanced in many differ-
   ent 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 fea-
   ture-tag.

18.4.2 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:

     + The name of the header.

     + A BNF specification of the header syntax.

     + A list or table specifying when the header may be used, encom-
       passing all methods, their request or response, the direction
       (C->S or S->C).

     + How the header shall be handled by proxies.

     + A description of the purpose of the header.

18.4.3 Registered entries

   All headers specified in section 13 in RFCXXXX are to be registered.

   Furthermore the following RTSP headers defined in other specifica-
   tions are registered:




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     + x-wap-profile defined in [35].

     + x-wap-profile-diff defined in [35].

     + x-wap-profile-warning defined in [35].

     + x-predecbufsize defined in [35].

     + x-initpredecbufperiod defined in [35].

     + x-initpostdecbufperiod defined in [35].

       Note: The use of "X-" is NOT RECOMMENDED but the above headers in
       the register list was defined prior to the clarification.

18.5 Transport Header registries

   The transport header contains a number of parameters which have pos-
   sibilities for future extensions. Therefore registries for these must
   be defined.

18.5.1 Transport Protocols

   A registry for the parameter transport-protocol shall be defined with
   the following rules:

     + Registering requires public available standards specification.

     + A contact person or organization with address and email.

     + A value definition that are following the BNF token definition.

     + A describing text that explains how the registered value are used
       in RTSP.

   This specification register 1 value:

     + Use of the RTP  [23] protocol for media transport. The usage is
       explained in RFC XXXX, appendix B.1.

18.5.2 Profile

   A registry for the parameter profile shall be defined with the fol-
   lowing rules:

     + Registering requires public available standards specification.





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     + A contact person or organization with address and email.

     + A value definition that are following the BNF token definition.

     + A definition of which Transport protocol(s) that this profile is
       valid for.

     + A describing text that explains how the registered value are used
       in RTSP.

     + The "RTP profile for audio and video conferences with minimal
       control"  [1] MUST only be used when the transport headers trans-
       port-protocol is "RTP".

18.5.3 Lower Transport

   A registry for the parameter lower-transport shall be defined with
   the following rules:

     + Registering requires public available standards specification.

     + A contact person or organization with address and email.

     + A value definition that are following the BNF token definition.

     + A describing text that explains how the registered value are used
       in RTSP. This includes

     + Indicates the use of the "User datagram protocol"  [7] for media
       transport.

     + Indicates the use Transmission control protocol  [9] for media
       transport.

18.5.4 Transport modes

   A registry for the transport parameter mode shall be defined with the
   following rules:

     + Registering requires a IETF standard tracks document.

     + A contact person or organization with address and email.

     + A value definition that are following the BNF token definition.

     + A describing text that explains how the registered value are used
       in RTSP.




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     + See RFC XXXX.

     + See RFC XXXX.

18.6 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:

     + Registering requires a IETF standard tracks document.

     + A registration shall name a contact person.

     + Name of the directive and a definition of the value, if any.

     + A describing text that explains how the cache directive is used
       for RTSP controlled media streams.

A RTSP Protocol State Machine

   The RTSP session state machine describe the behavior of the protocol
   from RTSP session initialization through RTSP session termination.

   State machine is defined on a per session basis which is uniquely
   identified by the RTSP session identifier. The session may contain
   zero 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.

   This state machine is one possible representation that helps explain
   how the protocol works and when different requests are allowed.  We
   find it a reasonable representation but does not mandate it, and
   other representations can be created.

A.1 States

   The state machine contains five 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-nm: Ready state without any medias.

     Ready: Session is ready to start playing.





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     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
   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 shall change to.





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   The response to valid request meeting the requisites is normally a
   2xx (SUCCESS) unless other noted in the response column. The excep-
   tions shall 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 3xx | 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 unre-
   coverable 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 SHALL assume that the session has been ended.

   The server will timeout the session after the period of time speci-
   fied 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 URL is equal to the media
   URL. For NRM>1 the presentation URL MUST be other than any of the
   medias that are part of the session. This applies to all states.





   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 6: None state-machine changing events


   The methods in Table 6 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.




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            Action  Requisite       New State  Response
            -------------------------------------------------
            SETUP                     Ready    NRM=1, RP=0.0
            SETUP   Needs Redirect    Init     3rr Redirect


   Table 7: State: Init


   The initial state of the state machine, see Table 7 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
   shows that a correct SETUP can in some cases be redirected to another
   URL and/or server by a 3rr response.


       Action         Requisite       New State  Response
       ----------------------------------------------------------
       SETUP                            Ready    NRM=1,RP=0.0
       SETUP          Needs Redirect    Init     3rr
       TEARDOWN       URL=*             Init     No session hdr.
       Timeout                          Init
       S->C:REDIRECT  Range hdr       Ready-nm   Set RedP
       S->C:REDIRECT  no range hdr      Init
       RedP reached                     Init


   Table 8: State: Ready-nm


   The optional Ready-nm state has no media streams and therefore can't  |
   play.  This state exist so that all session related parameters and    |
   resources can be kept while changing media stream(s). As seen in      |
   Table 8 the operations are limited to setting up a new media or tear- |
   ing down the session. The established session can also be redirected  |
   with the REDIRECT method.


   In the Ready state, see Table 9, 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 stream (same
   URL) 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 either * or the pre-
   sentations URI the whole session is torn down. If a media URL 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



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  Action         Requisite          New State  Response
  ---------------------------------------------------------------------
  SETUP          New URL              Ready    NRM+=1
  SETUP          Setten up URL        Ready    Change transport param.
  TEARDOWN       URL=*                Init     No session hdr
  TEARDOWN       Prs URL,NRM>1        Init     No session hdr
  TEARDOWN       md URL,NRM=1IFI    Ready-nm   Session hdr, NRM=0
  TEARDOWN       md URL,NRM=1         Init     No Session hdr, NRM=0
  TEARDOWN       md URL,NRM>1         Ready    Session hdr, NRM-=1
  PLAY           Prs URL, No range    Play     Play from RP
  PLAY           Prs URL, Range       Play     according to range
  S->C:REDIRECT  Range hdr            Ready    Set RedP
  S->C:REDIRECT  no range hdr         Init
  Timeout                             Init
  RedP reached                        Init


   Table 9: State: Ready

   response. If only a single media stream remains in the session when
   performing a TEARDOWN with a media URL , it is optional to keep the
   session. If the session still exist after the request a Session MUST
   be returned in the response. The number of media streams remaining
   after tearing down a media stream determines the new state.


   The Play state table, see Table 10, is the largest. The table con-
   tains an number of request that has presentation URL as a prerequi-
   site on the request URL, this is due to the exclusion of non-aggre-
   gated 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 two auto-
   matic 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
   header. If the client sends request close in time to these transi-
   tions it must be prepared for getting error message as the state may
   or may not have changed.

   SETUP and TEARDOWN requests with media URLs in aggregated sessions
   may not be handled by the server as it is optional functionality. Use
   the service discovery mechanism with OPTIONS to find out in before-
   hand if the server implements it. If the functionality is not



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Action         Requisite          New State  Response
------------------------------------------------------------------------
PAUSE          PrsURL,No range      Ready    Set RP to present point
PAUSE          PrsURL,Range>now     Play     Set RP & PP to given point
PAUSE          PrsURL,Range<=now    Ready    Set RP to Range Hdr.
PP reached                          Ready    RP = PP
End of media   All media            Play     No action, RP = Invalid
End of media   >=1 Media plays      Play     No action
End of range                        Play     Set RP = End of range
SETUP          New URL,IFI          Play     NRM+=1, 200, *A
SETUP          New URL              Play     455
SETUP          Setuped URL          Play     455
SETUP          Setuped URL, IFI     Play     Change transport param.
TEARDOWN       URL=*                Init     No session hdr
TEARDOWN       Prs URL,NRM>1        Init     No session hdr
TEARDOWN       md URL,NRM=1,IFI   Ready-nm   Session hdr
TEARDOWN       md URL,NRM>1,IFI     Play     Session hdr
TEARDOWN       md URL               Play     455
S->C:REDIRECT  Range hdr            Play     Set RedP
S->C:REDIRECT  no range hdr         Init     Stop Media Playout
RedP reached                        Init     Stop Media playout
Timeout                             Init     Stop Media playout


   Table 10: State: Play, *A: RTP-Info and Range header

   implemented but still tried by the client a "501 Not Implemented"
   response SHALL be received.

B Media Transport Alternatives


   This chapter defines how certain combinations of protocols, profiles  |
   and lower transports are used. This includes the usage of the Trans-  |
   port header's general source and destination parameters               |
   "src_addresses" and "dst_addresses".                                  |

B.1 RTP                                                                  |

   This section defines the interaction and needed media transport sig-  |
   nalling in regards to the RTP protocol [23].                          |

   RTSP allows media clients to control selected, non-contiguous sec-    |
   tions of media presentations, rendering those streams with an RTP     |
   media layer[23]. 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.      |



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   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. First we play NPT 10 through 15, then 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 40,000 through 55,200.                    |


     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.                         |

   For certain datatypes, tight integration between the RTSP layer and   |
   the RTP layer will be necessary. This by no means precludes the above |
   restriction. Combined RTSP/RTP media clients should use the RTP-Info  |
   field to determine whether incoming RTP packets were sent before or   |
   after a seek.                                                         |

   For continuous audio, the server SHOULD set the RTP marker bit at the |
   beginning of serving a new PLAY request. This allows the client to    |
   perform playout delay adaptation.                                     |

   For scaling (see Section 13.34), 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 13.35) 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.                   |

   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 13.33) header pro-    |
   vides the first sequence number of the next segment.                  |

   Below the available RTP profiles and lower layer transports are given |
   together with the necessary rules on how to signal that combination.  |

B.1.1 AVP                                                                |

   The usage of the "RTP Profile for Audio and Video Conferences with    |
   Minimal Control" [1] when using RTP for media transport over differ-  |
   ent lower layer transport protocols are defined below in regards to   |
   RTSP.                                                                 |



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   On such case is defined within this document, the use of embedded     |
   (interleaved) binary data as defined in section  11.11.  The usage of |
   this method is indicated by include the "interleaved" parameter.      |

   When using embedded binary data the "src_addresses" and               |
   "dst_addresses" 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 descibes sending of RTP [23] over lower transport layer UDP |
   [7] according to the profile "RTP Profile for Audio and Video Confer- |
   ences with Minimal Control" defined in RFC 1890 [1].                  |

   This profiles requires that 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. Embedded (interleaved) data when RTSP messages is trans- |
   ported over UDP SHOULD NOT be performed.                              |

   The RTP/UDP and RTCP/UDP flows can be established in two ways using   |
   the Transport header's parameters. The way provided in RFC 2326 was   |
   to use the necessary parameters from the set of "source", "destina-   |
   tion", "client_port", and "server_port". This has the advantage of    |
   being compatible with all RTP capable RTSP servers and clients. How-  |
   ever this method does not provide a possibility to specify non-con-   |
   tinues port ranges for RTP and RTCP.  The other way is to use the     |
   parameters "src_addresses", and "dst_addresses". This method provides |
   total flexibility in specifying address and port number for each      |
   transport flow.  However the disadvantage is that it is not supported |
   by non-updated clients, i.e. clients not supporting the "play.basic"  |
   feature-tag.                                                          |

   When using the "source", "destination", "client_port", and            |
   "server_port" the packets are be addressed in the following way for   |
   media playback:                                                       |

     + RTP/UDP packet from the server to the client SHALL be sent to the |
       address specified in the "destination" parameter and first even   |
       port number given in client_port range. If there is only a single |
       port number given that MUST be given.                             |

     + The server SHOULD send its RTP/UDP packets from the address spec- |
       ified in "source" parameter and from the first even port number   |
       specified in "server_port" parameter.                             |

     + If there is specified a range in "client_port" parameter that     |
       contains at least two port numbers, the RTCP/UDP packets from     |



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       server to client SHALL be sent to address specified in the "des-  |
       tination" parameter and first odd port number part of the range   |
       specified in the client_port parameter.                           |

     + The Server SHOULD send its RTCP/UDP packets from the address      |
       specified in "source" parameter and from the first odd port num-  |
       ber specified in "server_port" parameter.                         |

     + RTCP/UDP packets from the client to the server SHALL be sent to   |
       the address specified in the "source" parameter and first odd     |
       port number given in client_port range.                           |

     + The client SHOULD send its RTCP/UDP packets from the address      |
       specified in "destination" parameter and from the first odd port  |
       number specified in "server_port" parameter.                      |

   The usage of "src_addresses" and "dst_addresses" parameters to spec-  |
   ify the address and port numbers are done in the following way for    |
   media playback, i.e. Mode=PLAY:                                       |

     + The "src_addresses" and "dst_addresses" parameters MUST contain   |
       either 1 or 2 address and port pairs.                             |

     + Each address and port pair MUST contain both and address and a    |
       port number.                                                      |

     + 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.                               |

     + 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 "dst_addresses" parameter.                                    |

     + 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 "dst_addresses" parameter. If no second pair is given RTCP |
       SHALL NOT be sent.                                                |

     + 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 "dst_addresses" parameter. If no second pair is given RTCP |
       SHALL NOT be sent.                                                |

     + RTP and RTCP Packets SHOULD be sent from the corresponding        |
       receiver port, i.e. RTCP packets from server should be sent from  |
       the "src_addresses" parameters second address port pair.          |




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B.1.3 AVP/TCP                                                            |

   Note that this combination is not yet defined using sperate TCP con-  |
   nections. However the use of embedded (interleaved) binary data       |
   transported on the RTSP connection is possible as specified in sec-   |
   tion  11.11. When using this declared combination of interleaved      |
   binary data the RTSP messages MUST be transported over TCP.           |

   A possible future for this profile would be to define the use of a    |
   combination of the two drafts "Connection-Oriented Media Transport in |
   SDP" [36] and "Framing RTP and RTCP Packets over Connection-Oriented  |
   Transport" [37].                                                      |

B.2 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 chapter provides the necessary steps that needs to be      |
   meet.                                                                 |

   The following things needs to be considered when adding a new proto-  |
   col of profile for use with RTSP:                                     |

     + 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.                        |

     + The useful combinations of protocol/profile/lower-layer needs to  |
       be defined and for each combination declare the necessary parame- |
       ters to use in the Transport header.                              |

     + For new media protocols the interaction with RTSP needs to be     |
       addressed. One important factor will be the media synchroniza-    |
       tion.                                                             |

   See the IANA section ( 18) on how to register the necessary           |
   attributes.                                                           |


C Use of SDP for RTSP Session Descriptions

   The Session Description Protocol (SDP, RFC 2327 [24]) may be used to
   describe streams or presentations in RTSP. This description is typi-
   cally returned in reply to a DESCRIBE request on a URL from a server
   to a client, received via HTTP from a server to a client.

   This appendix describes how an SDP file determines the operation of
   an RTSP session.  SDP provides no mechanism by which a client can



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   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).

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 2327 [24]):

C.1.1 Control URL

   The "a=control:" attribute is used to convey the control URL. This
   attribute is used both for the session and media descriptions. If
   used for individual media, it indicates the URL to be used for con-
   trolling that particular media stream. If found at the session level,
   the attribute indicates the URL for aggregate control.


   control-attribute  =  "a=" "control" ":" url


   Example:

     a=control:rtsp://example.com/foo



   This attribute may contain either relative and absolute URLs, follow-
   ing the rules and conventions set out in RFC 2396 [22]. Implementa-
   tions should look for a base URL in the following order:

     1.   the RTSP Content-Base field;

     2.   the RTSP Content-Location field;

     3.   the RTSP request URL.

   If this attribute contains only an asterisk (*), then the URL is
   treated as if it were an empty embedded URL, and thus inherits the
   entire base URL.

   For SDP retrieved from a container file, there are certain things to
   consider. Lets say that the container file has the following URL:
   "rtsp://example.com/container.mp4". A media level relative URL needs
   to contain the file name container.mp4 in the beginning to be
   resolved correctly relative to the before given URL. An alternative
   if one does not desire to enter the container files name is to ensure



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   that the base URL for the SDP document becomes: "rtsp://exam-
   ple.com/container.mp4/", i.e. an extra trailing slash.  When using
   the URL resolution rules in RFC 2396 that will resolve correctly.
   However as a warning if the session level control URL is a * that
   control URL will be equal to "rtsp://example.com/container.mp4/" and
   include the slash.

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 synchro-
   nization. If the session is unicast, the port number serves as a rec-
   ommendation from the server to the client; the client still has to
   include it in its SETUP request and may ignore this recommendation.
   If the server has no preference, it SHOULD set the port number value
   to zero.

   Example:

     m=audio 0 RTP/AVP 31



C.1.3 Payload Type(s)

   The payload type(s) are specified in the "m=" field. In case the pay-
   load type is a static payload type from RFC 1890 [1], no other infor-
   mation is required. In case it is a dynamic payload type, the media
   attribute "rtpmap" is used to specify what the media is. The "encod-
   ing name" within the "rtpmap" attribute may be one of those specified
   in RFC 1890 (Sections 5 and 6), or an MIME type registered with IANA,
   or an experimental encoding with a "X-" prefix as specified in SDP
   (RFC 2327 [24]). Codec-specific parameters are not specified in this
   field, but rather in the "fmtp" attribute described below. Implemen-
   tors seeking to register new encodings should follow the procedure in
   RFC 1890 [1]. If the media type is not suited to the RTP AV profile,
   then it is recommended that a new profile be created and the appro-
   priate profile name be used in lieu of "RTP/AVP" in the "m=" field.

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 the packetization
   interval is conveyed using the "ptime" attribute.

C.1.5 Range of Presentation




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   The "a=range" attribute defines the total time range of the stored
   session. (The length of live sessions can be deduced from the "t" and
   "r" parameters.) Unless the presentation contains media streams of
   different durations, the length attribute is a session-level
   attribute. In case of different length the range attribute MUST be
   given at media level for all media. The unit is specified first, fol-
   lowed by the value range. The units and their values are as defined
   in Section 3.4, 3.5 and 3.6. '

   Examples:

     a=range:npt=0-34.4368
     a=range:clock=19971113T2115-19971113T2203



C.1.6 Time of Availability

   The "t=" field MUST contain suitable values for the start and stop
   times for both aggregate and non-aggregate stream control. With
   aggregate 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. With non-aggregate control, the
   values should reflect the actual period for which the session is
   available in keeping with SDP semantics, and not depend on other
   means (such as the life of the web page containing the description)
   for this purpose.

C.1.7 Connection Information

   In SDP, the "c=" field contains the destination address for the media
   stream. However, for on-demand unicast streams and some multicast
   streams, the destination address is specified by the client via the
   SETUP request. Unless the media content has a fixed destination
   address, the "c=" field is to be set to a suitable null value. For
   addresses of type "IP4", this value is "0.0.0.0".

C.1.8 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 13.22) to only
   allow session establishment if this attribute value still corresponds
   to that of the current description.  The attribute value is opaque
   and may contain any character allowed within SDP attribute values.




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   Example:

     a=etag:158bb3e7c7fd62ce67f12b533f06b83a




     One could argue that the "o=" field provides identical func-
     tionality. However, it does so in a manner that would put con-
     straints 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 and multiple
   media sections are specified, each section MUST have the control URL
   specified via the "a=control:" attribute.

   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   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 URL 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 ini-
   tialization 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 media stream
   information via DESCRIBE.

C.3 Aggregate Control Available

   In this scenario, the server has multiple streams that can be con-
   trolled as a whole. In this case, there are both a media-level



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   "a=control:" attributes, which are used to specify the stream URLs,
   and a session-level "a=control:" attribute which is used as the
   request URL for aggregate control. If the media-level URL is rela-
   tive, it is resolved to absolute URLs according to Section C.1.1
   above.

   If the presentation comprises only a single stream, the media-level
   "a=control:" attribute may be omitted altogether. However, if the
   presentation contains more than one stream, each media stream section
   MUST contain its own "a=control" attribute.

   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I contain
   i=<more info>
   e=adm@example.com
   c=IN IP4 0.0.0.0
   t=0 0
   a=control:rtsp://example.com/movie/
   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 URLs rtsp://exam-
   ple.com/movie/trackID=1 and rtsp://example.com/movie/trackID=2 to set
   up the video and audio streams, respectively. The URL rtsp://exam-
   ple.com/movie/ controls the whole movie.

   A client is not required to issues SETUP requests for all streams
   within an aggregate object. Servers SHOULD allow the client to ask
   for only a subset of the streams.

D Minimal RTSP implementation

D.1 Client

   A client implementation MUST be able to do the following :            |

     + Generate the following requests: SETUP, TEARDOWN, PLAY.           |

     + Include the following headers in requests: CSeq, Connection, Ses- |
       sion, Transport.                                                  |



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     + Parse and understand the following headers in responses: CSeq,    |
       Connection, Session, Transport, Content-Language, Content-Encod-  |
       ing, Content-Length, Content-Type.                                |

     + Understand the class of each error code received and notify the   |
       end-user, if one is present, of error codes in classes 4xx and    |
       5xx. The notification requirement may be relaxed if the end-user  |
       explicitly does not want it for one or all status codes.          |

     + Expect and respond to asynchronous requests from the server, such |
       as REDIRECT. This does not necessarily mean that it should imple- |
       ment the REDIRECT method, merely that it MUST respond positively  |
       or negatively to any request received from the server.            |


   Though not required, the following are RECOMMENDED.

     + Implement RTP/AVP/UDP as a valid transport.

     + Inclusion of the User-Agent header.

     + Understand SDP session descriptions as defined in Appendix C

     + Accept media initialization formats (such as SDP) from standard
       input, command line, or other means appropriate to the operating
       environment to act as a "helper application" for other applica-
       tions (such as web browsers).


     There may be RTSP applications different from those initially
     envisioned by the contributors to the RTSP specification for
     which the requirements above do not make sense. Therefore, the
     recommendations above serve only as guidelines instead of
     strict requirements.

D.1.1 Basic Playback

   To support on-demand playback of media streams, the client MUST addi-
   tionally be able to do the following:

     + generate the PAUSE request;

     + implement the REDIRECT method, and the Location header.

D.1.2 Authentication-enabled

   In order to access media presentations from RTSP servers that require
   authentication, the client MUST additionally be able to do the



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   following:

     + recognize the 401 (Unauthorized) status code;

     + parse and include the WWW-Authenticate header;

     + implement Basic Authentication and Digest Authentication.

D.2 Server

   A minimal server implementation MUST be able to do the following:     |

     + Implement the following methods: SETUP, TEARDOWN, OPTIONS and     |
       PLAY.                                                             |

     + Include the following headers in responses: Connection, Content-  |
       Length, Content-Type, Content-Language, Content-Encoding, Times-  |
       tamp, Transport, Public, and Via, and Unsupported.  RTP-compliant |
       implementations MUST also implement the RTP-Info field.           |

     + Parse and respond appropriately to the following headers in       |
       requests: Connection, Proxy-Require, Session, Transport, and      |
       Require.                                                          |

   Though not required, the following are highly recommended at the time
   of publication for practical interoperability with initial implemen-
   tations and/or to be a "good citizen".

     + Implement RTP/AVP/UDP as a valid transport.

     + Inclusion of the Server header.

     + Implement the DESCRIBE method.

     + Generate SDP session descriptions as defined in Appendix C


     There may be RTSP applications different from those initially
     envisioned by the contributors to the RTSP specification for
     which the requirements above do not make sense. Therefore, the
     recommendations above serve only as guidelines instead of
     strict requirements.

D.2.1 Basic Playback

   To support on-demand playback of media streams, the server MUST addi-
   tionally be able to do the following:




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     + Recognize the Range header, and return an error if seeking is not
       supported.

     + Implement the PAUSE method.

   In addition, in order to support commonly-accepted user interface
   features, the following are highly recommended for on-demand media
   servers:

     + Include and parse the Range header, with NPT units. Implementa-
       tion of SMPTE units is recommended.

     + Include the length of the media presentation in the media ini-
       tialization information.

     + Include mappings from data-specific timestamps to NPT. When RTP
       is used, the rtptime portion of the RTP-Info field may be used to
       map RTP timestamps to NPT.


     Client implementations may use the presence of length informa-
     tion to determine if the clip is seekable, and visably disable
     seeking features for clips for which the length information is
     unavailable. A common use of the presentation length is to
     implement a "slider bar" which serves as both a progress indi-
     cator and a timeline positioning tool.

   Mappings from RTP timestamps to NPT are necessary to ensure correct
   positioning of the slider bar.

D.2.2 Authentication-enabled

   In order to correctly handle client authentication, the server MUST
   additionally be able to do the following:

     + Generate the 401 (Unauthorized) status code when authentication
       is required for the resource.

     + Parse and include the WWW-Authenticate header

     + Implement Basic Authentication and Digest Authentication

E Open Issues

     1.   Should we add the header Accept-Ranges as proposed in this
          specification?





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     2.   Upon receiving a response on a REDIRECT request can the server
          close the session or should it wait for a TEARDOWN request
          from the client?

     3.   The proxy indications in the two header tables in chapter 13
          needs review.

     4.   Should the Allow header be possible to use optional in request
          or responses besides the now specified 405 error code?

     5.   What text should be written on use of authorization in this
          spec?

     6.   How does entity tags relate to the If-Match header? The usage
          in SDP must also be clarified related to syntax, etc.

     7.   Should the Last-Modified header be required on other level
          than optional?

     8.   How to handle range headers for negative scale playback.

     9.   The minimal implementation must be looked over to see if it
          complies with the specification. All must and should shall be
          included in the minimal. Feature-tags for these needs to be
          defined. Further feature-tags needs to be discussed.

     10.  The list specifying which status codes are allowed on which
          request methods seem to be in error and need review.

F Changes

   Compared to RFC 2326, the following issues are addressed:

     + http://rtsp.org/bug448521 - "URLs in Rtp-Info need to be quoted".
       URLs in RTP-info header now MAY be quoted if needed.

     + http://rtsp.org/bug448525 - Syntax for SSRC should be clarified.
       Require 8*8 HEX and corresponding text added.

     + http://rtsp.org/bug461083 - "Body w/o Content-Length clarifica-
       tion". This is clarified and any message with a message body is
       required to have a Content-Length header.

     + http://rtsp.org/bug477407 - Transport BNF doesn't properly deal
       with semicolon and comma

     + http://rtsp.org/bug477413 - Transport BNF: mode parameter issues




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     + http://rtsp.org/bug477416 - "BNF error section 3.6 NPT", Added an
       optional [NPT] definition. Fixed so that the same possibilities
       exist for all time formats.

     + http://rtsp.org/bug477421 - "When to send response". A clarifying
       note in the status code chapter that when sending 400 responses,
       the server MUST NOT add cseq if missing.

     + http://rtsp.org/bug507347 - Removal of destination redirection in
       the transport header.

     + http://rtsp.org/bug477404 - "Errors in table in chapter 12".  The
       table has been updated using the SIP structure. However the table
       become to big to fit in a single page and has been split.

     + http://rtsp.org/bug477419 - Updating HTTP references to rfc2616
       by adding public, and content-base header. Section references in
       header chapter updated. Known effects on RTSP due to HTTP clari-
       fications:

       - Content-Encoding header can include encoding of type "iden-
         tity".

     + http://rtsp.org/bug500803 - Rewritten the complete chapter on the
       state machine.

     + http://rtsp.org/bug513753 - Created a IANA section defining four
       registries.

     + http://rtsp.org/bug477427 - A new subsection in the connections
       chapter clarifying how the server and client may handle transport
       connections. Includes defining a feature-tag.

     + - Accept-Ranges response header is added. This header clarifies
       which range formats that can be used for a resource.

     + - Added Headers Timestamp, Via, Unsupported as required for a
       minimal server implementation.

     + http://rtsp.org/bug477425 - "Inconsistency between timeformats".
       Fixed so that all formats has the same capabilities as NPT.

     + http://rtsp.org/bug499573 - "Incorrect grammar on Server header".
       Added corrected BNF for User-Agent and Server header as a comple-
       ment to the reference.

     + The definition in the introduction of the RTSP session has been
       changed.



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     + Updated RTSP URL's and source and destination parameters in the
       transport header to handle IPv6 addresses.

     + All BNF definitions are updated according to the rules defined in
       RFC 2234 [14].

     + The use of status code 303 "See Other" has been decapitated as it
       does not make sense to use in RTSP.

     + Added status code 350, 351 and updated usage of the other redi-
       rect status codes, see chapter  12.3.

     + Removed Queued play (http://rtsp.org/bug508211) and decapitated
       use of PLAY for keep-alive while in playing state.

     + Explicitly wrote out the possibilities to use multiple ranges to
       allow for editing.

     + Text specifying the special behavior of PLAY for live content.

     + When sending response 451 and 458 the response body should con-
       tain the offending parameters.

     + 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.

     + Added wording on the usage of Connection:Close for RTSP.

     + Put requirement on CSeq header that the value is increased by one
       for each new RTSP request.

     + Added requirement that the Date header must be used for all mes-
       sages with entity. Also the Server should always include it.

     + Removed possibility to use Range header combined with Scale
       header to indicate when it shall be activated, due to that it
       can't work as defined. Also added rule that lack of scale header
       in response indicate lack of support. Feature-tags for scaled
       playback 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 fea-
       ture negotiation in RTSP.




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     + Clarified that the timestamp header can be used to resolve
       retransmission ambiguities.

     + Added two transport header parameters to be used to signal RTCP
       port for server and client when not assigned in pairs. Shall be
       used for NAT traversal with mechanisms like STUN. The interoper-
       ability issue is solved by requiring a client to know that a
       server supports this specification.

     + Defined a IANA registries for the transport headers parameters,
       transport-protocol, profile, lower-transport, and mode.

     + The OPTIONS method has been clarified on how to use the Public
       and Allow headers.

     + The Session header text has been expanded with a explanation on
       keep alive and which methods to use.

     + http://rtsp.org/bug503949 - Range header format for PAUSE is
       unclear. This has been resolved by requiring a ranged pause to
       only contain a single value as a beginning of an open range.

     + Servers may optional implement SETUP and TEARDOWN of a single
       media while in PLAY state. This is signaled using an feature-tag
       (play.setup).

     + The transport headers interleave parameter's text was made more
       strict and use formal requirements levels. However no change on
       how it is used was made.

     + Added a fragment part to the RTSP URL. This seem to be indicated
       by the note below the definition however it was not part of the
       BNF.

     + The RECORD and ANNOUNCE methods are removed as they are lacking
       implementation and not considered necessary in the core specifi-
       cation. Any work on these methods should be done as a extension
       document to RTSP.

     + The description on how rtspu and rtsps is not part of the core
       specification and will require external description.

     + The Transport headers RTP port parameters has been updated to
       support non-continuous port numbers. Also a possibility for the
       client to specify SSRC has been added.

     + Clarified that RTP-Info URLs that are relative uses the request
       URL as base URL. Also clarified that the URL that must be used is



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       the SETUP.

     + Included two new general address parameters "src_addresses" and
       "dst_addresses" to be used to give address source and destination
       of media traffic.

     + Updated the text on the transport headers "destination" parameter
       regarding what security precautions the server shall perform.

     + Wrote a new chapter 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 chapter also includes
       guidelines what to think of when writing usage guidelines for new
       protocols and profiles.

     + The embedded (interleaved) binary data and its transport parame-
       ter was clarified to being symmetric and that it is the server
       that sets the channel numbers.

     + Added a new chapter describing the available mechanisms to deter-
       mine if functionality is supported, called "Capability Handling".
       Renamed option-tags to feature-tags.

     + Added a contributors chapter with people who has contribute
       actual text to the specification.

     + Added text that requires the Range to always be present in PLAY
       responses. Clarified what should be sent in case of live streams.

   Note that this list does not reflect minor changes in wording or cor-
   rection of typographical errors.

   A word-by-word diff from RFC 2326 can be found at
   http://rtsp.org/2002/drafts

G 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



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   USA
   electronic mail: anrao@cisco.com

   Robert Lanphier
   RealNetworks
   P.O. Box 91123
   Seattle, WA 98111-9223
   USA
   electronic mail: robla@real.com

   Magnus Westerlund
   Ericsson AB, ERA/TVA/A
   Torshamsgatan 23
   SE-164 80 STOCKHOLM
   SWEDEN
   electronic mail: magnus.westerlund@ericsson.com

H Contributors

   The following people has made written contribution included in the    |
   specification:                                                        |

     + Tom Marshall has contributed with text about the usage of 3rr     |
       status codes.                                                     |

     + Thomas Zheng has contributed with text regarding the usage of the |
       Range in PLAY responses.                                          |

     + Aravind Narasimhan has contributed with updated text regarding    |
       the allowed usage of destination.                                 |


I 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 men-
   tioned, the following individuals have contributed to this specifica-
   tion:

   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,



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   Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
   Marshall, Rob McCool, Aravind Narasimhan, 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, and David Walker.

   [1] H. Schulzrinne, "RTP profile for audio and video conferences with
   minimal control," RFC 1890, Internet Engineering Task Force, Jan.
   1996.

   [2] R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. Berners-Lee,
   "Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet Engi-
   neering Task Force, Jan. 1997.

   [3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst, "Internationaliza-
   tion of the hypertext markup language," RFC 2070, Internet Engineer-
   ing Task Force, Jan.  1997.

   [4] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.

   [5] ISO/IEC, "Information technology -- generic coding of moving pic-
   tures and associated audio informaiton -- part 6: extension for digi-
   tal storage media and control," Draft International Standard ISO
   13818-6, International Organization for Standardization ISO/IEC
   JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.

   [6] J. Franks, P. Hallam-Baker, and J. Hostetler, "An extension to
   HTTP: digest access authentication," RFC 2069, Internet Engineering
   Task Force, Jan.  1997.

   [7] J. Postel, "User datagram protocol," RFC STD 6, 768, Internet
   Engineering Task Force, Aug. 1980.

   [8] B. Hinden and C. Partridge, "Version 2 of the reliable data pro-
   tocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr. 1990.

   [9] J. Postel, "Transmission control protocol," RFC STD 7, 793,
   Internet Engineering Task Force, Sept. 1981.

   [10] 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.

   [11] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
   RFC 1961, Internet Engineering Task Force, June 1996.



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   [12] J. Miller, P. Resnick, and D. Singer, "Rating services and rat-
   ing systems (and their machine readable descriptions)," Recommenda-
   tion REC-PICS-services-961031, W3C (World Wide Web Consortium),
   Boston, Massachusetts, Oct. 1996.

   [13] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
   distribution label syntax and communication protocols," Recommenda-
   tion REC-PICS-labels-961031, W3C (World Wide Web Consortium), Boston,
   Massachusetts, Oct. 1996.

   [14] D. Crocker and P. Overell, "Augmented BNF for syntax specifica-
   tions: ABNF," RFC 2234, Internet Engineering Task Force, Nov. 1997.

   [15] B. Braden, "Requirements for internet hosts - application and
   support," RFC STD 3, 1123, Internet Engineering Task Force, Oct.
   1989.

   [16] R. Elz, "A compact representation of IPv6 addresses," RFC 1924,
   Internet Engineering Task Force, Apr. 1996.

   [17] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
   locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
   1994.

   [18] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
   2279, Internet Engineering Task Force, Jan. 1998.

   [19] B. Braden, "T/TCP -- TCP extensions for transactions functional
   specification," RFC 1644, Internet Engineering Task Force, July 1994.

   [20] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
   Reading, Massachusetts: Addison-Wesley, 1994.

   [21] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
   protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
   1998.

   [22] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
   identifiers (URI): generic syntax," RFC 2396, Internet Engineering
   Task Force, Aug. 1998.

   [23] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
   a transport protocol for real-time applications," RFC 1889, Internet
   Engineering Task Force, Jan. 1996.

   [24] M. Handley and V. Jacobson, "SDP: session description protocol,"
   RFC 2327, Internet Engineering Task Force, Apr. 1998.




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   [25] R. Fielding, "Relative uniform resource locators," RFC 1808,
   Internet Engineering Task Force, June 1995.

   [26] R. Fielding, "Hypertext Transfer Protocol -- HTTP/1.1," RFC
   2616, Internet Engineering Task Force, June 1999.

   [27] T. Dierks, C. Allen, "The TLS Protocol, Version 1.0," RFC 2246,
   Internet Engineering Task Force, Januari 1999.

   [28] International Telecommunication Union, "Visual telephone systems
   and equipment for local area networks which provide a non-guaranteed
   quality of service," Recommendation H.323, Telecommunications Stan-
   darization Sector of ITU, Geneva, Switzerland, May 1996.

   [29] T. Narten, H. Alvestrand, "Guidelines for Writing an IANA Con-
   siderations Section in RFCs," RFC2434, Internet Engineering Task
   Force, October 1998.

   [30] R. Hinden, B. Carpenter, L. Masinter, "Format for Literal IPv6
   Addresses in URL's," RFC 2732, Internet Engineering Task Force,
   December 1999.

   [31] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy, "STUN - Simple
   Traversal of UDP Through Network Address Translators," Internet Engi-
   neering Task Force, Work in Progress, October 2002.

   [32] P. Srisuresh, K. Egevang, "Traditional IP Network Address Trans-
   lator (Traditional NAT)," RFC 3022, Internet Engineering Task Force,
   January 2001.

   [33] M. Westerlund, "How to make Real-Time Streaming Protocol (RTSP)
   traverse Network Address Translators (NAT) and interact with Fire-
   walls.", Internet Engineering Task Force Draft, draft-ietf-mmusic-
   rtsp-nat-00.txt, Work in Progress, Feb 2003.

   [34] A. Narasimhan, A. Narasimhan, "MUTE and UNMUTE extension to
   RTSP", Internet Engineering Task Force Draft, draft-sergent-rtsp-
   mute-00.txt, Work in Progress, Feb 2002.

   [35] Third Generation Partnership Project (3GPP), "Transparent end-
   to-end Packet-switched Streaming Service (PSS); Protocols and codecs"
   3GPP Technical Specification 26.234, Release 5.

   [36] D. Yon, "Connection-Oriented Media Transport in SDP", Internet
   Engineering Task Force Draft, draft-ietf-mmusic-sdp-comedia-04.txt,
   July 2002.





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   [37] John Lazzaro, "Framing RTP and RTCP Packets over Connection-Ori-
   ented Transport", Internet Engineering Task Force Draft , draft-laz-
   zaro-avt-rtp-framing-contrans-00.txt, January 2003.


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   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
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H. Schulzrinne et. al.                                      [Page 133]


                           Table of Contents


1          Introduction  . . . . . . . . . . . . . . . . . . . . . .   3
1.1        The Update of the RTSP Specification  . . . . . . . . . .   3
1.2        Purpose . . . . . . . . . . . . . . . . . . . . . . . . .   4
1.3        Requirements  . . . . . . . . . . . . . . . . . . . . . .   6
1.4        Terminology . . . . . . . . . . . . . . . . . . . . . . .   6
1.5        Protocol Properties . . . . . . . . . . . . . . . . . . .   8
1.6        Extending RTSP  . . . . . . . . . . . . . . . . . . . . .  10
1.7        Overall Operation . . . . . . . . . . . . . . . . . . . .  11
1.8        RTSP States . . . . . . . . . . . . . . . . . . . . . . .  12
1.9        Relationship with Other Protocols . . . . . . . . . . . .  12
2          Notational Conventions  . . . . . . . . . . . . . . . . .  13
3          Protocol Parameters . . . . . . . . . . . . . . . . . . .  13
3.1        RTSP Version  . . . . . . . . . . . . . . . . . . . . . .  13
3.2        RTSP URL  . . . . . . . . . . . . . . . . . . . . . . . .  14
3.3        Session Identifiers . . . . . . . . . . . . . . . . . . .  15
3.4        SMPTE Relative Timestamps . . . . . . . . . . . . . . . .  15
3.5        Normal Play Time  . . . . . . . . . . . . . . . . . . . .  16
3.6        Absolute Time . . . . . . . . . . . . . . . . . . . . . .  17
3.7        Feature-tags  . . . . . . . . . . . . . . . . . . . . . .  18
4          RTSP Message  . . . . . . . . . . . . . . . . . . . . . .  18
4.1        Message Types . . . . . . . . . . . . . . . . . . . . . .  19
4.2        Message Headers . . . . . . . . . . . . . . . . . . . . .  19
4.3        Message Body  . . . . . . . . . . . . . . . . . . . . . .  19
4.4        Message Length  . . . . . . . . . . . . . . . . . . . . .  19
5          General Header Fields . . . . . . . . . . . . . . . . . .  20
6          Request . . . . . . . . . . . . . . . . . . . . . . . . .  20
6.1        Request Line  . . . . . . . . . . . . . . . . . . . . . .  20
6.2        Request Header Fields . . . . . . . . . . . . . . . . . .  21
7          Response  . . . . . . . . . . . . . . . . . . . . . . . .  22
7.1        Status-Line . . . . . . . . . . . . . . . . . . . . . . .  22
7.1.1      Status Code and Reason Phrase . . . . . . . . . . . . . .  22
7.1.2      Response Header Fields  . . . . . . . . . . . . . . . . .  25
8          Entity  . . . . . . . . . . . . . . . . . . . . . . . . .  25
8.1        Entity Header Fields  . . . . . . . . . . . . . . . . . .  27
8.2        Entity Body . . . . . . . . . . . . . . . . . . . . . . .  27
9          Connections . . . . . . . . . . . . . . . . . . . . . . .  27
9.1        Pipelining  . . . . . . . . . . . . . . . . . . . . . . .  28
9.2        Reliability and Acknowledgements  . . . . . . . . . . . .  28
9.3        The usage of connections  . . . . . . . . . . . . . . . .  29
9.4        Use of IPv6 . . . . . . . . . . . . . . . . . . . . . . .  30
10         Capability Handling . . . . . . . . . . . . . . . . . . .  31
11         Method Definitions  . . . . . . . . . . . . . . . . . . .  32
11.1       OPTIONS . . . . . . . . . . . . . . . . . . . . . . . . .  33
11.2       DESCRIBE  . . . . . . . . . . . . . . . . . . . . . . . .  34
11.3       SETUP . . . . . . . . . . . . . . . . . . . . . . . . . .  35



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11.4       PLAY  . . . . . . . . . . . . . . . . . . . . . . . . . .  37
11.5       PAUSE . . . . . . . . . . . . . . . . . . . . . . . . . .  40
11.6       TEARDOWN  . . . . . . . . . . . . . . . . . . . . . . . .  43
11.7       GET_PARAMETER . . . . . . . . . . . . . . . . . . . . . .  43
11.8       SET_PARAMETER . . . . . . . . . . . . . . . . . . . . . .  44
11.9       REDIRECT  . . . . . . . . . . . . . . . . . . . . . . . .  45
11.10      PING  . . . . . . . . . . . . . . . . . . . . . . . . . .  47
11.11      Embedded (Interleaved) Binary Data  . . . . . . . . . . .  47
12         Status Code Definitions . . . . . . . . . . . . . . . . .  49
12.1       Success 1xx . . . . . . . . . . . . . . . . . . . . . . .  49
12.1.1     100 Continue  . . . . . . . . . . . . . . . . . . . . . .  49
12.2       Success 2xx . . . . . . . . . . . . . . . . . . . . . . .  49
12.2.1     250 Low on Storage Space  . . . . . . . . . . . . . . . .  49
12.3       Redirection 3xx . . . . . . . . . . . . . . . . . . . . .  49
12.3.1     TBW . . . . . . . . . . . . . . . . . . . . . . . . . . .  50
12.3.2     301 Moved Permanently . . . . . . . . . . . . . . . . . .  50
12.3.3     302 Found . . . . . . . . . . . . . . . . . . . . . . . .  50
12.3.4     303 See Other . . . . . . . . . . . . . . . . . . . . . .  50
12.3.5     304 Not Modified  . . . . . . . . . . . . . . . . . . . .  50
12.3.6     305 Use Proxy . . . . . . . . . . . . . . . . . . . . . .  51
12.4       Client Error 4xx  . . . . . . . . . . . . . . . . . . . .  51
12.4.1     400 Bad Request . . . . . . . . . . . . . . . . . . . . .  51
12.4.2     405 Method Not Allowed  . . . . . . . . . . . . . . . . .  51
12.4.3     451 Parameter Not Understood  . . . . . . . . . . . . . .  51
12.4.4     452 reserved  . . . . . . . . . . . . . . . . . . . . . .  51
12.4.5     453 Not Enough Bandwidth  . . . . . . . . . . . . . . . .  51
12.4.6     454 Session Not Found . . . . . . . . . . . . . . . . . .  51
12.4.7     455 Method Not Valid in This State  . . . . . . . . . . .  52
12.4.8     456 Header Field Not Valid for Resource . . . . . . . . .  52
12.4.9     457 Invalid Range . . . . . . . . . . . . . . . . . . . .  52
12.4.10    458 Parameter Is Read-Only  . . . . . . . . . . . . . . .  52
12.4.11    459 Aggregate Operation Not Allowed . . . . . . . . . . .  52
12.4.12    460 Only Aggregate Operation Allowed  . . . . . . . . . .  52
12.4.13    461 Unsupported Transport . . . . . . . . . . . . . . . .  52
12.4.14    462 Destination Unreachable . . . . . . . . . . . . . . .  52
12.5       Server Error 5xx  . . . . . . . . . . . . . . . . . . . .  53
12.5.1     551 Option not supported  . . . . . . . . . . . . . . . .  53
13         Header Field Definitions  . . . . . . . . . . . . . . . .  53
13.1       Accept  . . . . . . . . . . . . . . . . . . . . . . . . .  55
13.2       Accept-Encoding . . . . . . . . . . . . . . . . . . . . .  55
13.3       Accept-Language . . . . . . . . . . . . . . . . . . . . .  55
13.4       Accept-Ranges . . . . . . . . . . . . . . . . . . . . . .  57
13.5       Allow . . . . . . . . . . . . . . . . . . . . . . . . . .  59
13.6       Authorization . . . . . . . . . . . . . . . . . . . . . .  59
13.7       Bandwidth . . . . . . . . . . . . . . . . . . . . . . . .  59
13.8       Blocksize . . . . . . . . . . . . . . . . . . . . . . . .  60
13.9       Cache-Control . . . . . . . . . . . . . . . . . . . . . .  60
13.10      Connection  . . . . . . . . . . . . . . . . . . . . . . .  63



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13.11      Content-Base  . . . . . . . . . . . . . . . . . . . . . .  63
13.12      Content-Encoding  . . . . . . . . . . . . . . . . . . . .  63
13.13      Content-Language  . . . . . . . . . . . . . . . . . . . .  63
13.14      Content-Length  . . . . . . . . . . . . . . . . . . . . .  64
13.15      Content-Location  . . . . . . . . . . . . . . . . . . . .  64
13.16      Content-Type  . . . . . . . . . . . . . . . . . . . . . .  64
13.17      CSeq  . . . . . . . . . . . . . . . . . . . . . . . . . .  64
13.18      Date  . . . . . . . . . . . . . . . . . . . . . . . . . .  64
13.19      Expires . . . . . . . . . . . . . . . . . . . . . . . . .  65
13.20      From  . . . . . . . . . . . . . . . . . . . . . . . . . .  66
13.21      Host  . . . . . . . . . . . . . . . . . . . . . . . . . .  66
13.22      If-Match  . . . . . . . . . . . . . . . . . . . . . . . .  66
13.23      If-Modified-Since . . . . . . . . . . . . . . . . . . . .  66
13.24      Last-Modified . . . . . . . . . . . . . . . . . . . . . .  66
13.25      Location  . . . . . . . . . . . . . . . . . . . . . . . .  67
13.26      Proxy-Authenticate  . . . . . . . . . . . . . . . . . . .  67
13.27      Proxy-Require . . . . . . . . . . . . . . . . . . . . . .  67
13.28      Public  . . . . . . . . . . . . . . . . . . . . . . . . .  67
13.29      Range . . . . . . . . . . . . . . . . . . . . . . . . . .  68
13.30      Referer . . . . . . . . . . . . . . . . . . . . . . . . .  69
13.31      Retry-After . . . . . . . . . . . . . . . . . . . . . . .  69
13.32      Require . . . . . . . . . . . . . . . . . . . . . . . . .  69
13.33      RTP-Info  . . . . . . . . . . . . . . . . . . . . . . . .  70
13.34      Scale . . . . . . . . . . . . . . . . . . . . . . . . . .  72
13.35      Speed . . . . . . . . . . . . . . . . . . . . . . . . . .  72
13.36      Server  . . . . . . . . . . . . . . . . . . . . . . . . .  73
13.37      Session . . . . . . . . . . . . . . . . . . . . . . . . .  73
13.38      Supported . . . . . . . . . . . . . . . . . . . . . . . .  75
13.39      Timestamp . . . . . . . . . . . . . . . . . . . . . . . .  75
13.40      Transport . . . . . . . . . . . . . . . . . . . . . . . .  76
13.41      Unsupported . . . . . . . . . . . . . . . . . . . . . . .  81
13.42      User-Agent  . . . . . . . . . . . . . . . . . . . . . . .  82
13.43      Vary  . . . . . . . . . . . . . . . . . . . . . . . . . .  82
13.44      Via . . . . . . . . . . . . . . . . . . . . . . . . . . .  82
13.45      WWW-Authenticate  . . . . . . . . . . . . . . . . . . . .  82
14         Caching . . . . . . . . . . . . . . . . . . . . . . . . .  82
15         Examples  . . . . . . . . . . . . . . . . . . . . . . . .  83
15.1       Media on Demand (Unicast) . . . . . . . . . . . . . . . .  83
15.2       Streaming of a Container file . . . . . . . . . . . . . .  85
15.3       Single Stream Container Files . . . . . . . . . . . . . .  88
15.4       Live Media Presentation Using Multicast . . . . . . . . .  90
16         Syntax  . . . . . . . . . . . . . . . . . . . . . . . . .  91
16.1       Base Syntax . . . . . . . . . . . . . . . . . . . . . . .  91
16.2       RTSP Protocol Definition  . . . . . . . . . . . . . . . .  92
16.2.1     Message Syntax  . . . . . . . . . . . . . . . . . . . . .  92
16.2.2     Header Syntax . . . . . . . . . . . . . . . . . . . . . .  96
17         Security Considerations . . . . . . . . . . . . . . . . .  97
18         IANA Considerations . . . . . . . . . . . . . . . . . . .  99



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18.1       Feature-tags  . . . . . . . . . . . . . . . . . . . . . . 100
18.1.1     Description . . . . . . . . . . . . . . . . . . . . . . . 100
18.1.2     Registering New Feature-tags with IANA  . . . . . . . . . 100
18.1.3     Registered entries  . . . . . . . . . . . . . . . . . . . 100
18.2       RTSP Methods  . . . . . . . . . . . . . . . . . . . . . . 101
18.2.1     Description . . . . . . . . . . . . . . . . . . . . . . . 101
18.2.2     Registering New Methods with IANA . . . . . . . . . . . . 101
18.2.3     Registered Entries  . . . . . . . . . . . . . . . . . . . 101
18.3       RTSP Status Codes . . . . . . . . . . . . . . . . . . . . 101
18.3.1     Description . . . . . . . . . . . . . . . . . . . . . . . 101
18.3.2     Registering New Status Codes with IANA  . . . . . . . . . 101
18.3.3     Registered Entries  . . . . . . . . . . . . . . . . . . . 102
18.4       RTSP Headers  . . . . . . . . . . . . . . . . . . . . . . 102
18.4.1     Description . . . . . . . . . . . . . . . . . . . . . . . 102
18.4.2     Registering New Headers with IANA . . . . . . . . . . . . 102
18.4.3     Registered entries  . . . . . . . . . . . . . . . . . . . 102
18.5       Transport Header registries . . . . . . . . . . . . . . . 103
18.5.1     Transport Protocols . . . . . . . . . . . . . . . . . . . 103
18.5.2     Profile . . . . . . . . . . . . . . . . . . . . . . . . . 103
18.5.3     Lower Transport . . . . . . . . . . . . . . . . . . . . . 104
18.5.4     Transport modes . . . . . . . . . . . . . . . . . . . . . 104
18.6       Cache Directive Extensions  . . . . . . . . . . . . . . . 105
A          RTSP Protocol State Machine . . . . . . . . . . . . . . . 105
A.1        States  . . . . . . . . . . . . . . . . . . . . . . . . . 105
A.2        State variables . . . . . . . . . . . . . . . . . . . . . 106
A.3        Abbreviations . . . . . . . . . . . . . . . . . . . . . . 106
A.4        State Tables  . . . . . . . . . . . . . . . . . . . . . . 106
B          Media Transport Alternatives  . . . . . . . . . . . . . . 110
B.1        RTP . . . . . . . . . . . . . . . . . . . . . . . . . . . 110
B.1.1      AVP . . . . . . . . . . . . . . . . . . . . . . . . . . . 111
B.1.2      AVP/UDP . . . . . . . . . . . . . . . . . . . . . . . . . 112
B.1.3      AVP/TCP . . . . . . . . . . . . . . . . . . . . . . . . . 114
B.2        Future Additions  . . . . . . . . . . . . . . . . . . . . 114
C          Use of SDP for RTSP Session Descriptions  . . . . . . . . 114
C.1        Definitions . . . . . . . . . . . . . . . . . . . . . . . 115
C.1.1      Control URL . . . . . . . . . . . . . . . . . . . . . . . 115
C.1.2      Media Streams . . . . . . . . . . . . . . . . . . . . . . 116
C.1.3      Payload Type(s) . . . . . . . . . . . . . . . . . . . . . 116
C.1.4      Format-Specific Parameters  . . . . . . . . . . . . . . . 116
C.1.5      Range of Presentation . . . . . . . . . . . . . . . . . . 116
C.1.6      Time of Availability  . . . . . . . . . . . . . . . . . . 117
C.1.7      Connection Information  . . . . . . . . . . . . . . . . . 117
C.1.8      Entity Tag  . . . . . . . . . . . . . . . . . . . . . . . 117
C.2        Aggregate Control Not Available . . . . . . . . . . . . . 118
C.3        Aggregate Control Available . . . . . . . . . . . . . . . 118
D          Minimal RTSP implementation . . . . . . . . . . . . . . . 119
D.1        Client  . . . . . . . . . . . . . . . . . . . . . . . . . 119
D.1.1      Basic Playback  . . . . . . . . . . . . . . . . . . . . . 120



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Internet Draft                    RTSP                     March 3, 2003


D.1.2      Authentication-enabled  . . . . . . . . . . . . . . . . . 120
D.2        Server  . . . . . . . . . . . . . . . . . . . . . . . . . 121
D.2.1      Basic Playback  . . . . . . . . . . . . . . . . . . . . . 121
D.2.2      Authentication-enabled  . . . . . . . . . . . . . . . . . 122
E          Open Issues . . . . . . . . . . . . . . . . . . . . . . . 122
F          Changes . . . . . . . . . . . . . . . . . . . . . . . . . 123
G          Author Addresses  . . . . . . . . . . . . . . . . . . . . 127
H          Contributors  . . . . . . . . . . . . . . . . . . . . . . 128
I          Acknowledgements  . . . . . . . . . . . . . . . . . . . . 128










































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