[Docs] [txt|pdf] [Tracker] [WG] [Email] [Diff1] [Diff2] [Nits] [IPR]

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
                                                           A. Narasimhan
                                                                     Sun
draft-ietf-mmusic-rfc2326bis-05.txt
October 27, 2003
Expires: April, 2004

                  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
   material or to cite them other than as "work in progress".

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

   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
   Proposed Standard.

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over the delivery of data with real-time
   properties. RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video. Sources of data can include both live data feeds and stored
   clips. This protocol is intended to control multiple data delivery
   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and provide a means for choosing delivery
   mechanisms based upon RTP (RFC 3550).

H. Schulzrinne et. al.                                        [Page 1]


Internet Draft                    RTSP                  October 27, 2003


1 Introduction

1.1 The Update of the RTSP Specification

   This is the draft to an update of RTSP which is currently a proposed
   standard defined in RFC 2326  [21]. Many flaws have been found in
   RTSP since it was published. While this draft tries to address the
   flaws, not all known issues have been resolved.

   The goal of the current work on RTSP is to progress it to draft
   standard status. Whether this is possible without first publishing
   RTSP as a proposed standard depends on the changes necessary to make
   the protocol work. The list of changes in chapter F indicates the
   issues that have already been addressed. The currently open issues
   are listed in chapter E.

   There is also a list of reported bugs available at
   "http://rtspspec.sourceforge.net". These bugs should be taken into
   account when reading this specification. While a lot of these bugs
   are addressed, not all are yet accounted for in this specification.
   Input on the unresolved bugs and other issues can be sent via e-mail
   to the MMUSIC WG's mailing list mmusic@ietf.org and the authors.

   Take special notice of the following:

        o The example section  15 has not yet been revised since the
          changes to protocol have not been completed.

        o The BNF chapter  16 has not been compiled completely.

        o Not all of the contents of RFC 2326 are part of this draft.
          In an attempt to prevent the draft from exploding in size, the
          specification has been reduced and split. The content of this
          draft is the core specification of the protocol. It contains
          the general idea behind RTSP and the basic functionality
          necessary to establish an on-demand play-back session. It also
          contains the mechanisms for extending the protocol. Any other
          functionality will be published as extension documents. Two
          proposals exist at this time:

        o NAT and FW traversal mechanisms for RTSP are described in a
          document called "How to make Real-Time Streaming Protocol
          (RTSP) traverse Network Address Translators (NAT) and interact
          with Firewalls." [33].

        o The MUTE extension [34] contains a proposal on adding
          functionality to mute and unmute media streams in an
          aggregated media session without affecting the time-line of



H. Schulzrinne et. al.                                        [Page 2]


Internet Draft                    RTSP                  October 27, 2003


          the playback.

   There have also been discussions about the following extensions to
   RTSP:

        o Transport security for RTSP messages (rtsps).

        o Unreliable transport of RTSP messages (rtspu).

        o The Record functionality.

        o 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.

        o A RTSP MIB.

   However, so far, they have not become concrete proposals.

1.2 Purpose

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

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

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


        The mechanisms of RTSP's operation over UDP were left out
        of this spec. because they were poorly defined in RFC 2336
        [21] and the tradeoff in size and complexity of this spec.
        for a small gain in a targeted problem space was not deemed
        justifiable.

   The set of streams to be controlled is defined by a presentation



H. Schulzrinne et. al.                                        [Page 3]


Internet Draft                    RTSP                  October 27, 2003


   description. This memorandum does not define a format for the
   presentation description. The streams controlled by RTSP may use RTP
   [1] for their data transport, but the operation of RTSP does not
   depend on the transport mechanism used to carry continuous media. 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:

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

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

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

        o Both a RTSP server and client can issue requests.

        o Data is usually carried out-of-band by a different protocol.
          Session descriptions returned in a DESCRIBE response (see
          Section 11.2) and interleaving of RTP with RTSP over TCP are
          exceptions to this rule (see Section 11.11).

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

        o 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
             presentation 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



H. Schulzrinne et. al.                                        [Page 4]


Internet Draft                    RTSP                  October 27, 2003


             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.

   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 concept of controlling multiple streams
             using a single timeline, generally maintained by the
             server. A client, for example, uses aggregate control when
             it issues a single play or pause message to simultaneously
             control both the audio and video in a movie.

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

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

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

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

        Container file: A file which may contain multiple media streams
             which often 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



H. Schulzrinne et. al.                                        [Page 5]


Internet Draft                    RTSP                  October 27, 2003


             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 examples of continuous media are audio and
             motion video. Continuous 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 meta-information in the
             form of entity-header fields and content in the form of an
             entity-body, as described in Section 8.

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

        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
             initialization phase of stream setup.

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

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

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

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



H. Schulzrinne et. al.                                        [Page 6]


Internet Draft                    RTSP                  October 27, 2003


        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
             presentation, such as the set of encodings, network
             addresses and information about the content. Other IETF
             protocols such as SDP (RFC 2327 [24]) use the term
             "session" for a live presentation. The presentation
             description may take several different formats, including
             but not limited to the session description format SDP.

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

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

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

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




H. Schulzrinne et. al.                                        [Page 7]


Internet Draft                    RTSP                  October 27, 2003


1.5 Protocol Properties

   RTSP has the following properties:

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

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

        Secure: RTSP re-uses web security mechanisms, either at the
             transport level (TLS, RFC 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 does not preclude the use of an
             unreliable datagram protocol (UDP) (RFC 768 [7]), a
             reliable datagram protocol (RDP, RFC 1151, not widely used
             [8]) or a reliable stream protocol such as TCP (RFC 793
             [9]) as it implements application-level reliability. The
             use of a connectionless datagram protocol such as UDP or
             RDP requires additional definition that may be provided as
             extensions to the core RTSP specification.

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

        Separation of stream control and conference initiation: Stream
             control 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
             presentation 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



H. Schulzrinne et. al.                                        [Page 8]


Internet Draft                    RTSP                  October 27, 2003


             [11]) firewalls. 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 continuous 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 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 possible. The protocol would not address how several
        clients negotiate 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:

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

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



H. Schulzrinne et. al.                                        [Page 9]


Internet Draft                    RTSP                  October 27, 2003


   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
   magnitude of changes supported:

        o 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).

        o 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 methods it supports using the Public response
          header.

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

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

1.7 Overall Operation

   Each presentation and media stream may be identified by a RTSP URL.
   The overall presentation and the properties of the media the
   presentation 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
   maintains a common time axis. For simplicity of exposition and



H. Schulzrinne et. al.                                       [Page 10]


Internet Draft                    RTSP                  October 27, 2003


   without loss of generality, it is assumed that the presentation
   description contains exactly one such presentation. A presentation
   may contain several media streams.

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

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

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

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

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

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
   controlled by RTSP requests issued sequentially on different TCP
   connections. Therefore, the server needs to maintain "session state"
   to be able to correlate RTSP requests with a stream. The state
   transitions are described in Appendix A.

   Many methods in RTSP do not contribute to state. However, the



H. Schulzrinne et. al.                                       [Page 11]


Internet Draft                    RTSP                  October 27, 2003


   following play a central role in defining the allocation and usage of
   stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, PING
   and TEARDOWN.

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

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

        PAUSE: Temporarily halts a stream without freeing server
             resources.

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

        PING: Prevents the identified session from being timed out.

        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
   interact with HTTP in that the initial contact with streaming content
   is often to be made through a web page. The current protocol
   specification aims to allow different hand-off points between a web
   server and the media server implementing RTSP. For example, the
   presentation description can be retrieved using HTTP or RTSP, which
   reduces roundtrips in web-browser-based scenarios, yet also allows
   for 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 stateful; they may set parameters
   and continue to control a media stream long after the request has
   been acknowledged.


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



H. Schulzrinne et. al.                                       [Page 12]


Internet Draft                    RTSP                  October 27, 2003


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

2 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
   difference 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
   understanding of why things are the way that they are in RTSP.

3 Protocol Parameters

3.1 RTSP Version

   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-
   specific syntax and semantics for RTSP URLs. The RTSP URL is case
   sensitive.


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




H. Schulzrinne et. al.                                       [Page 13]


Internet Draft                    RTSP                  October 27, 2003


        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
   protocol (within the Internet, TCP), while the scheme rtspu
   identifies an unreliable protocol (within the Internet, UDP). The
   scheme rtsps identifies a reliable transport using secure transport,
   perhaps TLS [27]. The rtspu and rtsps is not defined in this
   specification, and are for future extensions of the protocol to
   define.

   If the port is empty or not given, port 554 SHALL be assumed. The
   semantics 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
   literal IPv6 addresses in RTSP URLs using the host specification in
   RFC 2732 [30].

   A presentation or a stream is identified by a textual media
   identifier, 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
   individual stream within the presentation. Note that some request
   methods can only be applied to streams, not presentations and vice
   versa.

   For example, the RTSP 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



H. Schulzrinne et. al.                                       [Page 14]


Internet Draft                    RTSP                  October 27, 2003


   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
        individual 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
   characters 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
   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 )



H. Schulzrinne et. al.                                       [Page 15]


Internet Draft                    RTSP                  October 27, 2003


   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
   relative 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),
   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



H. Schulzrinne et. al.                                       [Page 16]


Internet Draft                    RTSP                  October 27, 2003


   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
        optimized 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


   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 features in
   RTSP. These tags are used in Require (Section 13.32), Proxy-Require
   (Section 13.27), Unsupported (Section 13.41), and Supported (Section



H. Schulzrinne et. al.                                       [Page 17]


Internet Draft                    RTSP                  October 27, 2003


   13.38) header fields.

   Syntax:

   feature-tag  =  token


   Feature tag needs to indicate if they apply to servers only, proxies   |
   only, or both server and proxies.

   The creator of a new RTSP feature-tag should either prefix the
   feature-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
   themselves 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])

   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.




H. Schulzrinne et. al.                                       [Page 18]


Internet Draft                    RTSP                  October 27, 2003


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
   Content-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
        chunked transfer encoding unnecessary.

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



H. Schulzrinne et. al.                                       [Page 19]


Internet Draft                    RTSP                  October 27, 2003


                   /  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
   version 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
           /  "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



H. Schulzrinne et. al.                                       [Page 20]


Internet Draft                    RTSP                  October 27, 2003


   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.

   One example would be as follows:



     OPTIONS * RTSP/1.0



   An OPTIONS in this form will determine the capabilities of the server
   or the proxy that first receives the request. If one needs to address
   the server explicitly, then one should use an absolute URL with the
   server's address.



H. Schulzrinne et. al.                                       [Page 21]


Internet Draft                    RTSP                  October 27, 2003


     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

   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:



H. Schulzrinne et. al.                                       [Page 22]


Internet Draft                    RTSP                  October 27, 2003


        o 1xx: Informational - Request received, continuing process

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

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

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

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

   The individual values of the numeric status codes defined for
   RTSP/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
                     /  "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



H. Schulzrinne et. al.                                       [Page 23]


Internet Draft                    RTSP                  October 27, 2003


                     /  "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
   x00 status code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if an
   unrecognized status code of 431 is received by the client, it can
   safely assume that there was something wrong with its request and
   treat the response as if it had received a 400 status code. In such
   cases, user agents SHOULD present to the user the entity returned
   with the response, since that entity is likely to include human-
   readable information which will explain the unusual status.

7.1.2 Response Header Fields

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



H. Schulzrinne et. al.                                       [Page 24]


Internet Draft                    RTSP                  October 27, 2003


   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
   combination with a change in the protocol version. However, new or
   experimental header fields MAY be given the semantics of response-
   header fields if all parties in the communication recognize them to
   be response-header fields. Unrecognized header fields are treated as
   entity-header fields.

8 Entity

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

   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



H. Schulzrinne et. al.                                       [Page 25]


Internet Draft                    RTSP                  October 27, 2003



          Code  Reason                            Method
          _______________________________________________________
          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
          503   Service Unavailable               all
          504   Gateway Timeout                   all
          505   RTSP Version Not Supported        all


H. Schulzrinne et. al.                                       [Page 26]


Internet Draft                    RTSP                  October 27, 2003


   Table 1: Status codes and their usage with RTSP methods

                     /  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:

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

        o one connection per request/response transaction;

        o 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
   connections, 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




H. Schulzrinne et. al.                                       [Page 27]


Internet Draft                    RTSP                  October 27, 2003


   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                                      |

   The transmission of RTSP over UDP was optionally to implement and      |
   specified in RFC 2326. However that definition was not satisfactory    |
   for interoperable implementations. Due to lack of interest, this       |
   specification does not specify how RTSP over UDP shall be              |
   implemented. However to maintain backwards compatibility in the        |
   message format certain RTSP headers must be maintained.  These         |
   mechanism are described below. The next section Unreliable Transport   |
   (section  9.3) provides documentation of certain features that are     |
   necessary for transport protocols like UDP.                            |

   Any RTSP request according to this specification SHALL NOT be sent to  |
   a multicast address. Any RTSP request SHALL be acknowledged. 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 cannot typically take advantage of the              |
        application-layer retransmission since the transport stack   |
        will not deliver the application-layer retransmission        |
        before the first attempt has reached the receiver. If the    |
        packet loss is caused by congestion, multiple                |
        retransmissions at different layers will exacerbate the      |
        congestion.                                                  |

   Each request carries a sequence number in the CSeq header (Section     |
   13.17), which MUST be incremented by one for each distinct request     |
   transmitted to the destination end-point.  The initial sequence        |
   number MAY be chosen arbitrary, but is RECOMMENDED to begin with 0.    |
   If a request is repeated because of lack of acknowledgement, the       |
   request MUST carry the original sequence number (i.e., the sequence    |
   number is not incremented).                                            |

9.3 Unreliable Transport                                                  |

   This section provides some information to future specification of      |
   RTSP over unreliable transport.                                        |



H. Schulzrinne et. al.                                       [Page 28]


Internet Draft                    RTSP                  October 27, 2003


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

   The Timestamp header (Section 13.39) is used to avoid the              |
   retransmission ambiguity problem [20] and obviates the need for        |
   Karn's algorithm.                                                      |

   If a request is repeated because of lack of acknowledgement, the       |
   request must carry the original sequence number (i.e., the sequence    |
   number is not incremented).                                            |

   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.        |

   The default port for the RTSP server is 554 for UDP.

9.4 The usage of connections

   Systems implementing RTSP MUST support carrying RTSP over TCP.  The    |
   default port for the RTSP server is 554 for TCP. A number of RTSP      |
   packets destined for the same control end point may be 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 (entity).        |
   Otherwise, an RTSP packet is terminated with an empty line             |
   immediately following the last message header.

   TCP can be used for both persistent connections and for one message
   exchange per connection, as presented above. This section gives
   further 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



H. Schulzrinne et. al.                                       [Page 29]


Internet Draft                    RTSP                  October 27, 2003


   connection 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 second one is to allow for application
   layer mobility. A server and client supporting non-persistent
   connection can survive a loss of a TCP connection, e.g. due to a NAT
   timeout. When the client has discovered that the TCP connection has
   been lost, it can set up a new one when there is need to communicate.

   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
   connection directly after responding to a TEARDOWN request for the
   whole session. A server MUST NOT close the connection as a result of
   responding to a request with an error code. Doing this would prevent
   or result in extra overhead for the client when testing advanced or
   special types of requests.

   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
   connection.

   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
   RECOMMENDED 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
   notifying the client why,due to the lack of signalling channel. An
   example of such a case is when the server desires to send a REDIRECT
   request for a RTSP session to the client.

   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
   determine 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.



H. Schulzrinne et. al.                                       [Page 30]


Internet Draft                    RTSP                  October 27, 2003


9.5 Use of IPv6

   This specification has been updated so that it supports IPv6.
   However this support was not present in RFC 2326 therefore some
   interoperability issues exist. A RFC 2326 implementation can support
   IPv6 as 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 to this version of
   the protocol are basically done in two ways. First, new headers can
   be added. Secondly, new methods can be added. The capability handling
   mechanism is designed to handle 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 header which declares all methods supported for the
   indicated resource.

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

   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
   represents can vary significantly. 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



H. Schulzrinne et. al.                                       [Page 31]


Internet Draft                    RTSP                  October 27, 2003


   desired service. To determine support of a feature-tag several
   different headers can be used, each explained below:

        Supported: The supported header is used to determine the
             complete set of functionality that both client and server
             has. The intended usage is to determine before one needs to
             use a functionality 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 including 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 features 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 situations 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



H. Schulzrinne et. al.                                       [Page 32]


Internet Draft                    RTSP                  October 27, 2003


   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.


    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


   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
   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.

   The OPTIONS method can be used for RTSP session keep alive
   signalling, however this method is not the most recommended one, see


H. Schulzrinne et. al.                                       [Page 33]


Internet Draft                    RTSP                  October 27, 2003


   section  13.37 for a preference list. A keep alive OPTIONS request
   SHOULD use the media or aggregated control URI.

   Example:



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

     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
   purposefully 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
   initialization 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



H. Schulzrinne et. al.                                       [Page 34]


Internet Draft                    RTSP                  October 27, 2003


           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
           m=application 32416 UDP WB
           a=orient:portrait



   The DESCRIBE response MUST contain all media initialization
   information 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
   parameters, 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
        initialization for the set of streams that it describes,
        and discouraging 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:

        o via RTSP's DESCRIBE method;

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

        o via the command line or standard input (thus working as a
          browser helper application launched with an SDP file or other



H. Schulzrinne et. al.                                       [Page 35]


Internet Draft                    RTSP                  October 27, 2003


          media initialization 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. The SETUP method may be used in two       |
   different cases; Create a RTSP session or add a media to a session,    |
   and change the transport parameters of already set up media stream.    |
   Using SETUP to create or add media to a session when in PLAY state     |
   are not allowed. Otherwise SETUP can be used in all three states;      |
   INIT, and READY, for both purposes and in PLAY to change the           |
   transport parameters.                                                  |

   The Transport header, see section  13.40, specifies the transport      |
   parameters acceptable to the client for data transmission; the         |
   response will contain the transport parameters selected by the         |
   server. This allows the client to enumerate in priority order the      |
   transport mechanisms and parameters acceptable to it, while the        |
   server can select the most appropriate. All transport parameters       |
   SHOULD be included in the Transport header, the use of other headers   |
   for this purpose is discouraged due to middle boxes.                   |

   For the benefit of any intervening firewalls, a client SHOULD          |
   indicate the transport parameters even if it has no influence over     |
   these parameters, for example, where the server advertises a fixed     |
   multicast 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 initialization.                      |

   In a SETUP response the server SHOULD include the Accept-Ranges        |
   header (see section 13.4 to indicate which time formats that are       |
   acceptable to use for this media resource.                             |



     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0               |
           CSeq: 302                                                      |



H. Schulzrinne et. al.                                       [Page 36]


Internet Draft                    RTSP                  October 27, 2003


           Transport: RTP/AVP;unicast;client_port=4588-4589,              |
                      RTP/AVP/TCP;unicast;interleave=0-1                  |

     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;ssrc=2A3F93ED                 |
           Accept-Ranges: NPT                                             |



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

   The server MUST generate a session identifier in response to a         |
   successful SETUP request, unless a SETUP request to a server includes  |
   a session identifier, in which case the server MUST bundle this setup  |
   request into the existing session (aggregated session) or return       |
   error 459 (Aggregate Operation Not Allowed) (see Section  12.4.11).    |
   An Aggregate control URI MUST be used to control an aggregated         |
   session. This URI MUST be different from the stream control URIs of    |
   the individual media streams included in the aggregate. The Aggregate  |
   control URI is to be specified by the session description if the       |
   server supports aggregated control and aggregated control is desired   |
   for the session. However even if aggregated control is offered the     |
   client MAY chose to not set up the session in aggregated control.      |

   If an Aggregate control URI is not specified in the session            |
   description, it is probably a indication that non-aggregated control   |
   should be used. However a client MAY try to SETUP the session in       |
   aggregated control. If the server refuse to aggregate the specified    |
   media, the server SHALL use the 459 error code.  If the server allows  |
   the aggregation, then the client MUST create an URI for aggregate      |
   control of the session. This URI MUST contain the servers host         |
   address and MUST contain the port, if applicable (e.g. not default     |
   port). Once an URI is used to refer to an aggregation for a given      |
   session, that URI MUST be used to refer to that aggregation for the    |
   duration of the session.                                               |




H. Schulzrinne et. al.                                       [Page 37]


Internet Draft                    RTSP                  October 27, 2003


        While the session ID sometimes has enough information for    |
        aggregate control of a session, the Aggregate control URI    |
        is still important for some methods such as SET_PARAMETER    |
        where the control URI enables the resource in question to    |
        be easily identified. The Aggregate control URI is also      |
        useful for proxies, enabling them to route the request to    |
        the appropriate server, and for logging, where it is useful  |
        to note the actual resource that a request was operating     |
        on. Finally, presence of the Aggregate control URI allows    |
        for backwards compatibility with RFC 2326 [21].

   A session will exist until it is either removed by a TEARDOWN request
   or is timed-out by the server. The server MAY remove a session that
   has not demonstrated liveness signs from the client within a certain
   timeout period. The default timeout value is 60 seconds; the server
   MAY set this to a different value and indicate so in the timeout
   field of the Session header in the SETUP response. For further
   discussion see chapter  13.37. Signs of liveness for a RTSP session
   are:

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

        o If RTP is used as a transport for the underlying media
          streams, an RTCP sender or receiver report from the client for
          any of the media streams in that RTSP session.


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

   A client MAY issue a SETUP request for a stream that is already set    |
   up or playing in the session to change transport parameters, which a   |
   server MAY allow. If it does not allow this, it MUST respond with      |
   error 455 (Method Not Valid In This State). Reasons to support         |
   changing transport parameters, is to allow for application layer       |
   mobility and flexibility to utilize the best available transport as    |
   it becomes available.                                                  |

   In a SETUP response for a request to change the transport parameters   |
   while in Play state, the server SHOULD include the Range to indicate   |
   from what point the new transport parameters are used. Further if RTP  |
   is used for delivery the server SHOULD also include the RTP-Info       |
   header to indicate from what timestamp and RTP sequence number the     |
   change has taken place. If both RTP-Info and Range is included in the  |
   response the "rtp_time" parameter and range MUST be for the            |



H. Schulzrinne et. al.                                       [Page 38]


Internet Draft                    RTSP                  October 27, 2003


   corresponding time, i.e. be used in the same way as for PLAY to        |
   ensure the correct synchronization information is present.             |

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


11.4 PLAY

   The PLAY method tells the server to start sending data via the         |
   mechanism specified in SETUP. A client MUST NOT issue a PLAY request   |
   until any outstanding SETUP requests have been acknowledged as         |
   successful. PLAY requests are valid when the session is in READY       |
   state; the use of PLAY requests when the session is in PLAY state is   |
   deprecated. A PLAY request MUST include a Session header to indicate   |
   which session the request applies to.

   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
   sessions for each media.

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

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



     C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: npt=10-15, npt=20-25, npt=30-





H. Schulzrinne et. al.                                       [Page 39]


Internet Draft                    RTSP                  October 27, 2003


   See the description of the PAUSE request for further examples.

   A PLAY request without a Range header is legal. It SHALL start         |
   playing a stream from the beginning (npt=0-) unless the stream has     |
   been paused. If a stream has been paused via PAUSE, stream delivery    |
   resumes at the pause point. The stream SHALL play until the end of     |
   the media.

   The Range header MUST NOT contain a time parameter. The usage of time
   in PLAY method has been deprecated.

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

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

   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. If   |
   the medias that are part of an aggregate has different lengths, the    |
   PLAY request SHALL be performed as long as the given range is valid    |
   for any media, for example the longest media. Media will be sent       |
   whenever it is available for the given play-out point.                 |

   After playing the desired range, the presentation is NOT               |
   automatically paused, media delivery simply stops. A PAUSE request     |
   MUST be issued before another PLAY request can be 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 PLAY state.                                      |

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



H. Schulzrinne et. al.                                       [Page 40]


Internet Draft                    RTSP                  October 27, 2003


   pause point in a Range header SHALL be included.

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



   C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
         CSeq: 833
         Session: 12345678
         Range: smpte=0:10:20-

   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-
         RTP-Info:url=rtsp://example.com/twister.en;
            seq=14783;rtptime=2345962545



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



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




   All range specifiers in this specification allow for ranges with
   unspecified begin times (e.g. "npt=-30"). When used in a PLAY
   request, the server treats this as a request to start/resume playback
   from the current pause point, ending at the end time specified in the



H. Schulzrinne et. al.                                       [Page 41]


Internet Draft                    RTSP                  October 27, 2003


   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
   supported.

   The use of PLAY for keep-alive signaling, i.e. PLAY request without a  |
   range header in PLAY state, has also been depreciated. 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.

11.5 PAUSE

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


   Example:                                                               |



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



H. Schulzrinne et. al.                                       [Page 42]


Internet Draft                    RTSP                  October 27, 2003


   "pause point". The time parameter in the Range MUST NOT be used. The
   Range 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 normal 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
           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,



H. Schulzrinne et. al.                                       [Page 43]


Internet Draft                    RTSP                  October 27, 2003


   the pause point in the media stream MUST be maintained. A subsequent
   PLAY request without Range header SHALL resume from the pause point
   and play until media end.

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

   The pause point after any PAUSE request SHALL be returned to the       |
   client by adding a Range header with what remains unplayed of the      |
   PLAY request's ranges, i.e. including all the remaining ranges part    |
   of multiple range specification. If one desires to resume playing a    |
   ranged request, one simple included the Range header from the PAUSE    |
   response. Note that this server behavior was not mandated previously   |
   and servers implementing according to RFC 2326 will probably not       |
   return the range header.                                               |

   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        |
   second, overlapping range. Regardless of when the PAUSE request        |
   arrives, it sets the pause point to 14. The below example messages is  |
   for the above case when the PAUSE request arrives before the first     |
   occurrence of NPT=14.                                                  |



     C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0                    |
           CSeq: 834                                                      |
           Session: 12345678                                              |
           Range: npt=10-15, npt=13-20                                    |

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



H. Schulzrinne et. al.                                       [Page 44]


Internet Draft                    RTSP                  October 27, 2003


           Server: PhonyServer 1.0                                        |
           Range: npt=10-15, npt=13-20                                    |
           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                   |
           CSeq: 835                                                      |
           Session: 12345678                                              |
           Range: npt=14-                                                 |

     S->C: RTSP/1.0 200 OK                                                |
           CSeq: 835                                                      |
           Date: 23 Jan 1997 15:35:09 GMT                                 |
           Server: PhonyServer 1.0                                        |
           Range: npt=14-15, npt=13-20                                    |
           Session: 12345678                                              |



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

   Examples:



     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 834
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 834
           Session: 12345678
           Date: 23 Jan 1997 15:35:06 GMT
           Range: npt=45.76-

     C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range:  86-




H. Schulzrinne et. al.                                       [Page 45]


Internet Draft                    RTSP                  October 27, 2003


     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Session: 12345678
           Date: 23 Jan 1997 15:35:07 GMT
           Range: npt=45.76-



11.6 TEARDOWN

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

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

   A TEARDOWN using a media URI in an aggregated session MAY only be      |
   done in Ready state. Such a request only removes the indicated media   |
   stream and associated resources from the session.  This may result in  |
   that a session returns to non-aggregated control. In the response to   |
   TEARDOWN request resulting in that the session still exist SHALL       |
   contain a Session header to indicate this.                             |

   Note, the indication with the session header if sessions state remain  |
   may not be done correctly by a RFC 2326 client, but will be for any    |
   server signalling the "play.basic" 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






H. Schulzrinne et. al.                                       [Page 46]


Internet Draft                    RTSP                  October 27, 2003


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
   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
   presentation 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



H. Schulzrinne et. al.                                       [Page 47]


Internet Draft                    RTSP                  October 27, 2003


   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.

   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



H. Schulzrinne et. al.                                       [Page 48]


Internet Draft                    RTSP                  October 27, 2003


   A redirect request informs the client that it MUST connect to another  |
   server location. The REDIRECT request MAY contain the header           |
   Location, which indicates that the client should issue requests for    |
   that URL. The lack of a Location header in the response SHALL be       |
   interpreted as that the server can't any longer fulfill the current    |
   request, but has no alternative at the present where the client        |
   continue.                                                              |

   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 the Location header is included    |
   it SHALL contain a full absolute URI pointing out the resource to      |
   reconnect too, i.e. the Location SHALL NOT contain only host and       |
   port.                                                                  |

   If a REDIRECT request does not contain a Session header, it is next-   |
   hop and applies also to the control connection.  If the Location       |
   header is included it SHOULD only contain an absolute URI containing   |
   the host address and OPTIONAL the port number. If there are proxies    |
   in the request chain, they SHOULD do all of the following: (1)         |
   respond to the REDIRECT 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 session 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-.

   A server upon receiving a successful (2xx) response for a REDIRECT     |
   request without any Range header SHALL consider the session as         |
   removed and can free any session state. For this type of requests the  |
   rest of this paragraph applies. The server MAY close the signalling    |
   connection upon receiving the response for REDIRECT requests without   |
   a Session header. The client SHOULD close the signaling connection     |
   after having given the 2xx response to a REDIRECT response, unless it  |
   has several sessions on the server. If the client has multiple         |



H. Schulzrinne et. al.                                       [Page 49]


Internet Draft                    RTSP                  October 27, 2003


   session on the server it SHOULD close the connection when it has       |
   received and responded to REDIRECT requests for all sessions.          |

   A client receiving a REDIRECT request with a Range header SHALL issue  |
   a TEARDOWN request when the in indicated redirect point is reached.    |
   The client SHOULD for REDIRECT requests with Range header close the    |
   signalling connection after a 2xx response on its TEARDOWN request.    |
   The normal connection considerations apply for the server. This        |
   differentiation from REDIRECT requests without range headers is to     |
   allow clear an explicit state handling. As the state in the server     |
   needs to be kept until the point of redirection, the handling becomes  |
   more clear if the client is required to tear down the session at that  |
   point.                                                                 |

   If the client wants to continue to send or receive media for this      |
   resource, the client will have to establish a new session 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
           Range: npt=0- ;time=19960213T143205Z
           Session: uZ3ci0K+Ld-M



11.10 PING

   This method is a bi-directional mechanism for server or client
   liveness 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



H. Schulzrinne et. al.                                       [Page 50]


Internet Draft                    RTSP                  October 27, 2003


   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,
   followed by the length of the encapsulated binary data as a binary,
   two-byte integer in network byte order. The stream data follows
   immediately afterwards, without a CRLF, but including the upper-layer
   protocol headers. Each $ block contains exactly one upper-layer
   protocol data unit, e.g., one RTP packet.




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



H. Schulzrinne et. al.                                       [Page 51]


Internet Draft                    RTSP                  October 27, 2003


      +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+





   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
   directions.


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




H. Schulzrinne et. al.                                       [Page 52]


Internet Draft                    RTSP                  October 27, 2003


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



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
   indicate what time period it may still be able to record. Since other
   processes on the server may be consuming storage space
   simultaneously, 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.

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

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

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



H. Schulzrinne et. al.                                       [Page 53]


Internet Draft                    RTSP                  October 27, 2003


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
   automatically to the given URI. This response MUST NOT contain a
   message-body.

12.3.3 302 Found

   The requested resource reside temporarily at the URI given by the
   Location header. The Location header MUST be included in the
   response. 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. The user client SHOULD redirect automatically to the given
   URI. This response MUST NOT contain a message-body.

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 may 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:

        o Date

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

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

   This response is independent for the DESCRIBE and SETUP requests.
   That is, a 304 response to DESCRIBE does NOT imply that the resource
   content is unchanged 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.




H. Schulzrinne et. al.                                       [Page 54]


Internet Draft                    RTSP                  October 27, 2003


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
   modifications [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
   failure.

12.4.6 454 Session Not Found

   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



H. Schulzrinne et. al.                                       [Page 55]


Internet Draft                    RTSP                  October 27, 2003


   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
   modified. 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
   specification.

12.4.14 462 Destination Unreachable

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

12.5 Server Error 5xx




H. Schulzrinne et. al.                                       [Page 56]


Internet Draft                    RTSP                  October 27, 2003


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
   syntax, 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
   processing is summarized in Table 4 and Table 5.

   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;


H. Schulzrinne et. al.                                       [Page 57]


Internet Draft                    RTSP                  October 27, 2003


        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
             context of the message.

        m: The header field is mandatory.

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

        o: The header field is optional.

        *: The header field is 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
   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
   applicable" means that the header field MUST NOT be present in a
   request. If one is placed in a request by mistake, it MUST be ignored
   by the Client/Server receiving the request. Similarly, a header field
   labeled "not applicable" for a response means that the Client/Server



H. Schulzrinne et. al.                                       [Page 58]


Internet Draft                    RTSP                  October 27, 2003


   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
   double quotas ("). Any URI parameters are contained within these
   quotas. If the URI is not enclosed in double quotas, any semicolon-
   delimited parameters are header-parameters, not URI parameters.




13.1 Accept

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

        The "level" parameter for presentation descriptions is
        properly 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
   presentation description and any reason phrases, not the media
   content.

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:  |



H. Schulzrinne et. al.                                       [Page 59]


Internet Draft                    RTSP                  October 27, 2003


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


   This header has the same syntax as [H14.5]. However new range-units    |
   are defined. Inclusion of any of the time formats indicates            |
   acceptance by the server for PLAY and PAUSE requests with this         |
   format. The headers value is valid for the resource specified by the   |
   URI in the request, this response corresponds to. A server is SHOULD   |
   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
   during an RTSP session, e.g., due to modem retraining.


   Bandwidth  =  "Bandwidth" ":" 1*DIGIT


   Example:



H. Schulzrinne et. al.                                       [Page 60]


Internet Draft                    RTSP                  October 27, 2003





   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                              -   -    -    -     -   -
   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*   m*   -
   Referer               R           o   o    o    o     o   o
   Require               R           o   o    o    o     o   o
   Retry-After        3rr,503        o   o    o    -     -   -


H. Schulzrinne et. al.                                       [Page 61]


Internet Draft                    RTSP                  October 27, 2003


   Header           Where Proxy DES OPT SETUP PLAY PAUSE TRD
   _________________________________________________________
   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.


     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




H. Schulzrinne et. al.                                       [Page 62]


Internet Draft                    RTSP                  October 27, 2003





   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                              -   -   -   -                 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


H. Schulzrinne et. al.                                       [Page 63]


Internet Draft                    RTSP                  October 27, 2003


   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_PARAMETER, SET_PARAMETER,REDIRECT, and PING.


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

   Cache-Control should only be specified in a SETUP request and its
   response. Note: Cache-Control does not govern the caching of
   responses as for HTTP, but rather of the stream identified by the
   SETUP request. Responses to RTSP requests are not cacheable, except
   for responses to DESCRIBE.


   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"


H. Schulzrinne et. al.                                       [Page 64]


Internet Draft                    RTSP                  October 27, 2003


                            /   "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
             anywhere. This allows an origin server to prevent caching
             even by caches that have been configured to return stale
             responses to client requests.

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

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

        no-transform: An intermediate cache (proxy) may find it useful
             to convert the media type of a certain stream. A proxy
             might, for example, convert between video formats to save
             cache space or to reduce the amount of traffic on a slow
             link. Serious operational problems may occur, however, when
             these transformations have been applied to streams intended
             for certain kinds of applications. For example,
             applications for medical imaging, scientific data analysis
             and those using end-to-end authentication all depend on
             receiving a stream that is bit-for-bit identical to the
             original 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.

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



H. Schulzrinne et. al.                                       [Page 65]


Internet Draft                    RTSP                  October 27, 2003


        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
             meaning 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 client has supplied its own validator in the request,
             the supplied validator might differ from the validator
             currently stored with the cache entry. In this case, the
             cache MAY use either validator in making its own request
             without affecting semantic transparency.

             However, the choice of validator might affect performance.
             The best approach is for the intermediate cache to use its
             own validator when making its request. If the server
             replies with 304 (Not Modified), then the cache can return
             its now validated copy to the client with a 200 (OK)



H. Schulzrinne et. al.                                       [Page 66]


Internet Draft                    RTSP                  October 27, 2003


             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
   messages SHOULD be limited to error messages when the server is
   unable to recover and therefore see it necessary to close the
   connection. The reason is that the client 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

   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



H. Schulzrinne et. al.                                       [Page 67]


Internet Draft                    RTSP                  October 27, 2003


   [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
   number. Any retransmitted request must contain the same sequence
   number as the original (i.e. the sequence number is not incremented
   for retransmissions of the same request). For each new RTSP request
   the CSeq value SHALL be incremented by one. The initial sequence
   number MAY be any number. 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.

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
   interpretation 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



H. Schulzrinne et. al.                                       [Page 68]


Internet Draft                    RTSP                  October 27, 2003


   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).

13.20 From

   See [H14.22].

13.21 Host

   The Host HTTP request header field [H14.23] is not needed for RTSP,    |
   and SHALL NOT be sent. It SHALL be silently ignored if received.



H. Schulzrinne et. al.                                       [Page 69]


Internet Draft                    RTSP                  October 27, 2003


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
   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



H. Schulzrinne et. al.                                       [Page 70]


Internet Draft                    RTSP                  October 27, 2003


   See [H14.33].

13.27 Proxy-Require

   The Proxy-Require request-header field is used to indicate proxy-      |
   sensitive features that MUST be supported by the proxy. Any Proxy-     |
   Require header features that are not supported by the proxy MUST be    |
   negatively acknowledged by the proxy to the client using the           |
   Unsupported header. Any feature tag included in the Proxy-Require      |
   does not apply to the server. To ensure that a feature is supported    |
   by both proxies and servers the tag must be included in also a         |
   Require header.

   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                                              |



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
   methods. 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:


   Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN






H. Schulzrinne et. al.                                       [Page 71]


Internet Draft                    RTSP                  October 27, 2003


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
   specification 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 normally not meaningful.  The header MAY contain a time parameter
   in UTC, specifying the time at which the operation is to be made
   effective. This functionality SHALL only be used with the REDIRECT
   method.  Servers supporting the Range header MUST understand the NPT
   range format and SHOULD understand the SMPTE range format. The Range
   response header indicates what range of time is actually 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 first point, but
   excluding the second point. In other words, a range of A-B starts
   exactly at time A, but stops just before B. Only the start time of a
   media unit such as a video or audio frame is relevant. 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 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



H. Schulzrinne et. al.                                       [Page 72]


Internet Draft                    RTSP                  October 27, 2003


        in the future, although a server may refuse to keep server
        resources for extended idle periods.

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

   Example:


       Range: npt=10-15



   However, range intervals can also decrease if the Scale header (see
   section  13.34) indicates a negative scale value. For example, this
   would be the case when a playback in reverse is desired.

   Example:


       Scale: -1
       Range: npt=15-10



   Decreasing ranges are still half open intervals as described above.
   Thus, For range A-B, A is closed and B is open. In the above example,
   15 is closed and 10 is open. An exception to this rule is the case
   when B=0 in a decreasing range. In this case, the range is closed on
   both ends, as otherwise there would be no way to reach 0 on a reverse
   playback.

   Example:


       Scale: -1
       Range: npt=15-0



   In this range both 15 and 0 are closed.

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

13.30 Referer

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



H. Schulzrinne et. al.                                       [Page 73]


Internet Draft                    RTSP                  October 27, 2003


   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
   Supported). This header does not apply to proxies, for the same
   functionality 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
           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



H. Schulzrinne et. al.                                       [Page 74]


Internet Draft                    RTSP                  October 27, 2003


           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
   transmitted 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
   parameters in the PLAY response. For streams using RTP as transport
   protocol the RTP-Info header SHALL be part of a 200 response to PLAY.

   The RTP-Info response-header field is used to set RTP-specific
   parameters in the PLAY response. These parameters correspond to the
   synchronization source identified by the ssrc parameter of the
   Transport response header in the SETUP reponse. For streams using RTP
   as transport protocol 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
             control, 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



H. Schulzrinne et. al.                                       [Page 75]


Internet Draft                    RTSP                  October 27, 2003


             number indicated by seq actually has the timestamp
             indicated by rtptime.) 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 timestamps to NPT. Furthermore, in order to
             ensure that this information is available at the
             necessary time (immediately at startup or after a
             seek), and that it is delivered reliably, this mapping
             is placed in the RTSP control channel.

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

   Additionally, the RTP-Info header parameter fields only apply to a     |
   single SSRC within a stream (the SSRC reported in the transport        |
   response header; see section  13.40). If there are multiple            |
   synchronization sources (SSRCs) present within a RTP session, RTCP     |
   must be used to map RTP and NTP timestamps for those sources, for      |
   both synchronization and drift-checking.

   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
   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.




H. Schulzrinne et. al.                                       [Page 76]


Internet Draft                    RTSP                  October 27, 2003


   Example:


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



13.34 Scale

   A scale value of 1 indicates normal play at the normal forward
   viewing rate. If not 1, the value corresponds to the rate with
   respect to normal viewing rate. For example, a ratio of 2 indicates
   twice the normal viewing rate ("fast forward") and a ratio of 0.5
   indicates half the normal viewing rate. In other words, a ratio of 2
   has normal play time increase at twice the wallclock rate. For every
   second of elapsed (wallclock) time, 2 seconds of content will be
   delivered.  A negative value indicates reverse direction.

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

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

   If the server does not implement the possibility to scale, it will
   not return a Scale header. A server supporting Scale operations for
   PLAY SHALL indicate this with the use of the "play.scale" feature-
   tags.


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


   When indicating a negative scale for a reverse playback, the Range
   header must indicate a decreasing range as described in section
   13.29.

   Example of playing in reverse at 3.5 times normal rate:


     Scale: -3.5
     Range: npt=15-10



H. Schulzrinne et. al.                                       [Page 77]


Internet Draft                    RTSP                  October 27, 2003


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.
   Implementation by the server is OPTIONAL. The default is the bit rate
   of the stream.

   The parameter value is expressed as a decimal ratio, e.g., a value of
   2.0 indicates that data is to be delivered twice as fast as normal. A
   speed of zero is invalid. All speeds may not be possible to support.
   Therefore the actual used speed MUST be included in the response.
   The lack of a response header is indication of lack of support from
   the server of this functionality. Support of the speed functionality
   are indicated by the "play.speed" 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           |
   presentation at a higher or lower rate is necessary. Implementors      |
   should keep in mind that bandwidth for the session may be negotiated   |
   beforehand (by means other than RTSP), and therefore re-negotiation    |
   may be necessary. When data is delivered over UDP, it is highly        |
   recommended that means such as RTCP be used to track packet loss       |
   rates. If the data transport is performed over public best-effort      |
   networks the sender SHOULD perform congestion control of the           |
   stream(s). This can 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



H. Schulzrinne et. al.                                       [Page 78]


Internet Draft                    RTSP                  October 27, 2003


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

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

   Note that RFC 2326 servers and client may in some cases not include    |
   or return a Session header when expected according to the above text.  |
   Any client or server is RECOMMENDED to be forgiving of this error if   |
   possible (which it is in many cases).


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


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

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

        RTCP: If RTP is used for media transport RTCP SHOULD be used. If  |



H. Schulzrinne et. al.                                       [Page 79]


Internet Draft                    RTSP                  October 27, 2003


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

        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      |
             downside 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 also 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 liveness timer has not been          |
             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 limited. The same considerations as for PING apply     |
             regarding 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. RTSP requests for a given session   |
   can use different URIs (Presentation and media URIs).  Note, that      |
   there are restrictions depending on the session which URIs that are    |
   acceptable for a given method. However, multiple "user" sessions for   |
   the same URI from the same client will require use of different        |



H. Schulzrinne et. al.                                       [Page 80]


Internet Draft                    RTSP                  October 27, 2003


   session identifiers.                                                   |

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

   The response 454 (Session Not Found) SHALL be returned if the session  |
   identifier 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
   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 ]



H. Schulzrinne et. al.                                       [Page 81]


Internet Draft                    RTSP                  October 27, 2003


   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
   destination port for a single stream. It sets those values not
   already determined by a presentation description.

   Transports are comma separated, listed in order of preference.
   Parameters may be added to each transport, separated by a semicolon.

   The Transport header field MAY also be used to change certain
   transport 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
   transportspec 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.
   Additionally, it may only contain the unicast or multicast transport
   parameter.


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

   The 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:




H. Schulzrinne et. al.                                       [Page 82]


Internet Draft                    RTSP                  October 27, 2003


   General parameters:

        unicast / multicast: This parameter is a mutually exclusive
             indication of whether unicast or multicast delivery will be
             attempted. One of the two values MUST be specified. Clients
             that are capable of handling both unicast and multicast
             transmission 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 specified, the recipient may be a
             different party than the originator of the request. To
             avoid becoming the unwitting perpetrator of a remote-
             controlled denial-of-service attack, a server 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
             recipient itself and the credentials passed along to the
             server. However, in certain cases, such as when recipient
             address is a multicast group, or when the recipient is
             unable to communicate with the server in an out-of-band
             manner, this may not be possible. In these cases server may
             chose another method such as a server-resident
             authorization list to ensure that the request originator
             has the proper credentials to request stream delivery to
             the recipient.

             This parameter SHALL NOT be used when src_addr and dst_addr  |
             is used in a transport declaration. For IPv6 addresses it    |
             is RECOMMENDED that they be given as fully qualified domain  |
             to make it backwards compatible with RFC 2326                |
             implementations.

        source: If the source address for the stream is different than
             can be derived from the RTSP endpoint address (the server
             in playback), the source address SHOULD be specified. To



H. Schulzrinne et. al.                                       [Page 83]


Internet Draft                    RTSP                  October 27, 2003


             maintain backwards compatibility with RFC 2326, any IPv6
             host's address must be given as a fully qualified domain
             name. This parameter SHALL NOT be used when src_addr and
             dst_addr is used in a transport declaration.


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

        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_addr: A general destination 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 indicated by the
             use of 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".

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

        src_addr: 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 indicated 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".

             The address or addresses indicated in the src_addr
             parameter SHOULD be used both for sending and receiving of



H. Schulzrinne et. al.                                       [Page 84]


Internet Draft                    RTSP                  October 27, 2003


             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
             provided, 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.




H. Schulzrinne et. al.                                       [Page 85]


Internet Draft                    RTSP                  October 27, 2003


        port: This parameter provides the RTP/RTCP port pair for a
             multicast 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 is used in a transport declaration.

        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 is used in a transport declaration.

        ssrc: The ssrc parameter, if included in a SETUP response,
             indicates the RTP SSRC [23] value that will be used by the
             media server for RTP packets within the stream. It is
             expressed as an eight digit hexadecimal value. If the
             server does not act as a synchronization source for stream
             data (for instance, server is a translator, reflector,
             etc.) the value will be the "packet sender's SSRC" that
             would have been used in the RTCP Receiver Reports generated
             by the server, regardless of whether the server actually
             generates RTCP RRs. If there are multiple sources within
             the stream, the ssrc parameter only indicates the value for
             a single synchronization source. Other sources must be
             deduced from the actual RTP/RTCP stream.

             The functionality of specifying the ssrc parameter in a
             SETUP request is deprecated as it is incompatible with the
             specification of RTP in RFC 3550. If the parameter is
             included in the transport header of a SETUP request, the
             server MAY ignore it, and choose an appropriate SSRC for
             the stream.  The server MAY set the ssrc parameter in the
             transport header of the response.


   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 ]



H. Schulzrinne et. al.                                       [Page 86]


Internet Draft                    RTSP                  October 27, 2003


                           /   ";" "append"
                           /   ";" "ttl" "=" ttl
                           /   ";" "layers" "=" 1*DIGIT
                           /   ";" "port" "=" port-spec
                           /   ";" "client_port" "=" port-spec
                           /   ";" "server_port" "=" port-spec
                           /   ";" "ssrc" "=" ssrc
                           /   ";" "mode" "=" mode-spec
                           /   ";" "dest_addr" "=" addr-list
                           /   ";" "src_addr" "=" 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                =  quoted-host-port *("/" quoted-host-port)
   quoted-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"



H. Schulzrinne et. al.                                       [Page 87]


Internet Draft                    RTSP                  October 27, 2003


13.41 Unsupported

   The Unsupported response-header field lists the features not
   supported by the server. In the case where the feature was specified
   via the Proxy-Require field (Section 13.27), if there is a proxy on
   the path between the client and the server, the proxy MUST send a
   response message with a status code of 551 (Option Not Supported).
   The request SHALL NOT be forwarded.

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



H. Schulzrinne et. al.                                       [Page 88]


Internet Draft                    RTSP                  October 27, 2003


   On receiving a SETUP or PLAY request, a proxy ascertains whether it
   has an up-to-date copy of the continuous media content and its
   description. It can determine whether the copy is up-to-date by
   issuing a SETUP or DESCRIBE request, respectively, and comparing the
   Last-Modified header with that of the cached copy. If the copy is not
   up-to-date, it modifies the SETUP transport parameters as appropriate
   and forwards the request to the origin server. Subsequent control
   commands such as PLAY or PAUSE then pass the proxy unmodified. The
   proxy delivers the continuous media data to the client, while
   possibly making a local copy for later reuse. The exact behavior
   allowed to the cache is given by the cache-response directives
   described in Section 13.9. A cache MUST answer any DESCRIBE requests
   if it is currently serving the stream to the requestor, as it is
   possible that low-level details of the stream description may have
   changed on the origin-server.

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

   To the client, an RTSP proxy cache appears like a regular media
   server, to the media origin server like a client. Just as an HTTP
   cache has to store the content type, content language, and so on for
   the objects it caches, a media cache has to store the presentation
   description.  Typically, a cache eliminates all transport-references
   (that is, 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
   presentation and all its streams, including the codecs that are
   available, dynamic RTP payload types, the protocol stack, and content
   information such as language or copyright restrictions. It may also



H. Schulzrinne et. al.                                       [Page 89]


Internet Draft                    RTSP                  October 27, 2003


   give an indication about the timeline of the movie.

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



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



H. Schulzrinne et. al.                                       [Page 90]


Internet Draft                    RTSP                  October 27, 2003


         User-Agent: PhonyClient/1.2
         Session: 23456789
         Range: smpte=0:10:00-

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



H. Schulzrinne et. al.                                       [Page 91]


Internet Draft                    RTSP                  October 27, 2003


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.
   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
   control 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



H. Schulzrinne et. al.                                       [Page 92]


Internet Draft                    RTSP                  October 27, 2003


         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
         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




H. Schulzrinne et. al.                                       [Page 93]


Internet Draft                    RTSP                  October 27, 2003


   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

   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
   description 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



H. Schulzrinne et. al.                                       [Page 94]


Internet Draft                    RTSP                  October 27, 2003


             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

       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






H. Schulzrinne et. al.                                       [Page 95]


Internet Draft                    RTSP                  October 27, 2003


   In the worst case, servers should send back:



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



   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



H. Schulzrinne et. al.                                       [Page 96]


Internet Draft                    RTSP                  October 27, 2003


         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

   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



H. Schulzrinne et. al.                                       [Page 97]


Internet Draft                    RTSP                  October 27, 2003


                      (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       =  "(" / ")" / "<" / ">" / "@"
                  /   "," / ";" / ":" / 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



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




H. Schulzrinne et. al.                                       [Page 98]


Internet Draft                    RTSP                  October 27, 2003


     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   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



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



H. Schulzrinne et. al.                                       [Page 99]


Internet Draft                    RTSP                  October 27, 2003


                  /  "305"           ; Use Proxy
                  /  "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>



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



H. Schulzrinne et. al.                                      [Page 100]


Internet Draft                    RTSP                  October 27, 2003


                /  Timestamp          ; Section 13.39
                /  Via                ; Section 13.44
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



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 )



H. Schulzrinne et. al.                                      [Page 101]


Internet Draft                    RTSP                  October 27, 2003


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 ] ]



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



H. Schulzrinne et. al.                                      [Page 102]


Internet Draft                    RTSP                  October 27, 2003


                           /   ";" "client_port" "=" port-spec
                           /   ";" "server_port" "=" port-spec
                           /   ";" "ssrc" "=" ssrc
                           /   ";" "client_ssrc" "=" ssrc
                           /   ";" "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                =  quoted-host-port *("/" quoted-host-port)
   quoted-host-port         =  <"> host [":" port]<">


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
             authentication 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 mechanisms. Also see [H15.6].

        Abuse of Server Log Information: RTSP and HTTP servers will
             presumably have similar logging mechanisms, and thus should
             be equally guarded in protecting the contents of those
             logs, thus protecting the privacy of the users of the
             servers. See [H15.1.1] for HTTP server recommendations
             regarding server logs.

        Transfer of Sensitive Information: There is no reason to believe
             that information transferred via RTSP may be any less
             sensitive than that normally transmitted via HTTP.
             Therefore, all of the precautions regarding the protection
             of data privacy and user privacy apply to implementors of
             RTSP clients, servers, and proxies. See [H15.1.2] for
             further details.



H. Schulzrinne et. al.                                      [Page 103]


Internet Draft                    RTSP                  October 27, 2003


        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 portions of the request URLs directly to file
             system calls. In such cases, file systems SHOULD follow the
             precautions outlined in [H15.5], such as checking for ".."
             in path components.

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

        Privacy Issues Connected to Accept Headers: Since may of the
             same "Accept" headers exist in RTSP as in HTTP, the same
             caveats outlined in [H15.1.4] with regards to their use
             should be followed.

        DNS Spoofing: Presumably, given the longer connection times
             typically associated to RTSP sessions relative to HTTP
             sessions, RTSP client DNS optimizations should be less
             prevalent.  Nonetheless, the recommendations provided in
             [H15.3] are still relevant to any implementation which
             attempts to rely on a DNS-to-IP mapping to hold beyond a
             single use of the mapping.

        Location Headers and Spoofing: If a single server supports
             multiple organizations that do not trust 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 invalidate 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



H. Schulzrinne et. al.                                      [Page 104]


Internet Draft                    RTSP                  October 27, 2003


             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-initiated 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.

        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
             delivery issues are not covered in this section, nor in the
             rest of this draft. RTSP implementations will most likely
             rely on other protocols such as RTP, IP multicast, RSVP and
             IGMP, and should address security considerations brought up
             in those and other applicable specifications.

        Persistently suspicious behavior: RTSP servers SHOULD return
             error code 403 (Forbidden) upon receiving a single instance
             of behavior which is deemed a security risk. RTSP servers
             SHOULD also be aware of attempts to probe the server for
             weaknesses and entry points and MAY arbitrarily disconnect
             and ignore further requests clients which are deemed to be
             in violation of local security policy.

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
   register. See also the section 1.6 "Extending RTSP". There is also a
   IANA registration of two SDP attributes.

   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".



H. Schulzrinne et. al.                                      [Page 105]


Internet Draft                    RTSP                  October 27, 2003


   A registration request to IANA MUST contain the following
   information:

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

        o Indication of who has change control over the feature (for
          example, IETF, ISO, ITU-T, other international standardization
          bodies, a consortium or a particular company or group of
          companies);

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

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

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
   contains 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.  The          |
   registration SHALL indicate if the feature tag applies to servers      |
   only, proxies only or both server and proxies. Any proprietary         |
   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.



H. Schulzrinne et. al.                                      [Page 106]


Internet Draft                    RTSP                  October 27, 2003


        play.basic: The minimal implementation for playback operations    |
             according to section D. Applies for both servers and         |
             proxies.                                                     |

        play.scale: Support of scale operations for media playback.       |
             Applies only for servers.                                    |

        play.speed: Support of the speed functionality for playback.      |
             Applies only for servers                                     |

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
   document. The reason is that new methods may radically change the
   protocols behavior and purpose.

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

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

        o A clear specification on what action and response a request
          with the method will result in. Which directions the method is
          used, C -> S or S -> C or both. How the use of headers, if
          any, modifies the behavior and effect of the method.

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

        o Describe how the method relates to network proxies.

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



H. Schulzrinne et. al.                                      [Page 107]


Internet Draft                    RTSP                  October 27, 2003


   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
   following:

        o The requested number.

        o A description what the status code means and the expected
          behavior of the sender and receiver of the code.

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
   different ways. An unknown header will be ignored by the receiving
   entity. If the new header is vital for a certain functionality, a
   feature-tag for the functionality can be created and demanded to be
   used by the counter-part with the inclusion of a Require header
   carrying the feature-tag.

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:

        o The name of the header.

        o A BNF specification of the header syntax.

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

        o How the header shall be handled by proxies.



H. Schulzrinne et. al.                                      [Page 108]


Internet Draft                    RTSP                  October 27, 2003


        o 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
   specifications are registered:

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

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

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

        o x-predecbufsize defined in [35].

        o x-initpredecbufperiod defined in [35].

        o 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
   possibilities 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:

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.

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

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

   This specification register 1 value:

        o Use of the RTP  [23] protocol for media transport. The usage



H. Schulzrinne et. al.                                      [Page 109]


Internet Draft                    RTSP                  October 27, 2003


          is explained in RFC XXXX, appendix B.1.

18.5.2 Profile

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

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.

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

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

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

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

18.5.3 Lower Transport

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

        o Registering requires public available standards specification.

        o A contact person or organization with address and email.

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

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

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

        o 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



H. Schulzrinne et. al.                                      [Page 110]


Internet Draft                    RTSP                  October 27, 2003


   following rules:

        o Registering requires a IETF standard tracks document.

        o A contact person or organization with address and email.

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

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

        o See RFC XXXX.

        o 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:

        o Registering requires a IETF standard tracks document.

        o A registration shall name a contact person.

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

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


18.7 SDP attributes

 This specification defines two SDP [24] attributes that it is requested
 that IANA register.



   SDP Attribute ("att-field"):

        Attribute name:     range
        Long form:          Media Range Attribute
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX



H. Schulzrinne et. al.                                      [Page 111]


Internet Draft                    RTSP                  October 27, 2003


        Values:             See ABNF definition.

        Attribute name:     control
        Long form:          RTSP control URL
        Type of name:       att-field
        Type of attribute:  Media and session level
        Subject to charset: No
        Purpose:            RFC XXXX
        Reference:          RFC XXXX
        Values:             Absolute or Relative URLs.




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     |
   one or more media streams depending on state. If a single media        |
   stream is part of the session it is in non-aggregated control. If two  |
   or more is part of the session it is in aggregated control.            |

   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 three states, described below. For each     |
   state there exist a table which shows which requests and events that   |
   is allowed and if they will result in a state change.                  |

        Init: Initial state no session exist.                             |

        Ready: Session is ready to start playing.                         |

        Play: Session is playing, i.e. sending media stream data in the   |
             direction S -> C.                                            |

A.2 State variables                                                       |

   This representation of the state machine needs more than its state to  |
   work. A small number of variables are also needed and is explained     |
   below.                                                                 |



H. Schulzrinne et. al.                                      [Page 112]


Internet Draft                    RTSP                  October 27, 2003


        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.      |

   The response to valid request meeting the requisites is normally a     |
   2xx (SUCCESS) unless other noted in the response column. The           |
   exceptions 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  |



H. Schulzrinne et. al.                                      [Page 113]


Internet Draft                    RTSP                  October 27, 2003


   state, except if the error is not possible to recover from. A          |
   unrecoverable error SHALL result the ending of the session. As it in   |
   the general case can't be determined if it was a unrecoverable error   |
   or not the client will be required to test. In the case that the next  |
   request after a 5xx is responded with 454 (Session Not Found) the      |
   client SHALL assume that the session has been ended.                   |

   The server will timeout the session after the period of time           |
   specified in the SETUP response, if no activity from the client is     |
   detected.  Therefore there exist a timeout event for all states        |
   except Init.                                                           |

   In the case that NRM=1 the presentation 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.                                    |


             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   |



H. Schulzrinne et. al.                                      [Page 114]


Internet Draft                    RTSP                  October 27, 2003


   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                   New         URL                      Ready
   NRM+=1                    SETUP              Setten   up   URL
   Ready       Change    transport    param.    TEARDOWN             Prs
   URL,NRM>1        Init     No session hdr          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        PAUSE              Prs   URL
   Ready       Return    PP                     S -> C:REDIRECT    Range
   hdr            Ready    Set RedP                 S -> C:REDIRECT   no
   range     hdr             Init         Session     is    removed
   Timeout
   Init                                                             RedP
   reached                          Ready    TEARDOWN of session


   Table 8: State: Ready


   In the Ready state, see Table 8, 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 the presentations     |
   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       |
   response. If only a single media stream remains in the session when    |
   performing a TEARDOWN with a media URL the session is removed. The     |
   number of media streams remaining after tearing down a media stream    |
   determines the new state.                                              |


   The Play state table, see Table 9, is the largest. The table contains  |
   an number of request that has presentation URL as a prerequisite on    |
   the request URL, this is due to the exclusion of non-aggregated        |
   stream control in sessions with more than one media stream.            |


H. Schulzrinne et. al.                                      [Page 115]


Internet Draft                    RTSP                  October 27, 2003



   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                     Play
   455                          SETUP              Setuped   URL
   Play      455                         SETUP             Setuped  URL,
   IFI     Play      Change  transport  param.     TEARDOWN          Prs
   URL,NRM>1       Init     No session hdr              TEARDOWN
   md    URL,NRM=1           Init       No   Session   hdr,   NRM=0
   TEARDOWN                 md         URL                      Play
   455                          S -> C:REDIRECT    Range   hdr
   Play      Set  RedP                     S -> C:REDIRECT    no   range
   hdr           Init        Session    is    removed               RedP
   reached                         Play     TEARDOWN of  session
   Timeout                                   Init         Stop     Media
   playout


   Table 9: State: Play

   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          |
   automatic transitions in "RedP reached" and "PP reached", however      |
   they are requested and acknowledge before they take place. The time    |
   at which the transition will happen is known by looking at the range   |
   header. If the client sends request close in time to these             |
   transitions it must be prepared for getting error message as the       |
   state may or may not have changed.

B Media Transport Alternatives

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




H. Schulzrinne et. al.                                      [Page 116]


Internet Draft                    RTSP                  October 27, 2003


B.1 RTP

   This section defines the interaction and needed media transport
   signalling in regards to the RTP protocol [23].

   RTSP allows media clients to control selected, non-contiguous
   sections 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.

   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



H. Schulzrinne et. al.                                      [Page 117]


Internet Draft                    RTSP                  October 27, 2003


   provides the first sequence number of the next segment.

   Note that more than one SSRC MAY be sent in the media stream.          |
   However without further extensions RTSP can't synchronize more than    |
   the single one indicated in the Transport header. In these cases RTCP  |
   needs to be used for synchronization.                                  |

   The transmission of RTCP SHOULD be done the whole time from RTP        |
   session creation by the SETUP request and continue until the session   |
   is removed by the TEARDOWN request, that is including during the       |
   period in Ready state. This ensures that neither end part times out    |
   the other. Thus ensuring liveness information to both end-points,      |
   allow for packet-loss detection at the end of playout period, ensure   |
   media synchronization in cases multiple SSRCs are used, and to keep    |
   synchronization information updated allowing for correct synch also    |
   at the beginning of a stream before any PLAY response has arrived.     |

   The sending of the RTCP BYE message is connected to the existence of   |
   the RTCP session and the SSRC. Therefore a client or server SHALL not  |
   send an RTCP BYE message until it has finished using an SSRC. A        |
   server SHOULD not stop using an SSRC until the RTP session is          |
   terminated. This is due to that a SSRC that has been used has an       |
   established synchronization context that ensures synchronization also  |
   if the PLAY response is late, for an subsequent PLAY request after     |
   the first one. Changing the server side SSRC will also prevent the     |
   server from synchronizing that new SSRC within RTSP as it is           |
   connected to the one declared in the SSRC parameter in the Transport   |
   header.

   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
   different lower layer transport protocols are defined below in
   regards to RTSP.

   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.




H. Schulzrinne et. al.                                      [Page 118]


Internet Draft                    RTSP                  October 27, 2003


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
   Conferences with Minimal Control" defined in RFC 3551 [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
   transported 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",
   "destination", "client_port", and "server_port". This has the
   advantage of being compatible with all RTP capable RTSP servers and
   clients. However this method does not provide a possibility to
   specify non-continues 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:

        o 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.

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

        o If there is specified a range in "client_port" parameter that
          contains at least two port numbers, the RTCP/UDP packets from
          server to client SHALL be sent to address specified in the
          "destination" parameter and first odd port number part of the
          range specified in the client_port parameter.

        o The Server SHOULD send its RTCP/UDP packets from the address
          specified in "source" parameter and from the first odd port
          number specified in "server_port" parameter.

        o RTCP/UDP packets from the client to the server SHALL be sent



H. Schulzrinne et. al.                                      [Page 119]


Internet Draft                    RTSP                  October 27, 2003


          to the address specified in the "source" parameter and first
          odd port number given in client_port range.

        o 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
   specify the address and port numbers are done in the following way
   for media playback, i.e. Mode=PLAY:

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

        o Each address and port pair MUST contain both and address and a
          port number.

        o The first address and port pair given in either of the
          parameters applies to the RTP stream. The second address and
          port pair if present applies to the RTCP stream.

        o The RTP/UDP packets from the server to the client SHALL be
          sent to the address and port given by first address and port
          pair of the "dst_addresses" parameter.

        o The RTCP/UDP packets from the server to the client SHALL be
          sent to the address and port given by the second address and
          port pair of the "dst_addresses" parameter. If no second pair
          is given RTCP SHALL NOT be sent.

        o The RTCP/UDP packets from the client to the server SHALL be
          sent to the address and port given by the second address and
          port pair of the "dst_addresses" parameter. If no second pair
          is given RTCP SHALL NOT be sent.

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

B.1.3 AVP/TCP

   Note that this combination is not yet defined using sperate TCP
   connections. However the use of embedded (interleaved) binary data
   transported on the RTSP connection is possible as specified in
   section  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



H. Schulzrinne et. al.                                      [Page 120]


Internet Draft                    RTSP                  October 27, 2003


   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
   protocol of profile for use with RTSP:

        o The protocol or profile needs to define a name tag
          representing it. This tag is required to be a ABNF "token" to
          be possible to use in the Transport header specification.

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

        o For new media protocols the interaction with RTSP needs to be
          addressed. One important factor will be the media
          synchronization.

   See the IANA section ( 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         |
   typically returned in reply to a DESCRIBE request on a URL from a      |
   server to a client, or received via HTTP from a server to a client.    |

   This appendix describes how an SDP file determines the operation of    |
   an RTSP session. SDP as is provides no mechanism by which a client     |
   can distinguish, without human guidance, between several media         |
   streams to be rendered simultaneously and a set of alternatives        |
   (e.g., two audio streams spoken in different languages). However the   |
   SDP extension "Grouping of Media Lines in the Session Description      |
   Protocol (SDP)" [40] may provide such functionality depending on       |
   need. Also future grouping semantics may in the future be developed.

C.1 Definitions

   The terms "session-level", "media-level" and other key/attribute



H. Schulzrinne et. al.                                      [Page 121]


Internet Draft                    RTSP                  October 27, 2003


   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         |
   controlling that particular media stream.  If found at the session     |
   level, the attribute indicates the URL for aggregate control           |
   (presentation URL). The session level URL SHALL be different from any  |
   media level URI. The presence of a session level control attribute     |
   SHALL be interpreted as support for aggregated control. The control    |
   attribute SHALL be present on media level unless the presentation      |
   only contains a single media stream, in which case the attribute MAY   |
   only be present on the session level.


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


   Example:


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



   This attribute MAY contain either relative and absolute URLs,
   following the rules and conventions set out in RFC 2396 [22].
   Implementations SHALL 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 SHALL    |
   be treated as if it were an empty embedded URL, and thus inherit 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  |



H. Schulzrinne et. al.                                      [Page 122]


Internet Draft                    RTSP                  October 27, 2003


   that the base URL for the SDP document becomes:                        |
   "rtsp://example.com/container.mp4/", i.e. an extra trailing slash.     |
   When using the URL resolution rules in RFC 2396 that will resolve      |
   correctly. However, please note that 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            |
   synchronization. If the session is a multicast, the port number        |
   indicated SHOULD be used for reception. The client MAY try to          |
   override the destination port, through the Transport header.  The      |
   servers MAY allow this, the response will indicate if allowed or not.  |
   If the session is unicast, the port number is the ones RECOMMENDED by  |
   the server to the client, about which receiver ports to use; the       |
   client MUST still include its receiver ports in its SETUP request.     |
   The client MAY ignore this recommendation. If the server has no        |
   preference, it SHOULD set the port number value to zero.               |

   The "m=" lines contain information about what transport protocol,      |
   profile, and possibly lower-layer shall be used for the media stream.  |
   The combination of transport, profile and lower layer, like            |
   RTP/AVP/UDP needs to be defined for how to be used with RTSP.  The     |
   currently defined combinations are defined in section B, further       |
   combinations MAY be specified.                                         |

   TODO: Write something about the usage of Grouping of media line, RFC   |
   3388 [40].                                                             |


   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       |
   payload type is a static payload type from RFC 3551 [1], no other      |
   information is required. In case it is a dynamic payload type, the     |
   media attribute "rtpmap" is used to specify what the media is. The     |
   "encoding name" within the "rtpmap" attribute may be one of those      |
   specified in RFC 3551 (Sections 5 and 6), or an MIME type registered   |
   with IANA, or an experimental encoding with a "X-" prefix as           |



H. Schulzrinne et. al.                                      [Page 123]


Internet Draft                    RTSP                  October 27, 2003


   specified in SDP (RFC 2327 [24]). Codec-specific parameters are not    |
   specified in this field, but rather in the "fmtp" attribute described  |
   below.                                                                 |

C.1.4 Format-Specific Parameters                                          |

   Format-specific parameters are conveyed using the "fmtp" media         |
   attribute. The syntax of the "fmtp" attribute is specific to the       |
   encoding(s) that the attribute refers to. Note that some of the        |
   format specific parameters may be specified outside of the fmtp        |
   parameters, like for example the "ptime" attribute for most audio      |
   encodings.                                                             |

C.1.5 Range of Presentation                                               |

   The "a=range" attribute defines the total time range of the stored     |
   session or an individual media. Non-seekable Live sessions can be      |
   indicated, while the length of live sessions can be deduced from the   |
   "t" and "r" SDP parameters.                                            |

   The attribute is both a session and a media level attribute. For       |
   presentations that contains media streams of the same durations, the   |
   range attribute SHOULD only be used at session-level. In case of       |
   different length the range attribute MUST be given at media level for  |
   all media, and SHOULD NOT be given at session level. If the attribute  |
   is present at both media level and session level the media level       |
   values SHALL be used.

   The unit is specified first, followed by the value range. The units
   and their values are as defined in Section 3.4, 3.5 and 3.6. Any open
   ended range (start-), i.e. without stop range, is of unspecified
   duration and SHALL be considered as non-seekable content unless this
   property is overridden.

   This attribute is defined in ABNF [14] as:

   a-range-def = "a" "=" "range" ":" ranges-specifier CRLF


   Examples:


     a=range:npt=0-34.4368
     a=range:clock=19971113T2115-19971113T2203
     Non seekable stream of unknown duration:
     a=range:npt=0-





H. Schulzrinne et. al.                                      [Page 124]


Internet Draft                    RTSP                  October 27, 2003


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.  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.

C.1.7 Connection Information

   In SDP, the "c=" field contains the destination address for the media  |
   stream. For a media destination address that is a IPv6 one, the SDP    |
   extension defined in [38] needs to be used. For on-demand unicast      |
   streams and some multicast streams, the destination address MAY be     |
   specified by the client via the SETUP request, thus overriding any     |
   specified address. To identify streams without a fixed destination     |
   address, where the client must specify a destination address, the      |
   "c=" field SHOULD be set to a null value. For addresses of type        |
   "IP4", this value SHALL be "0.0.0.0", and for type "IP6", this value   |
   SHALL be "0:0:0:0:0:0:0:0", i.e. the unspecified address according to  |
   RFC 3513 [39].

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.

   Example:


     a=etag:158bb3e7c7fd62ce67f12b533f06b83a




        One could argue that the "o=" field provides identical
        functionality. However, it does so in a manner that would
        put constraints on servers that need to support multiple
        session description types other than SDP for the same piece
        of media content.

C.2 Aggregate Control Not Available



H. Schulzrinne et. al.                                      [Page 125]


Internet Draft                    RTSP                  October 27, 2003


   If a presentation does not support aggregate control no session level  |
   "a=control:" attribute is specified. For a SDP with multiple media     |
   sections specified, each section will have its own control 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
   initialization information even if it is delivered to the media
   client through non-RTSP means. This is necessary as there is no
   mechanism to indicate that the client should request more detailed
   media stream information via DESCRIBE.

C.3 Aggregate Control Available

   In this scenario, the server has multiple streams that can be
   controlled as a whole. In this case, there are both a media-level
   "a=control:" attributes, which are used to specify the stream URLs,
   and a session-level "a=control:" attribute which is used as the
   request URL for aggregate control. If the media-level URL is
   relative, it is resolved to absolute URLs according to Section C.1.1
   above.

   Example:                                                               |



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




H. Schulzrinne et. al.                                      [Page 126]


Internet Draft                    RTSP                  October 27, 2003


   M->C: RTSP/1.0 200 OK                                                  |
         CSeq: 1                                                          |
         Date: 23 Jan 1997 15:35:06 GMT                                   |
         Content-Type: application/sdp                                    |
         Content-Base: rtsp://example.com/movie/                          |
         Content-Length: 164                                              |

         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:*                                                      |
         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://example.com/movie/trackID=1 and                                 |
   rtsp://example.com/movie/trackID=2 to set up the video and audio       |
   streams, respectively. The URL rtsp://example.com/movie/ , which is    |
   resolved from the "*", controls the whole presentation (movie).        |

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

C.4 RTSP external SDP delivery                                            |

   There are some considerations that needs to be made when the session   |
   description is delivered to client outside of RTSP, for example in     |
   HTTP or email.                                                         |

   First of all the SDP needs to contain absolute URIs, relative will in  |
   most cases not work as the delivery will not correctly forward the     |
   base URI. And as SDP might be temporarily stored on file system        |
   before being loaded into a RTSP capable client, thus if possible to    |
   transport the base URI it still would need to be merged into the       |
   file.                                                                  |

   The writing of the SDP session availability information, i.e. "t="     |
   and "r=", needs to be carefully considered. When the SDP is fetched    |



H. Schulzrinne et. al.                                      [Page 127]


Internet Draft                    RTSP                  October 27, 2003


   by the DESCRIBE method it is with very high probability that the it    |
   is valid. However the same are much less certain for SDPs distributed  |
   using other methods. Therefore the publisher of the SDP should take    |
   care to follow the recommendations about availability in the SDP       |
   specification [24].

D Minimal RTSP implementation

D.1 Client

   A client implementation MUST be able to do the following :

        o Generate the following requests: SETUP, TEARDOWN, PLAY.

        o Include the following headers in requests: CSeq, Connection,
          Session, Transport.

        o Parse and understand the following headers in responses:
          CSeq, Connection, Session, Transport, Content-Language,
          Content-Encoding, Content-Length, Content-Type.

        o 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.

        o Expect and respond to asynchronous requests from the server,
          such as REDIRECT. This does not necessarily mean that it
          should implement 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.

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

        o Inclusion of the User-Agent header.

        o Understand SDP session descriptions as defined in Appendix C

        o 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 applications (such as web browsers).


        There may be RTSP applications different from those



H. Schulzrinne et. al.                                      [Page 128]


Internet Draft                    RTSP                  October 27, 2003


        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
   additionally be able to do the following:

        o generate the PAUSE request;

        o 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
   following:

        o recognize the 401 (Unauthorized) status code;

        o parse and include the WWW-Authenticate header;

        o implement Basic Authentication and Digest Authentication.

D.2 Server

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

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

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

        o 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
   implementations and/or to be a "good citizen".

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



H. Schulzrinne et. al.                                      [Page 129]


Internet Draft                    RTSP                  October 27, 2003


        o Inclusion of the Server header.

        o Implement the DESCRIBE method.

        o 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
   additionally be able to do the following:

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

        o 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:

        o Include and parse the Range header, with NPT units.
          Implementation of SMPTE units is recommended.

        o Include the length of the media presentation in the media
          initialization information.

        o 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
        information 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 indicator and a timeline
        positioning tool.

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



H. Schulzrinne et. al.                                      [Page 130]


Internet Draft                    RTSP                  October 27, 2003


D.2.2 Authentication-enabled

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

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

        o Parse and include the WWW-Authenticate header

        o Implement Basic Authentication and Digest Authentication

E Open Issues

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

        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



H. Schulzrinne et. al.                                      [Page 131]


Internet Draft                    RTSP                  October 27, 2003


   Compared to RFC 2326, the following issues are addressed:

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

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

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

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

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

        o 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.

        o 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.

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

        o 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.

        o 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 clarifications:

          - Content-Encoding header can include encoding of type
            "identity".

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

        o http://rtsp.org/bug513753 - Created a IANA section defining
          several registries.




H. Schulzrinne et. al.                                      [Page 132]


Internet Draft                    RTSP                  October 27, 2003


        o http://rtsp.org/bug477427 - A new subsection in the
          connections chapter clarifying how the server and client may
          handle transport connections.

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

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

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

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

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

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

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

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

        o Added status code 350, 351 and updated usage of the other
          redirect status codes, see chapter  12.3.

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

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

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

        o When sending response 451 and 458 the response body should
          contain the offending parameters.

        o 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.




H. Schulzrinne et. al.                                      [Page 133]


Internet Draft                    RTSP                  October 27, 2003


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

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

        o Added requirement that the Date header must be used for all
          messages with entity. Also the Server should always include
          it.

        o 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.

        o 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.

        o The Supported header was borrowed from SIP to help with the
          feature negotiation in RTSP.

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

        o 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
          interoperability issue is solved by requiring a client to know
          that a server supports this specification.

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

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

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

        o 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.

        o 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.



H. Schulzrinne et. al.                                      [Page 134]


Internet Draft                    RTSP                  October 27, 2003


        o 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.

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

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

        o 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.

        o 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 the SETUP.

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

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

        o 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.

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

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

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

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



H. Schulzrinne et. al.                                      [Page 135]


Internet Draft                    RTSP                  October 27, 2003


          streams.

        o Clarified the usage of "a=range" and how to indicate live
          content that are not seekable with this header.

        o Depreciated the use of the Range header "time=" parameter due
          to synchronization problems in PLAY and PAUSE methods.

   Note that this list does not reflect minor changes in wording or
   correction 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
   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

   Aravind Narasimhan
   Sun Microsystems, Inc.
   101 Park Avenue, 3rd & 4th Floor
   New York, NY
   USA



H. Schulzrinne et. al.                                      [Page 136]


Internet Draft                    RTSP                  October 27, 2003


   electronic mail: aravind.narasimhan@sun.com

H Contributors

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

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

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

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
   mentioned, the following individuals have contributed to this
   specification:

   Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
   Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
   Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
   Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
   John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,
   Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
   Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
   Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
   Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,
   Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
   Chesire, David Walker, and Geetha Srikantan.

   [1] H. Schulzrinne, "RTP profile for audio and video conferences with
   minimal control," RFC 3351, Internet Engineering Task Force, July
   2003.

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

   [3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
   "Internationalization of the hypertext markup language," RFC 2070,
   Internet Engineering Task Force, Jan.  1997.




H. Schulzrinne et. al.                                      [Page 137]


Internet Draft                    RTSP                  October 27, 2003


   [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
   pictures and associated audio informaiton -- part 6: extension for
   digital storage media and control," Draft International Standard ISO
   13818-6, International Organization for Standardization ISO/IEC
   JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.

   [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
   protocol (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.

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

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

   [14] D. Crocker and P. Overell, "Augmented BNF for syntax
   specifications:  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.



H. Schulzrinne et. al.                                      [Page 138]


Internet Draft                    RTSP                  October 27, 2003


   [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 3550, Internet
   Engineering Task Force, July 2003.

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

   [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
   Standarization Sector of ITU, Geneva, Switzerland, May 1996.

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



H. Schulzrinne et. al.                                      [Page 139]


Internet Draft                    RTSP                  October 27, 2003


   [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
   Engineering Task Force, Work in Progress, October 2002.

   [32] P. Srisuresh, K. Egevang, "Traditional IP Network Address
   Translator (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
   Firewalls.", 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.

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

   [38] S. Olson, G. Camarill, A. B. Roach, "Support for IPv6 in Session
   Description Protocol (SDP)," RFC 3266, Internet Engineering Task
   Force, June 2002.

   [39] R. Hinden, S. Deering, "Internet Protocol Version 6 (IPv6)
   Addressing Architecture," RFC 3513, Internet Engineering Task Force,
   April 2003.

   [40] G. Camarillo, G. Eriksson, J. Holler, H. Schulzrinne, "Grouping
   of Media Lines in the Session Description Protocol (SDP)," RFC 3388,
   Internet Engineering Task Force, December 2002.







H. Schulzrinne et. al.                                      [Page 140]


Internet Draft                    RTSP                  October 27, 2003


   IPR Notice

   The IETF takes no position regarding the validity or scope of any
   intellectual property or other rights that might be claimed to
   pertain to the implementation or use of the technology described in
   this document or the extent to which any license under such rights
   might or might not be available; neither does it represent that it
   has made any effort to identify any such rights. Information on the
   IETF's procedures with respect to rights in standards-track and
   standards-related documentation can be found in BCP-11. Copies of
   claims of rights made available for publication and any assurances of
   licenses to be made available, or the result of an attempt made to
   obtain a general license or permission for the use of such
   proprietary rights by implementors or users of this specification can
   be obtained from the IETF Secretariat.

   The IETF invites any interested party to bring to its attention any
   copyrights, patents or patent applications, or other proprietary
   rights which may cover technology that may be required to practice
   this standard. Please address the information to the IETF Executive
   Director.


   Full Copyright Statement

   Copyright (C) The Internet Society (2003). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implmentation may be prepared, copied, published and
   distributed, in whole or in part, without restriction of any kind,
   provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of
   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

   The limited permissions granted above are perpetual and will not be
   revoked by the Internet Society or its successors or assigns.

   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
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION



H. Schulzrinne et. al.                                      [Page 141]


Internet Draft                    RTSP                  October 27, 2003


   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

















































H. Schulzrinne et. al.                                      [Page 142]


                           Table of Contents



   1          Introduction ........................................    2
   1.1        The Update of the RTSP Specification ................    2
   1.2        Purpose .............................................    3
   1.3        Requirements ........................................    5
   1.4        Terminology .........................................    5
   1.5        Protocol Properties .................................    8
   1.6        Extending RTSP ......................................    9
   1.7        Overall Operation ...................................   10
   1.8        RTSP States .........................................   11
   1.9        Relationship with Other Protocols ...................   12
   2          Notational Conventions ..............................   13
   3          Protocol Parameters .................................   13
   3.1        RTSP Version ........................................   13
   3.2        RTSP URL ............................................   13
   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 ........................................   17
   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 ...............................   19
   6          Request .............................................   20
   6.1        Request Line ........................................   20
   6.2        Request Header Fields ...............................   20
   7          Response ............................................   22
   7.1        Status-Line .........................................   22
   7.1.1      Status Code and Reason Phrase .......................   22
   7.1.2      Response Header Fields ..............................   24
   8          Entity ..............................................   25
   8.1        Entity Header Fields ................................   25
   8.2        Entity Body .........................................   27
   9          Connections .........................................   27
   9.1        Pipelining ..........................................   27
   9.2        Reliability and Acknowledgements ....................   28  |
   9.3        Unreliable Transport ................................   28  |
   9.4        The usage of connections ............................   29
   9.5        Use of IPv6 .........................................   31



H. Schulzrinne et. al.                                      [Page 143]


Internet Draft                    RTSP                  October 27, 2003


   10         Capability Handling .................................   31
   11         Method Definitions ..................................   32
   11.1       OPTIONS .............................................   33
   11.2       DESCRIBE ............................................   34
   11.3       SETUP ...............................................   36
   11.4       PLAY ................................................   39
   11.5       PAUSE ...............................................   42
   11.6       TEARDOWN ............................................   46
   11.7       GET_PARAMETER .......................................   47
   11.8       SET_PARAMETER .......................................   47
   11.9       REDIRECT ............................................   48
   11.10      PING ................................................   50
   11.11      Embedded (Interleaved) Binary Data ..................   51
   12         Status Code Definitions .............................   53
   12.1       Success 1xx .........................................   53
   12.1.1     100 Continue ........................................   53
   12.2       Success 2xx .........................................   53
   12.2.1     250 Low on Storage Space ............................   53
   12.3       Redirection 3xx .....................................   53
   12.3.1     TBW .................................................   54
   12.3.2     301 Moved Permanently ...............................   54
   12.3.3     302 Found ...........................................   54
   12.3.4     303 See Other .......................................   54
   12.3.5     304 Not Modified ....................................   54
   12.3.6     305 Use Proxy .......................................   55
   12.4       Client Error 4xx ....................................   55
   12.4.1     400 Bad Request .....................................   55
   12.4.2     405 Method Not Allowed ..............................   55
   12.4.3     451 Parameter Not Understood ........................   55
   12.4.4     452 reserved ........................................   55
   12.4.5     453 Not Enough Bandwidth ............................   55
   12.4.6     454 Session Not Found ...............................   55
   12.4.7     455 Method Not Valid in This State ..................   55
   12.4.8     456 Header Field Not Valid for Resource .............   56
   12.4.9     457 Invalid Range ...................................   56
   12.4.10    458 Parameter Is Read-Only ..........................   56
   12.4.11    459 Aggregate Operation Not Allowed .................   56
   12.4.12    460 Only Aggregate Operation Allowed ................   56
   12.4.13    461 Unsupported Transport ...........................   56
   12.4.14    462 Destination Unreachable .........................   56
   12.5       Server Error 5xx ....................................   56
   12.5.1     551 Option not supported ............................   57
   13         Header Field Definitions ............................   57
   13.1       Accept ..............................................   59
   13.2       Accept-Encoding .....................................   59
   13.3       Accept-Language .....................................   59
   13.4       Accept-Ranges .......................................   59
   13.5       Allow ...............................................   60



H. Schulzrinne et. al.                                      [Page 144]


Internet Draft                    RTSP                  October 27, 2003


   13.6       Authorization .......................................   60
   13.7       Bandwidth ...........................................   60
   13.8       Blocksize ...........................................   62
   13.9       Cache-Control .......................................   64
   13.10      Connection ..........................................   67
   13.11      Content-Base ........................................   67
   13.12      Content-Encoding ....................................   67
   13.13      Content-Language ....................................   67
   13.14      Content-Length ......................................   67
   13.15      Content-Location ....................................   68
   13.16      Content-Type ........................................   68
   13.17      CSeq ................................................   68
   13.18      Date ................................................   68
   13.19      Expires .............................................   68
   13.20      From ................................................   69
   13.21      Host ................................................   69
   13.22      If-Match ............................................   70
   13.23      If-Modified-Since ...................................   70
   13.24      Last-Modified .......................................   70
   13.25      Location ............................................   70
   13.26      Proxy-Authenticate ..................................   70
   13.27      Proxy-Require .......................................   71
   13.28      Public ..............................................   71
   13.29      Range ...............................................   72
   13.30      Referer .............................................   73
   13.31      Retry-After .........................................   74
   13.32      Require .............................................   74
   13.33      RTP-Info ............................................   75
   13.34      Scale ...............................................   77
   13.35      Speed ...............................................   78
   13.36      Server ..............................................   78
   13.37      Session .............................................   78
   13.38      Supported ...........................................   81
   13.39      Timestamp ...........................................   81
   13.40      Transport ...........................................   82
   13.41      Unsupported .........................................   88
   13.42      User-Agent ..........................................   88
   13.43      Vary ................................................   88
   13.44      Via .................................................   88
   13.45      WWW-Authenticate ....................................   88
   14         Caching .............................................   88
   15         Examples ............................................   89
   15.1       Media on Demand (Unicast) ...........................   89
   15.2       Streaming of a Container file .......................   92
   15.3       Single Stream Container Files .......................   94
   15.4       Live Media Presentation Using Multicast .............   96
   16         Syntax ..............................................   97
   16.1       Base Syntax .........................................   97



H. Schulzrinne et. al.                                      [Page 145]


Internet Draft                    RTSP                  October 27, 2003


   16.2       RTSP Protocol Definition ............................   98
   16.2.1     Message Syntax ......................................   98
   16.2.2     Header Syntax .......................................  102
   17         Security Considerations .............................  103
   18         IANA Considerations .................................  105
   18.1       Feature-tags ........................................  106
   18.1.1     Description .........................................  106
   18.1.2     Registering New Feature-tags with IANA ..............  106
   18.1.3     Registered entries ..................................  106
   18.2       RTSP Methods ........................................  107
   18.2.1     Description .........................................  107
   18.2.2     Registering New Methods with IANA ...................  107
   18.2.3     Registered Entries ..................................  107
   18.3       RTSP Status Codes ...................................  107
   18.3.1     Description .........................................  107
   18.3.2     Registering New Status Codes with IANA ..............  108
   18.3.3     Registered Entries ..................................  108
   18.4       RTSP Headers ........................................  108
   18.4.1     Description .........................................  108
   18.4.2     Registering New Headers with IANA ...................  108
   18.4.3     Registered entries ..................................  109
   18.5       Transport Header registries .........................  109
   18.5.1     Transport Protocols .................................  109
   18.5.2     Profile .............................................  110
   18.5.3     Lower Transport .....................................  110
   18.5.4     Transport modes .....................................  110
   18.6       Cache Directive Extensions ..........................  111
   18.7       SDP attributes ......................................  111
   A          RTSP Protocol State Machine .........................  112
   A.1        States ..............................................  112  |
   A.2        State variables .....................................  112  |
   A.3        Abbreviations .......................................  113  |
   A.4        State Tables ........................................  113  |
   B          Media Transport Alternatives ........................  116
   B.1        RTP .................................................  117
   B.1.1      AVP .................................................  118
   B.1.2      AVP/UDP .............................................  119
   B.1.3      AVP/TCP .............................................  120
   B.2        Future Additions ....................................  121
   C          Use of SDP for RTSP Session Descriptions ............  121
   C.1        Definitions .........................................  121
   C.1.1      Control URL .........................................  122
   C.1.2      Media Streams .......................................  123  |
   C.1.3      Payload Type(s) .....................................  123
   C.1.4      Format-Specific Parameters ..........................  124  |
   C.1.5      Range of Presentation ...............................  124  |
   C.1.6      Time of Availability ................................  125
   C.1.7      Connection Information ..............................  125



H. Schulzrinne et. al.                                      [Page 146]


Internet Draft                    RTSP                  October 27, 2003


   C.1.8      Entity Tag ..........................................  125
   C.2        Aggregate Control Not Available .....................  125
   C.3        Aggregate Control Available .........................  126
   C.4        RTSP external SDP delivery ..........................  127  |
   D          Minimal RTSP implementation .........................  128
   D.1        Client ..............................................  128
   D.1.1      Basic Playback ......................................  129
   D.1.2      Authentication-enabled ..............................  129
   D.2        Server ..............................................  129
   D.2.1      Basic Playback ......................................  130
   D.2.2      Authentication-enabled ..............................  131
   E          Open Issues .........................................  131
   F          Changes .............................................  131
   G          Author Addresses ....................................  136
   H          Contributors ........................................  137
   I          Acknowledgements ....................................  137



































H. Schulzrinne et. al.                                      [Page 147]


Html markup produced by rfcmarkup 1.129b, available from https://tools.ietf.org/tools/rfcmarkup/