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RFC 2326


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Real Time Streaming Protocol (RTSP)

Part 1 of 4, p. 1 to 19
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Network Working Group                                     H. Schulzrinne
Request for Comments: 2326                                   Columbia U.
Category: Standards Track                                         A. Rao
                                                                Netscape
                                                             R. Lanphier
                                                            RealNetworks
                                                              April 1998

                  Real Time Streaming Protocol (RTSP)

Status of this Memo

   This document specifies an Internet standards track protocol for the
   Internet community, and requests discussion and suggestions for
   improvements.  Please refer to the current edition of the "Internet
   Official Protocol Standards" (STD 1) for the standardization state
   and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

   Copyright (C) The Internet Society (1998).  All Rights Reserved.

Abstract

   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 1889).

Table of Contents

   * 1 Introduction .................................................  5
        + 1.1 Purpose ...............................................  5
        + 1.2 Requirements ..........................................  6
        + 1.3 Terminology ...........................................  6
        + 1.4 Protocol Properties ...................................  9
        + 1.5 Extending RTSP ........................................ 11
        + 1.6 Overall Operation ..................................... 11
        + 1.7 RTSP States ........................................... 12
        + 1.8 Relationship with Other Protocols ..................... 13
   * 2 Notational Conventions ....................................... 14
   * 3 Protocol Parameters .......................................... 14
        + 3.1 RTSP Version .......................................... 14

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        + 3.2 RTSP URL .............................................. 14
        + 3.3 Conference Identifiers ................................ 16
        + 3.4 Session Identifiers ................................... 16
        + 3.5 SMPTE Relative Timestamps ............................. 16
        + 3.6 Normal Play Time ...................................... 17
        + 3.7 Absolute Time ......................................... 18
        + 3.8 Option Tags ........................................... 18
             o 3.8.1 Registering New Option Tags with IANA .......... 18
   * 4 RTSP Message ................................................. 19
        + 4.1 Message Types ......................................... 19
        + 4.2 Message Headers ....................................... 19
        + 4.3 Message Body .......................................... 19
        + 4.4 Message Length ........................................ 20
   * 5 General Header Fields ........................................ 20
   * 6 Request ...................................................... 20
        + 6.1 Request Line .......................................... 21
        + 6.2 Request Header Fields ................................. 21
   * 7 Response ..................................................... 22
        + 7.1 Status-Line ........................................... 22
             o 7.1.1 Status Code and Reason Phrase .................. 22
             o 7.1.2 Response Header Fields ......................... 26
   * 8 Entity ....................................................... 27
        + 8.1 Entity Header Fields .................................. 27
        + 8.2 Entity Body ........................................... 28
   * 9 Connections .................................................. 28
        + 9.1 Pipelining ............................................ 28
        + 9.2 Reliability and Acknowledgements ...................... 28
   * 10 Method Definitions .......................................... 29
        + 10.1 OPTIONS .............................................. 30
        + 10.2 DESCRIBE ............................................. 31
        + 10.3 ANNOUNCE ............................................. 32
        + 10.4 SETUP ................................................ 33
        + 10.5 PLAY ................................................. 34
        + 10.6 PAUSE ................................................ 36
        + 10.7 TEARDOWN ............................................. 37
        + 10.8 GET_PARAMETER ........................................ 37
        + 10.9 SET_PARAMETER ........................................ 38
        + 10.10 REDIRECT ............................................ 39
        + 10.11 RECORD .............................................. 39
        + 10.12 Embedded (Interleaved) Binary Data .................. 40
   * 11 Status Code Definitions ..................................... 41
        + 11.1 Success 2xx .......................................... 41
             o 11.1.1 250 Low on Storage Space ...................... 41
        + 11.2 Redirection 3xx ...................................... 41
        + 11.3 Client Error 4xx ..................................... 42
             o 11.3.1 405 Method Not Allowed ........................ 42
             o 11.3.2 451 Parameter Not Understood .................. 42
             o 11.3.3 452 Conference Not Found ...................... 42

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             o 11.3.4 453 Not Enough Bandwidth ...................... 42
             o 11.3.5 454 Session Not Found ......................... 42
             o 11.3.6 455 Method Not Valid in This State ............ 42
             o 11.3.7 456 Header Field Not Valid for Resource ....... 42
             o 11.3.8 457 Invalid Range ............................. 43
             o 11.3.9 458 Parameter Is Read-Only .................... 43
             o 11.3.10 459 Aggregate Operation Not Allowed .......... 43
             o 11.3.11 460 Only Aggregate Operation Allowed ......... 43
             o 11.3.12 461 Unsupported Transport .................... 43
             o 11.3.13 462 Destination Unreachable .................. 43
             o 11.3.14 551 Option not supported ..................... 43
   * 12 Header Field Definitions .................................... 44
        + 12.1 Accept ............................................... 46
        + 12.2 Accept-Encoding ...................................... 46
        + 12.3 Accept-Language ...................................... 46
        + 12.4 Allow ................................................ 46
        + 12.5 Authorization ........................................ 46
        + 12.6 Bandwidth ............................................ 46
        + 12.7 Blocksize ............................................ 47
        + 12.8 Cache-Control ........................................ 47
        + 12.9 Conference ........................................... 49
        + 12.10 Connection .......................................... 49
        + 12.11 Content-Base ........................................ 49
        + 12.12 Content-Encoding .................................... 49
        + 12.13 Content-Language .................................... 50
        + 12.14 Content-Length ...................................... 50
        + 12.15 Content-Location .................................... 50
        + 12.16 Content-Type ........................................ 50
        + 12.17 CSeq ................................................ 50
        + 12.18 Date ................................................ 50
        + 12.19 Expires ............................................. 50
        + 12.20 From ................................................ 51
        + 12.21 Host ................................................ 51
        + 12.22 If-Match ............................................ 51
        + 12.23 If-Modified-Since ................................... 52
        + 12.24 Last-Modified........................................ 52
        + 12.25 Location ............................................ 52
        + 12.26 Proxy-Authenticate .................................. 52
        + 12.27 Proxy-Require ....................................... 52
        + 12.28 Public .............................................. 53
        + 12.29 Range ............................................... 53
        + 12.30 Referer ............................................. 54
        + 12.31 Retry-After ......................................... 54
        + 12.32 Require ............................................. 54
        + 12.33 RTP-Info ............................................ 55
        + 12.34 Scale ............................................... 56
        + 12.35 Speed ............................................... 57
        + 12.36 Server .............................................. 57

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        + 12.37 Session ............................................. 57
        + 12.38 Timestamp ........................................... 58
        + 12.39 Transport ........................................... 58
        + 12.40 Unsupported ......................................... 62
        + 12.41 User-Agent .......................................... 62
        + 12.42 Vary ................................................ 62
        + 12.43 Via ................................................. 62
        + 12.44 WWW-Authenticate .................................... 62
   * 13 Caching ..................................................... 62
   * 14 Examples .................................................... 63
        + 14.1 Media on Demand (Unicast) ............................ 63
        + 14.2 Streaming of a Container file ........................ 65
        + 14.3 Single Stream Container Files ........................ 67
        + 14.4 Live Media Presentation Using Multicast .............. 69
        + 14.5 Playing media into an existing session ............... 70
        + 14.6 Recording ............................................ 71
   * 15 Syntax ...................................................... 72
        + 15.1 Base Syntax .......................................... 72
   * 16 Security Considerations ..................................... 73
   * A. RTSP Protocol State Machines ................................ 76
        + A.1 Client State Machine .................................. 76
        + A.2 Server State Machine .................................. 77
   * B. Interaction with RTP ........................................ 79
   * C. Use of SDP for RTSP Session Descriptions .................... 80
        + C.1 Definitions ........................................... 80
             o C.1.1 Control URL .................................... 80
             o C.1.2 Media streams .................................. 81
             o C.1.3 Payload type(s) ................................ 81
             o C.1.4 Format-specific parameters ..................... 81
             o C.1.5 Range of presentation .......................... 82
             o C.1.6 Time of availability ........................... 82
             o C.1.7 Connection Information ......................... 82
             o C.1.8 Entity Tag ..................................... 82
        + C.2 Aggregate Control Not Available ....................... 83
        + C.3 Aggregate Control Available ........................... 83
   * D. Minimal RTSP implementation ................................. 85
        + D.1 Client ................................................ 85
             o D.1.1 Basic Playback ................................. 86
             o D.1.2 Authentication-enabled ......................... 86
        + D.2 Server ................................................ 86
             o D.2.1 Basic Playback ................................. 87
             o D.2.2 Authentication-enabled ......................... 87
   * E. Authors' Addresses .......................................... 88
   * F. Acknowledgements ............................................ 89
   * References ..................................................... 90
   * Full Copyright Statement ....................................... 92

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1 Introduction

1.1 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls
   either a single or several time-synchronized streams of continuous
   media such as audio and video. It does not typically deliver the
   continuous streams itself, although interleaving of the continuous
   media stream with the control stream is possible (see Section 10.12).
   In other words, RTSP acts as a "network remote control" for
   multimedia servers.

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

   There is no notion of an RTSP connection; instead, a server maintains
   a session labeled by an identifier. An RTSP session is in no way tied
   to a transport-level connection such as a TCP connection. During an
   RTSP session, an RTSP client may open and close many reliable
   transport connections to the server to issue RTSP requests.
   Alternatively, it may use a connectionless transport protocol such as
   UDP.

   The streams controlled by RTSP may use RTP [1], but the operation of
   RTSP does not depend on the transport mechanism used to carry
   continuous media.  The protocol is intentionally similar in syntax
   and operation to HTTP/1.1 [2] so that extension mechanisms to HTTP
   can in most cases also be added to RTSP. However, RTSP differs in a
   number of important aspects from HTTP:

     * RTSP introduces a number of new methods and has a different
       protocol identifier.
     * An RTSP server needs to maintain state by default in almost all
       cases, as opposed to the stateless nature of HTTP.
     * Both an RTSP server and client can issue requests.
     * Data is carried out-of-band by a different protocol. (There is an
       exception to this.)
     * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
       consistent with current HTML internationalization efforts [3].
     * The Request-URI always contains the absolute URI. Because of
       backward compatibility with a historical blunder, HTTP/1.1 [2]
       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.

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   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 be "invited" to join an existing
          conference, either to play back media into the presentation or
          to record all or a subset of the media in a 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 [2].

1.2 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].

1.3 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not
   listed here are defined as in HTTP/1.1.

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

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

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   Client:
          The client requests continuous media data from the media
          server.

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

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

   Continuous media:
          Data where there is a timing relationship between source and
          sink; that is, the sink must reproduce the timing relationship
          that existed at the source. The most common 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 metainformation in the form of
          entity-header fields and content in the form of an entity-
          body, as described in Section 8.

   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 or recording 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.

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   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 15 and transmitted via a connection or a
          connectionless protocol.

   Participant:
          Member of a conference. A participant may be a machine, e.g.,
          a media record or 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 [6]) 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:
          An RTSP response. If an HTTP response is meant, that is
          indicated explicitly.

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

   RTSP session:
          A complete RTSP "transaction", e.g., the viewing of a movie.
          A session typically consists of a client setting up a
          transport mechanism for the continuous media stream (SETUP),
          starting the stream with PLAY or RECORD, and closing the

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          stream with TEARDOWN.

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

1.4 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. All HTTP authentication
          mechanisms such as basic (RFC 2068 [2, Section 11.1]) and
          digest authentication (RFC 2069 [8]) are directly applicable.
          One may also reuse transport or network layer security
          mechanisms.

   Transport-independent:
          RTSP may use either an unreliable datagram protocol (UDP) (RFC
          768 [9]), a reliable datagram protocol (RDP, RFC 1151, not
          widely used [10]) or a reliable stream protocol such as TCP
          (RFC 793 [11]) as it implements application-level reliability.

   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.

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

   Separation of stream control and conference initiation:
          Stream control is divorced from inviting a media server to a
          conference. The only requirement is that the conference
          initiation protocol either provides or can be used to create a
          unique conference identifier. In particular, SIP [12] or H.323
          [13] may be used to invite a server to a conference.

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   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 [14]) 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
          [15,16]) 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

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     control mechanism.

1.5 Extending RTSP

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

     * A server may only be capable of playback thus has no need to
       support the RECORD request.
     * A server may not be capable of seeking (absolute positioning) if
       it is to support live events only.
     * Some servers may not support setting stream parameters and thus
       not support GET_PARAMETER and SET_PARAMETER.

   A server SHOULD implement all header fields described in Section 12.

   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 [2],
   where the methods described in [H19.6] 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:

     * 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 12.32).
     * New methods can be added. If the recipient of the message does
       not understand the request, it responds with error code 501 (Not
       implemented) and the sender should not attempt to use this method
       again. A client may also use the OPTIONS method to inquire about
       methods supported by the server. The server SHOULD list the
       methods it supports using the Public response header.
     * A new version of the protocol can be defined, allowing almost all
       aspects (except the position of the protocol version number) to
       change.

1.6 Overall Operation

   Each presentation and media stream may be identified by an 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

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

   The presentation description file contains a description of the media
   streams making up the presentation, including their encodings,
   language, and other parameters that enable the client to choose the
   most appropriate combination of media. In this presentation
   description, each media stream that is individually controllable by
   RTSP is identified by an RTSP 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.

1.7 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

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   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 Section A.

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

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

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

   PAUSE:
          Temporarily halts a stream without freeing server resources.

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

          RTSP methods that contribute to state use the Session header
          field (Section 12.37) to identify the RTSP session whose state
          is being manipulated. The server generates session identifiers
          in response to SETUP requests (Section 10.4).

1.8 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 data delivery
   takes place out-of-band in a different protocol. HTTP is an
   asymmetric protocol where the client issues requests and the server
   responds. In RTSP, both the media client and media server can issue
   requests. RTSP requests are also not stateless; they may set
   parameters and continue to control a media stream long after the

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   request has been acknowledged.

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

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

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

2 Notational Conventions

   Since many of the definitions and syntax are identical to HTTP/1.1,
   this specification only points to the section where they are defined
   rather than copying it. For brevity, [HX.Y] is to be taken to refer
   to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]).

   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 [17], with the
   difference that this RTSP specification maintains the "1#" notation
   for comma-separated lists.

   In this memo, 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

   [H3.1] applies, with HTTP replaced by RTSP.

3.2 RTSP URL

   The "rtsp" 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.

   rtsp_URL  =   ( "rtsp:" | "rtspu:" )
                 "//" host [ ":" port ] [ abs_path ]
   host      =   <A legal Internet host domain name of IP address
                 (in dotted decimal form), as defined by Section 2.1

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                 of RFC 1123 \cite{rfc1123}>
   port      =   *DIGIT

   abs_path is defined in [H3.2.1].

     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).

   If the port is empty or not given, port 554 is 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 [19]).

   A presentation or a stream is identified by a textual media
   identifier, using the character set and escape conventions [H3.2] of
   URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of
   streams, i.e., a presentation. Accordingly, requests described in
   Section 10 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

   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".

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   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 Conference Identifiers

   Conference identifiers are opaque to RTSP and are encoded using
   standard URI encoding methods (i.e., LWS is escaped with %). They can
   contain any octet value. The conference identifier MUST be globally
   unique. For H.323, the conferenceID value is to be used.

 conference-id =   1*xchar

     Conference identifiers are used to allow RTSP sessions to obtain
     parameters from multimedia conferences the media server is
     participating in. These conferences are created by protocols
     outside the scope of this specification, e.g., H.323 [13] or SIP
     [12]. Instead of the RTSP client explicitly providing transport
     information, for example, it asks the media server to use the
     values in the conference description instead.

3.4 Session Identifiers

   Session identifiers are opaque strings of arbitrary length. Linear
   white space must be URL-escaped. A session identifier MUST be chosen
   randomly and MUST be at least eight octets long to make guessing it
   more difficult. (See Section 16.)

     session-id   =   1*( ALPHA | DIGIT | safe )

3.5 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.

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

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

3.6 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position
   relative to the beginning of the presentation. 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 be used only for live events.

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

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

     The syntax conforms to ISO 8601. The npt-sec notation is optimized
     for automatic generation, the ntp-hhmmss notation for consumption
     by human readers. The "now" constant allows clients to request to

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     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.

3.7 Absolute Time

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

     utc-range    =   "clock" "=" utc-time "-" [ utc-time ]
     utc-time     =   utc-date "T" utc-time "Z"
     utc-date     =   8DIGIT                    ; < YYYYMMDD >
     utc-time     =   6DIGIT [ "." fraction ]   ; < HHMMSS.fraction >

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

     19961108T143720.25Z

3.8 Option Tags

   Option tags are unique identifiers used to designate new options in
   RTSP. These tags are used in Require (Section 12.32) and Proxy-
   Require (Section 12.27) header fields.

   Syntax:

     option-tag   =   1*xchar

   The creator of a new RTSP option should either prefix the option 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 option with the Internet Assigned Numbers Authority
   (IANA).

3.8.1 Registering New Option Tags with IANA

   When registering a new RTSP option, the following information should
   be provided:

     * Name and description of option. The name may be of any length,
       but SHOULD be no more than twenty characters long. The name MUST
       not contain any spaces, control characters or periods.
     * Indication of who has change control over the option (for
       example, IETF, ISO, ITU-T, other international standardization
       bodies, a consortium or a particular company or group of
       companies);

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     * A reference to a further description, if available, for example
       (in order of preference) an RFC, a published paper, a patent
       filing, a technical report, documented source code or a computer
       manual;
     * For proprietary options, contact information (postal and email
       address);



(page 19 continued on part 2)

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