    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);

    4 RTSP Message

       RTSP is a text-based protocol and uses the ISO 10646 character set in
       UTF-8 encoding (RFC 2279 [21]). 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 [21])

       RTSP messages can be carried over any lower-layer transport protocol
       that is 8-bit clean.

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

    4.1 Message Types

       See [H4.1]

    4.2 Message Headers

       See [H4.2]

    4.3 Message Body

       See [H4.3]

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    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 12.14) 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.

       3.     By the server closing the connection. (Closing the connection
              cannot be used to indicate the end of a request body, since
              that would leave no possibility for the server to send back a
              response.)

       Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
       transfer coding(see [H3.6]) 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. Even though Content-Length must be present if there is
         any entity body, the rules ensure reasonable behavior even if the
         length is not given explicitly.

    5 General Header Fields

       See [H4.5], except that Pragma, Transfer-Encoding and Upgrade headers
       are not defined:

          general-header     =     Cache-Control     ; Section 12.8
                             |     Connection        ; Section 12.10
                             |     Date              ; Section 12.18
                             |     Via               ; Section 12.43

    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.

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           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 10.2
                      |         "ANNOUNCE"              ; Section 10.3
                      |         "GET_PARAMETER"         ; Section 10.8
                      |         "OPTIONS"               ; Section 10.1
                      |         "PAUSE"                 ; Section 10.6
                      |         "PLAY"                  ; Section 10.5
                      |         "RECORD"                ; Section 10.11
                      |         "REDIRECT"              ; Section 10.10
                      |         "SETUP"                 ; Section 10.4
                      |         "SET_PARAMETER"         ; Section 10.9
                      |         "TEARDOWN"              ; Section 10.7
                      |         extension-method

      extension-method = token

      Request-URI = "*" | absolute_URI

      RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT

    6.2 Request Header Fields

      request-header  =          Accept                   ; Section 12.1
                      |          Accept-Encoding          ; Section 12.2
                      |          Accept-Language          ; Section 12.3
                      |          Authorization            ; Section 12.5
                      |          From                     ; Section 12.20
                      |          If-Modified-Since        ; Section 12.23
                      |          Range                    ; Section 12.29
                      |          Referer                  ; Section 12.30
                      |          User-Agent               ; Section 12.41

       Note that in contrast to HTTP/1.1 [2], RTSP requests always contain
       the absolute URL (that is, including the scheme, host and port)
       rather than just the absolute path.

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         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 itself, and is only
       allowed when the method used does not necessarily apply to a
       resource.  One example would be:

         OPTIONS * 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 11. 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.

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

         * 1xx: Informational - Request received, continuing process
         * 2xx: Success - The action was successfully received, understood,
           and accepted
         * 3xx: Redirection - Further action must be taken in order to
           complete the request
         * 4xx: Client Error - The request contains bad syntax or cannot be
           fulfilled
         * 5xx: Server Error - The server failed to fulfill an apparently
           valid request

       The individual values of the numeric status codes defined for
       RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
       presented below. The reason phrases listed here are only recommended
       - they may be replaced by local equivalents without affecting the
       protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and
       adds RTSP-specific status codes starting at x50 to avoid conflicts
       with newly defined HTTP status codes.

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

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       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.

       Code           reason

       100            Continue                         all

       200            OK                               all
       201            Created                          RECORD
       250            Low on Storage Space             RECORD

       300            Multiple Choices                 all
       301            Moved Permanently                all
       302            Moved Temporarily                all
       303            See Other                        all
       305            Use Proxy                        all

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       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            Invalid parameter                SETUP
       452            Illegal Conference Identifier    SETUP
       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
       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
       551            Option not support               all

          Table 1: Status codes and their usage with RTSP methods

    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
       the Status-Line. These header fields give information about the
       server and about further access to the resource identified by the
       Request-URI.

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       response-header  =     Location             ; Section 12.25
                        |     Proxy-Authenticate   ; Section 12.26
                        |     Public               ; Section 12.28
                        |     Retry-After          ; Section 12.31
                        |     Server               ; Section 12.36
                        |     Vary                 ; Section 12.42
                        |     WWW-Authenticate     ; Section 12.44

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

         entity-header       =    Allow               ; Section 12.4
                             |    Content-Base        ; Section 12.11
                             |    Content-Encoding    ; Section 12.12
                             |    Content-Language    ; Section 12.13
                             |    Content-Length      ; Section 12.14
                             |    Content-Location    ; Section 12.15
                             |    Content-Type        ; Section 12.16
                             |    Expires             ; Section 12.19
                             |    Last-Modified       ; Section 12.24
                             |    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.

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    8.2 Entity Body

       See [H7.2]

    9 Connections

       RTSP requests can be transmitted in several different ways:

         * persistent transport connections used for several
           request-response transactions;
         * one connection per request/response transaction;
         * connectionless mode.

       The type of transport connection is defined by the RTSP URI (Section
       3.2). For the scheme "rtsp", a persistent 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

       A client that supports persistent connections or connectionless mode
       MAY "pipeline" its requests (i.e., send multiple requests without
       waiting for each response). A server MUST send its responses to those
       requests in the same order that the requests were received.

    9.2 Reliability and Acknowledgements

       Requests are acknowledged by the receiver unless they are sent to a
       multicast group. If there is no acknowledgement, the sender may
       resend the same message after a timeout of one round-trip time (RTT).
       The round-trip time is estimated as in TCP (RFC 1123) [18], with an
       initial round-trip value of 500 ms. An implementation MAY cache the
       last RTT measurement as the initial value for future connections.

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

         If both the underlying reliable transport such as TCP and the RTSP
         application retransmit requests, it is possible that each packet
         loss results in two retransmissions. The receiver cannot typically
         take advantage of the application-layer retransmission since the

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         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.

         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) [22], can be beneficial.

       The Timestamp header (Section 12.38) is used to avoid the
       retransmission ambiguity problem [23, p. 301] and obviates the need
       for Karn's algorithm.

       Each request carries a sequence number in the CSeq header (Section
       12.17), which is incremented by one for each distinct request
       transmitted. 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).

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

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

    10 Method Definitions

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

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          method            direction        object     requirement
          DESCRIBE          C->S             P,S        recommended
          ANNOUNCE          C->S, S->C       P,S        optional
          GET_PARAMETER     C->S, S->C       P,S        optional
          OPTIONS           C->S, S->C       P,S        required
                                                        (S->C: optional)
          PAUSE             C->S             P,S        recommended
          PLAY              C->S             P,S        required
          RECORD            C->S             P,S        optional
          REDIRECT          S->C             P,S        optional
          SETUP             C->S             S          required
          SET_PARAMETER     C->S, S->C       P,S        optional
          TEARDOWN          C->S             P,S        required

          Table 2: Overview of RTSP methods, their direction, and what
          objects (P: presentation, S: stream) they operate on

       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.

    10.1 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 server state.

       Example:

         C->S:  OPTIONS * RTSP/1.0
                CSeq: 1
                Require: implicit-play
                Proxy-Require: gzipped-messages

         S->C:  RTSP/1.0 200 OK
                CSeq: 1
                Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE

       Note that these 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).

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    10.2 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
               Accept: application/sdp, application/rtsl, application/mheg

         S->C: RTSP/1.0 200 OK
               CSeq: 312
               Date: 23 Jan 1997 15:35:06 GMT
               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=whiteboard 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.

         Clear ground rules need to be established so that clients have an
         unambiguous means of knowing when to request media initialization
         information via DESCRIBE, and when not to. By forcing a DESCRIBE

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         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:

         * via RTSP's DESCRIBE method;
         * via some other protocol (HTTP, email attachment, etc.);
         * via the command line or standard input (thus working as a browser
           helper application launched with an SDP file or other media
           initialization format).

         In the interest of practical interoperability, it is highly
         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.

    10.3 ANNOUNCE

       The ANNOUNCE method serves two purposes:

       When sent from client to server, ANNOUNCE posts the description of a
       presentation or media object identified by the request URL to a
       server. When sent from server to client, ANNOUNCE updates the session
       description in real-time.

       If a new media stream is added to a presentation (e.g., during a live
       presentation), the whole presentation description should be sent
       again, rather than just the additional components, so that components
       can be deleted.

       Example:

         C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
               CSeq: 312
               Date: 23 Jan 1997 15:35:06 GMT
               Session: 47112344
               Content-Type: application/sdp
               Content-Length: 332

               v=0
               o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4

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               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

         S->C: RTSP/1.0 200 OK
               CSeq: 312

    10.4 SETUP

       The SETUP request for a URI specifies the transport mechanism to be
       used for the streamed media. A client can issue a SETUP request for a
       stream that is already playing 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". For the benefit of any
       intervening firewalls, a client must indicate the transport
       parameters even if it has no influence over these parameters, for
       example, where the server advertises a fixed 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.

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

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

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

       The server generates session identifiers in response to SETUP
       requests. If a SETUP request to a server includes a session
       identifier, the server MUST bundle this setup request into the

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       existing session or return error "459 Aggregate Operation Not
       Allowed" (see Section 11.3.10).

    10.5 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.

       The PLAY request positions the normal play time to the beginning of
       the range specified and delivers stream data until the end of the
       range is reached. PLAY requests may be pipelined (queued); a server
       MUST queue PLAY requests to be executed in order. That is, a PLAY
       request arriving while a previous PLAY request is still active is
       delayed until the first has been completed.

         This allows precise editing.

       For example, regardless of how closely spaced the two PLAY requests
       in the example below arrive, the server 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

         C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
               CSeq: 836
               Session: 12345678
               Range: npt=20-25

         C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
               CSeq: 837
               Session: 12345678
               Range: npt=30-

       See the description of the PAUSE request for further examples.

       A PLAY request without a Range header is legal. It starts playing a
       stream from the beginning unless the stream has been paused. If a
       stream has been paused via PAUSE, stream delivery resumes at the
       pause point. If a stream is playing, such a PLAY request causes no
       further action and can be used by the client to test server liveness.

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       The Range header may also contain a time parameter. This parameter
       specifies a time in UTC at which the playback should start. If the
       message is received after the specified time, playback is started
       immediately. The time parameter may be used to aid in synchronization
       of streams obtained from different sources.

       For a on-demand stream, the server replies 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 current position is returned in the reply. The unit of
       the range in the reply is the same as that in the request.

       After playing the desired range, the presentation is automatically
       paused, as if a PAUSE request had been issued.

       The following example plays the whole presentation starting at SMPTE
       time code 0:10:20 until the end of the clip. The playback is to start
       at 15:36 on 23 Jan 1997.

         C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
               CSeq: 833
               Session: 12345678
               Range: smpte=0:10:20-;time=19970123T153600Z

         S->C: RTSP/1.0 200 OK
               CSeq: 833
               Date: 23 Jan 1997 15:35:06 GMT
               Range: smpte=0:10:22-;time=19970123T153600Z

       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

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

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    10.6 PAUSE

       The PAUSE request causes the stream delivery to be interrupted
       (halted) temporarily. If the request URL names a stream, only
       playback and recording of that stream is halted. For example, for
       audio, this is equivalent to muting. If the request URL names a
       presentation or group of streams, delivery of all currently active
       streams within the presentation or group is halted. After resuming
       playback or recording, 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

       The PAUSE request may contain a Range header specifying when the
       stream or presentation is to be halted. We refer to this point as the
       "pause point". The header must contain exactly one value rather than
       a time range. The normal play time for the stream is set to the pause
       point. The pause request becomes effective the first time the server
       is encountering the time point specified in any of the currently
       pending PLAY requests. If the Range header specifies a time outside
       any currently pending PLAY requests, the error "457 Invalid Range" is
       returned. If a media unit (such as an audio or video frame) starts
       presentation at exactly the pause point, it is not played or
       recorded.  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.

       A PAUSE request discards all queued PLAY requests. However, the pause
       point in the media stream MUST be maintained. A subsequent PLAY
       request without Range header resumes from the pause point.

       For example, if the server has play requests 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 stops after completing the first

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       play request and discards the second play request.

       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 PLAY request 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 NPT to 14.

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

    10.7 TEARDOWN

       The TEARDOWN request stops the stream delivery for the given URI,
       freeing the resources associated with it. If the URI is the
       presentation URI for this presentation, any RTSP session identifier
       associated with the session is no longer valid. Unless all transport
       parameters are defined by the session description, a SETUP request
       has to be issued before the session can be played again.

       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

    10.8 GET_PARAMETER

       The GET_PARAMETER request retrieves the value of a parameter of a
       presentation or stream specified in the URI. 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

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         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.

    10.9 SET_PARAMETER

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

         A request SHOULD only contain a single parameter to allow the client
         to determine why a particular request failed. If the request contains
         several parameters, the server MUST only act on the request if all of
         the parameters can be set successfully. A server MUST allow a
         parameter to be set repeatedly to the same value, but it MAY disallow
         changing parameter values.

         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 Invalid Parameter

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               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.

    10.10 REDIRECT

       A redirect request informs the client that it must connect to another
       server location. It contains the mandatory header Location, which
       indicates that the client should issue requests for that URL. It may
       contain the parameter Range, which indicates when the redirection
       takes effect. If the client wants to continue to send or receive
       media for this URI, the client MUST issue a TEARDOWN request for the
       current session and a SETUP for the new session at the designated
       host.

       This example request redirects traffic for this URI to the new server
       at the given play time:

         S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
               CSeq: 732
               Location: rtsp://bigserver.com:8001
               Range: clock=19960213T143205Z-

    10.11 RECORD

       This method initiates recording a range of media data according to
       the presentation description. The timestamp reflects start and end
       time (UTC). If no time range is given, use the start or end time
       provided in the presentation description. If the session has already
       started, commence recording immediately.

       The server decides whether to store the recorded data under the
       request-URI or another URI. If the server does not use the request-
       URI, the response SHOULD be 201 (Created) and contain an entity which
       describes the status of the request and refers to the new resource,
       and a Location header.

       A media server supporting recording of live presentations MUST
       support the clock range format; the smpte format does not make sense.

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       In this example, the media server was previously invited to the
       conference indicated.

         C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
               CSeq: 954
               Session: 12345678
               Conference: 128.16.64.19/32492374

    10.12 Embedded (Interleaved) Binary Data

       Certain firewall designs and other circumstances may force a server
       to interleave RTSP methods and stream data. This interleaving should
       generally be avoided unless necessary since it complicates client and
       server operation and imposes additional overhead. 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 hexadecimal), 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.

       The channel identifier is defined in the Transport header with the
       interleaved parameter(Section 12.39).

       When the transport choice is RTP, RTCP messages are also interleaved
       by the server over the TCP connection. As a default, RTCP packets are
       sent on the first available channel higher than the RTP channel. The
       client MAY explicitly request RTCP packets on another channel. This
       is done by specifying two channels in the interleaved parameter of
       the Transport header(Section 12.39).

         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;interleaved=0-1

         S->C: RTSP/1.0 200 OK
               CSeq: 2
               Date: 05 Jun 1997 18:57:18 GMT
               Transport: RTP/AVP/TCP;interleaved=0-1

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               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

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

    11 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.

    11.1 Success 2xx

    11.1.1 250 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.

    11.2 Redirection 3xx

       See [H10.3].

       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.

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    11.3 Client Error 4xx

    11.3.1 405 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.

    11.3.2 451 Parameter Not Understood

       The recipient of the request does not support one or more parameters
       contained in the request.

    11.3.3 452 Conference Not Found

       The conference indicated by a Conference header field is unknown to
       the media server.

    11.3.4 453 Not Enough Bandwidth

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

    11.3.5 454 Session Not Found

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

    11.3.6 455 Method Not Valid in This State

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

    11.3.7 456 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.

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    11.3.8 457 Invalid Range

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

    11.3.9 458 Parameter Is Read-Only

       The parameter to be set by SET_PARAMETER can be read but not
       modified.

    11.3.10 459 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
       stream URL.

    11.3.11 460 Only Aggregate Operation Allowed

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

    11.3.12 461 Unsupported Transport

       The Transport field did not contain a supported transport
       specification.

    11.3.13 462 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.

    11.3.14 551 Option not supported

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

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    12 Header Field Definitions

       HTTP/1.1 [2] or other, non-standard header fields not listed here
       currently have no well-defined meaning and SHOULD be ignored by the
       recipient.

       Table 3 summarizes the header fields used by RTSP. Type "g"
       designates general request headers to be found in both requests and
       responses, type "R" designates request headers, type "r" designates
       response headers, and type "e" designates entity header fields.
       Fields marked with "req." in the column labeled "support" MUST be
       implemented by the recipient for a particular method, while fields
       marked "opt." are optional. Note that not all fields marked "req."
       will be sent in every request of this type. The "req."  means only
       that client (for response headers) and server (for request headers)
       MUST implement the fields. The last column lists the method for which
       this header field is meaningful; the designation "entity" refers to
       all methods that return a message body. Within this specification,
       DESCRIBE and GET_PARAMETER fall into this class.

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       Header               type   support   methods
       Accept               R      opt.      entity
       Accept-Encoding      R      opt.      entity
       Accept-Language      R      opt.      all
       Allow                r      opt.      all
       Authorization        R      opt.      all
       Bandwidth            R      opt.      all
       Blocksize            R      opt.      all but OPTIONS, TEARDOWN
       Cache-Control        g      opt.      SETUP
       Conference           R      opt.      SETUP
       Connection           g      req.      all
       Content-Base         e      opt.      entity
       Content-Encoding     e      req.      SET_PARAMETER
       Content-Encoding     e      req.      DESCRIBE, ANNOUNCE
       Content-Language     e      req.      DESCRIBE, ANNOUNCE
       Content-Length       e      req.      SET_PARAMETER, ANNOUNCE
       Content-Length       e      req.      entity
       Content-Location     e      opt.      entity
       Content-Type         e      req.      SET_PARAMETER, ANNOUNCE
       Content-Type         r      req.      entity
       CSeq                 g      req.      all
       Date                 g      opt.      all
       Expires              e      opt.      DESCRIBE, ANNOUNCE
       From                 R      opt.      all
       If-Modified-Since    R      opt.      DESCRIBE, SETUP
       Last-Modified        e      opt.      entity
       Proxy-Authenticate
       Proxy-Require        R      req.      all
       Public               r      opt.      all
       Range                R      opt.      PLAY, PAUSE, RECORD
       Range                r      opt.      PLAY, PAUSE, RECORD
       Referer              R      opt.      all
       Require              R      req.      all
       Retry-After          r      opt.      all
       RTP-Info             r      req.      PLAY
       Scale                Rr     opt.      PLAY, RECORD
       Session              Rr     req.      all but SETUP, OPTIONS
       Server               r      opt.      all
       Speed                Rr     opt.      PLAY
       Transport            Rr     req.      SETUP
       Unsupported          r      req.      all
       User-Agent           R      opt.      all
       Via                  g      opt.      all
       WWW-Authenticate     r      opt.      all

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       Overview of RTSP header fields

    12.1 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, application/sdp;level=2

    12.2 Accept-Encoding

         See [H14.3]

    12.3 Accept-Language

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

    12.4 Allow

       The Allow response 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.

       Example of use:
         Allow: SETUP, PLAY, RECORD, SET_PARAMETER

    12.5 Authorization

         See [H14.8]

    12.6 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.

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       Bandwidth = "Bandwidth" ":" 1*DIGIT

       Example:
         Bandwidth: 4000

    12.7 Blocksize

       This 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 (416) if the value is syntactically invalid.

    12.8 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"
                                |   "min-fresh"
                                |   "only-if-cached"
                                |   cache-extension
       cache-response-directive =   "public"
                                |   "private"
                                |   "no-cache"
                                |   "no-transform"
                                |   "must-revalidate"

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                                |   "proxy-revalidate"
                                |   "max-age" "=" delta-seconds
                                |   cache-extension
       cache-extension          =   token [ "=" ( token | quoted-string ) ]

       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.

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       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.)

    12.9 Conference

       This request header field establishes a logical connection between a
       pre-established conference and an RTSP stream. The conference-id must
       not be changed for the same RTSP session.

       Conference = "Conference" ":" conference-id Example:
         Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr

       A response code of 452 (452 Conference Not Found) is returned if the
       conference-id is not valid.

    12.10 Connection

       See [H14.10]

    12.11 Content-Base

       See [H14.11]

    12.12 Content-Encoding

       See [H14.12]

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    12.13 Content-Language

       See [H14.13]

    12.14 Content-Length

       This field contains the length of the content of the method (i.e.
       after the double CRLF following the last header). Unlike HTTP, it
       MUST be included in all messages that carry content beyond the header
       portion of the message. If it is missing, a default value of zero is
       assumed. It is interpreted according to [H14.14].

    12.15 Content-Location

       See [H14.15]

    12.16 Content-Type

       See [H14.18]. Note that the content types suitable for RTSP are
       likely to be restricted in practice to presentation descriptions and
       parameter-value types.

    12.17 CSeq

       The CSeq 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, there will be a corresponding response having 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).

    12.18 Date

       See [H14.19].

    12.19 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.

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       A stale cache entry may not normally be returned by a cache (either a
       proxy cache or an user agent cache) unless it is first validated with
       the origin server (or with an intermediate cache that has a fresh
       copy of the entity). See section 13 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 12.8).

    12.20 From

       See [H14.22].

    12.21 Host

       This HTTP request header field is not needed for RTSP. It should be
       silently ignored if sent.

    12.22 If-Match

       See [H14.25].

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       This 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.

    12.23 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

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

    12.25 Location

       See [H14.30].

    12.26 Proxy-Authenticate

       See [H14.33].

    12.27 Proxy-Require

       The Proxy-Require header 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 if not supported. Servers

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       should treat this field identically to the Require field.

       See Section 12.32 for more details on the mechanics of this message
       and a usage example.

    12.28 Public

       See [H14.35].

    12.29 Range

       This 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 3.5), npt (Section 3.6), and clock
       (Section 3.7) range units. Within RTSP, byte ranges [H14.36.1] are
       not meaningful and MUST NOT be used. The header may also contain a
       time parameter in UTC, specifying the time at which the operation is
       to be made effective. Servers supporting the Range header MUST
       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 or recorded. If the Range header is given in a
       time format that is not understood, the recipient should return "501
       Not Implemented".

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

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

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

         The notation is similar to that used for the HTTP/1.1 [2] 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 in the future, although a server may refuse
         to keep server resources for extended idle periods.

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    12.30 Referer

       See [H14.37]. The URL refers to that of the presentation description,
       typically retrieved via HTTP.

    12.31 Retry-After

       See [H14.38].

    12.32 Require

       The Require header is used by clients to query the server about
       options that it may or may not support. The server MUST respond to
       this header by using the Unsupported header to negatively acknowledge
       those options which are NOT supported.

         This is to make sure that the client-server interaction will
         proceed without delay when all options are understood by both
         sides, and only slow down if options are not understood (as in the
         case above). 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" ":"  1#option-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
                 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.

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       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 12.27).

    12.33 RTP-Info

       This field is used to set RTP-specific parameters in the PLAY
       response.

       url:
              Indicates the stream URL which for which the following RTP
              parameters correspond.

       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 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.

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       Syntax:

       RTP-Info        = "RTP-Info" ":" 1#stream-url 1*parameter
       stream-url      = "url" "=" url
       parameter       = ";" "seq" "=" 1*DIGIT
                       | ";" "rtptime" "=" 1*DIGIT

       Example:

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

    12.34 Scale

       A scale value of 1 indicates normal play or record 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 request contains a Range parameter, the new scale value will
       take effect at that time.

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

       Example of playing in reverse at 3.5 times normal rate:

         Scale: -3.5

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    12.35 Speed

       This request header fields parameter 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. If the request contains a Range parameter,
       the new speed value will take effect at that time.

       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.

    12.36 Server

       See [H14.39]

    12.37 Session

       This request and response header field identifies an RTSP session
       started by the media server in a SETUP response and concluded by
       TEARDOWN on the presentation URL. The session identifier is chosen by
       the media server (see Section 3.4). Once a client receives a Session
       identifier, it MUST return it for any request related to that
       session.  A server does not have to set up a session identifier if it
       has other means of identifying a session, such as dynamically
       generated URLs.

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

       The timeout parameter is only allowed in a response header. The
       server uses it to indicate to the client how long the server is
       prepared to wait between RTSP commands before closing the session due
       to lack of activity (see Section A). The timeout is measured in

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       seconds, with a default of 60 seconds (1 minute).

       Note that a session identifier identifies a RTSP session across
       transport sessions or connections. Control messages for more than one
       RTSP URL may be sent within a single RTSP session. Hence, it is
       possible that clients use the same session for controlling many
       streams constituting a presentation, as long as all the streams come
       from the same server. (See example in Section 14). However, multiple
       "user" sessions for the same URL from the same client MUST use
       different session identifiers.

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

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

    12.38 Timestamp

       The timestamp general header 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.

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

    12.39 Transport

       This request header 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 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 in the response to
       indicate the values actually chosen.

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       A Transport request header field may contain a list of transport
       options acceptable to the client. In that case, the server MUST
       return a single option which was actually chosen.

       The syntax for the transport specifier is

           transport/profile/lower-transport.

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

       Below are the configuration parameters associated with transport:

       General parameters:

       unicast | multicast:
              mutually exclusive indication of whether unicast or multicast
              delivery will be attempted. Default value is multicast.
              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 to which a stream will be sent. The client may
              specify the multicast address with the destination parameter.
              To avoid becoming the unwitting perpetrator of a remote-
              controlled denial-of-service attack, a server SHOULD
              authenticate the client and SHOULD log such attempts before
              allowing the client to direct a media stream to an address not
              chosen by the server. This is particularly important if RTSP
              commands are issued via UDP, but implementations cannot rely
              on TCP as reliable means of client identification by itself. A
              server SHOULD not allow a client to direct media streams to an
              address that differs from the address commands are coming
              from.

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

         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.

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       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.

       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.

       append:
              If the mode parameter includes RECORD, the append parameter
              indicates that the media data should append to the existing
              resource rather than overwrite it. If appending is requested
              and the server does not support this, it MUST refuse the
              request rather than overwrite the resource identified by the
              URI. The append parameter is ignored if the mode parameter
              does not contain RECORD.

       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 10.12.
              The argument provides the channel number to be used in the $
              statement. 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.

         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:

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

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

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

       ssrc:
              The ssrc parameter indicates the RTP SSRC [24, Sec. 3] value
              that should be (request) or will be (response) used by the
              media server. This parameter is only valid for unicast
              transmission. It identifies the synchronization source to be
              associated with the media stream.

       Transport           =    "Transport" ":"
                                1\#transport-spec
       transport-spec      =    transport-protocol/profile[/lower-transport]
                                *parameter
       transport-protocol  =    "RTP"
       profile             =    "AVP"
       lower-transport     =    "TCP" | "UDP"
       parameter           =    ( "unicast" | "multicast" )
                           |    ";" "destination" [ "=" address ]
                           |    ";" "interleaved" "=" channel [ "-" channel ]
                           |    ";" "append"
                           |    ";" "ttl" "=" ttl
                           |    ";" "layers" "=" 1*DIGIT
                           |    ";" "port" "=" port [ "-" port ]
                           |    ";" "client_port" "=" port [ "-" port ]
                           |    ";" "server_port" "=" port [ "-" port ]
                           |    ";" "ssrc" "=" ssrc
                           |    ";" "mode" = <"> 1\#mode <">
       ttl                 =    1*3(DIGIT)
       port                =    1*5(DIGIT)
       ssrc                =    8*8(HEX)
       channel             =    1*3(DIGIT)
       address             =    host
       mode                =    <"> *Method <"> | Method

       Example:
         Transport: RTP/AVP;multicast;ttl=127;mode="PLAY",
                    RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"

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

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    12.40 Unsupported

       The Unsupported response header lists the features not supported by
       the server. In the case where the feature was specified via the
       Proxy-Require field (Section 12.32), if there is a proxy on the path
       between the client and the server, the proxy MUST insert a message
       reply with an error message "551 Option Not Supported".

       See Section 12.32 for a usage example.

    12.41 User-Agent

       See [H14.42]

    12.42 Vary

       See [H14.43]

    12.43 Via

       See [H14.44].

    12.44 WWW-Authentica

       See [H14.46].

    13 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 or
       included with ANNOUNCE. (Since the responses for anything but
       DESCRIBE and GET_PARAMETER do not return any data, caching is not
       really an issue for these requests.) However, it is desirable for the
       continuous media data, typically delivered out-of-band with respect
       to RTSP, to be cached, as well as the session description.

       On receiving a SETUP or PLAY request, a proxy ascertains whether it
       has an up-to-date copy of the continuous media content and its
       description. It can determine whether the copy is up-to-date by
       issuing a SETUP or DESCRIBE request, respectively, and comparing the
       Last-Modified header with that of the cached copy. If the copy is not
       up-to-date, it modifies the SETUP transport parameters as appropriate
       and forwards the request to the origin server. Subsequent control
       commands such as PLAY or PAUSE then pass the proxy unmodified. The
       proxy delivers the continuous media data to the client, while
       possibly making a local copy for later reuse. The exact behavior
       allowed to the cache is given by the cache-response directives

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       described in Section 12.8. 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.

    14 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.

    14.1 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
       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
               Content-Type: application/sdp

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               v=0
               o=- 2890844526 2890842807 IN IP4 192.16.24.202
               s=RTSP Session
               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
               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
               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
               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
               Session: 12345678
               Range: smpte=0:10:00-

         A->C: RTSP/1.0 200 OK
               CSeq: 2
               Session: 12345678

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               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
               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
               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.

    14.2 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.

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       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://foo/twister RTSP/1.0
               CSeq: 1

         M->C: RTSP/1.0 200 OK
               CSeq: 1
               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
               a=control:rtsp://foo/twister
               t=0 0
               m=audio 0 RTP/AVP 0
               a=control:rtsp://foo/twister/audio
               m=video 0 RTP/AVP 26
               a=control:rtsp://foo/twister/video

         C->M: SETUP rtsp://foo/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://foo/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://foo/twister RTSP/1.0
               CSeq: 4
               Range: npt=0-
               Session: 12345678

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         M->C: RTSP/1.0 200 OK
               CSeq: 4
               Session: 12345678
               RTP-Info: url=rtsp://foo/twister/video;
                 seq=9810092;rtptime=3450012

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

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

         C->M: PAUSE rtsp://foo/twister RTSP/1.0
               CSeq: 6
               Session: 12345678

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

         C->M: SETUP rtsp://foo/twister RTSP/1.0
               CSeq: 7
               Transport: RTP/AVP;unicast;client_port=10000

         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 disallowed
       for that presentation by the server. 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.

    14.3 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:

              Accept: application/x-rtsp-mh, application/sdp

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              CSeq: 1

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

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

        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
              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

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       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.

    14.4 Live Media Presentation Using Multicast

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

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

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

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

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

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

               v=0
               o=- 2890844526 2890842807 IN IP4 192.16.24.202
               s=RTSP Session
               m=audio 3456 RTP/AVP 0
               a=control:rtsp://live.example.com/concert/audio
               c=IN IP4 224.2.0.1/16

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

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               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

    14.5 Playing media into an existing session

       A conference participant C wants to have the media server M play back
       a demo tape into an existing conference. C indicates to the media
       server that the network addresses and encryption keys are already
       given by the conference, so they should not be chosen by the server.
       The example omits the simple ACK responses.

         C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
               CSeq: 1
               Accept: application/sdp

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

               v=0
               o=- 2890844526 2890842807 IN IP4 192.16.24.202
               s=RTSP Session
               i=See above
               t=0 0
               m=audio 0 RTP/AVP 0

         C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0
               CSeq: 2
               Transport: RTP/AVP;multicast;destination=225.219.201.15;
                          port=7000-7001;ttl=127
               Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr

         M->C: RTSP/1.0 200 OK
               CSeq: 2
               Transport: RTP/AVP;multicast;destination=225.219.201.15;

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                          port=7000-7001;ttl=127
               Session: 91389234234
               Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr

         C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
               CSeq: 3
               Session: 91389234234

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

    14.6 Recording

       The conference participant client C asks the media server M to record
       the audio and video portions of a meeting. The client uses the
       ANNOUNCE method to provide meta-information about the recorded
       session to the server.

         C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
               CSeq: 90
               Content-Type: application/sdp
               Content-Length: 121

               v=0
               o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
               s=IETF Meeting, Munich - 1
               i=The thirty-ninth IETF meeting will be held in Munich, Germany
               u=http://www.ietf.org/meetings/Munich.html
               e=IETF Channel 1 <ietf39-mbone@uni-koeln.de>
               p=IETF Channel 1 +49-172-2312 451
               c=IN IP4 224.0.1.11/127
               t=3080271600 3080703600
               a=tool:sdr v2.4a6
               a=type:test
               m=audio 21010 RTP/AVP 5
               c=IN IP4 224.0.1.11/127
               a=ptime:40
               m=video 61010 RTP/AVP 31
               c=IN IP4 224.0.1.12/127

         M->C: RTSP/1.0 200 OK
               CSeq: 90

         C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
               CSeq: 91
               Transport: RTP/AVP;multicast;destination=224.0.1.11;
                          port=21010-21011;mode=record;ttl=127

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         M->C: RTSP/1.0 200 OK
               CSeq: 91
               Session: 50887676
               Transport: RTP/AVP;multicast;destination=224.0.1.11;
                          port=21010-21011;mode=record;ttl=127

         C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
               CSeq: 92
               Session: 50887676
               Transport: RTP/AVP;multicast;destination=224.0.1.12;
                          port=61010-61011;mode=record;ttl=127

         M->C: RTSP/1.0 200 OK
               CSeq: 92
               Transport: RTP/AVP;multicast;destination=224.0.1.12;
                          port=61010-61011;mode=record;ttl=127

         C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
               CSeq: 93
               Session: 50887676
               Range: clock=19961110T1925-19961110T2015

         M->C: RTSP/1.0 200 OK
               CSeq: 93

    15 Syntax

       The RTSP syntax is described in an augmented Backus-Naur form (BNF)
       as used in RFC 2068 [2].

    15.1 Base Syntax

       OCTET              =      <any 8-bit sequence of data>
       CHAR               =      <any US-ASCII character (octets 0 - 127)>
       UPALPHA            =      <any US-ASCII uppercase letter "A".."Z">
       LOALPHA            =      <any US-ASCII lowercase letter "a".."z">
       ALPHA              =      UPALPHA | LOALPHA

       DIGIT              =      <any US-ASCII digit "0".."9">
       CTL                =      <any US-ASCII control character
                                  (octets 0 - 31) and DEL (127)>
       CR                 =      <US-ASCII CR, carriage return (13)>
       LF                 =      <US-ASCII LF, linefeed (10)>

       SP                 =      <US-ASCII SP, space (32)>
       HT                 =      <US-ASCII HT, horizontal-tab (9)>
       <">                =      <US-ASCII double-quote mark (34)>
       CRLF               =      CR LF

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       LWS                =      [CRLF] 1*( SP | HT )
       TEXT               =      <any OCTET except CTLs>
       tspecials          =      "(" | ")" | "<" | ">" | "@"
                          |       "," | ";" | ":" | "\" | <">
                          |       "/" | "[" | "]" | "?" | "="
                          |       "{" | "}" | SP | HT

       token              =      1*<any CHAR except CTLs or tspecials>
       quoted-string      =      ( <"> *(qdtext) <"> )
       qdtext             =      <any TEXT except <">>
       quoted-pair        =      "\" 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 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 [H15.1] for client authentication issues,
              and [H15.2] for issues regarding support for multiple
              authentication mechanisms.

       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

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              users of the servers. See [H15.3] 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
              [H15.4] for further details.

       Attacks Based On File and Path Names:
              Though RTSP URLs are opaque handles that do not necessarily
              have file system semantics, it is anticipated that many
              implementations will translate 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 [H15.6] 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 [H15.7] 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 [H15.8] 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 headers 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. ([H15.9])

       In addition to the recommendations in the current HTTP specification
       (RFC 2068 [2], as of this writing), future HTTP specifications may
       provide additional guidance on security issues.

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       The following are added considerations for RTSP implementations.

       Concentrated denial-of-service attack:
              The protocol offers the opportunity for a remote-controlled
              denial-of-service attack. The attacker may initiate traffic
              flows to one or more IP addresses by specifying them as the
              destination in SETUP requests. While the attacker's IP address
              may be known in this case, this is not always useful in
              prevention of more attacks or ascertaining the attackers
              identity. Thus, an RTSP server SHOULD only allow client-
              specified destinations for RTSP-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 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 [8]
              authentication. In environments requiring tighter security for
              the control messages, the RTSP control stream may be
              encrypted.

       Stream issues:
              RTSP only provides for stream control. Stream delivery issues
              are not covered in this section, nor in the rest of this memo.
              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.

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    Appendix A: RTSP Protocol State Machines

       The RTSP client and server state machines describe the behavior of
       the protocol from RTSP session initialization through RTSP session
       termination.

       State is defined on a per object basis. An object is uniquely
       identified by the stream URL and the RTSP session identifier. Any
       request/reply using aggregate URLs denoting RTSP presentations
       composed of multiple streams will have an effect on the individual
       states of all the streams. For example, if the presentation /movie
       contains two streams, /movie/audio and /movie/video, then the
       following command:

         PLAY rtsp://foo.com/movie RTSP/1.0
         CSeq: 559
         Session: 12345678

       will have an effect on the states of movie/audio and movie/video.

         This example does not imply a standard way to represent streams in
         URLs or a relation to the filesystem. See Section 3.2.

       The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER,
       SET_PARAMETER do not have any effect on client or server state and
       are therefore not listed in the state tables.

    A.1 Client State Machine

       The client can assume the following states:

       Init:
              SETUP has been sent, waiting for reply.

       Ready:
              SETUP reply received or PAUSE reply received while in Playing
              state.

       Playing:
              PLAY reply received

       Recording:
              RECORD reply received

       In general, the client changes state on receipt of replies to
       requests. Note that some requests are effective at a future time or
       position (such as a PAUSE), and state also changes accordingly. If no
       explicit SETUP is required for the object (for example, it is

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       available via a multicast group), state begins at Ready. In this
       case, there are only two states, Ready and Playing. The client also
       changes state from Playing/Recording to Ready when the end of the
       requested range is reached.

       The "next state" column indicates the state assumed after receiving a
       success response (2xx). If a request yields a status code of 3xx, the
       state becomes Init, and a status code of 4xx yields no change in
       state. Messages not listed for each state MUST NOT be issued by the
       client in that state, with the exception of messages not affecting
       state, as listed above. Receiving a REDIRECT from the server is
       equivalent to receiving a 3xx redirect status from the server.

       state       message sent     next state after response
       Init        SETUP            Ready
                   TEARDOWN         Init
       Ready       PLAY             Playing
                   RECORD           Recording
                   TEARDOWN         Init
                   SETUP            Ready
       Playing     PAUSE            Ready
                   TEARDOWN         Init
                   PLAY             Playing
                   SETUP            Playing (changed transport)
       Recording   PAUSE            Ready
                   TEARDOWN         Init
                   RECORD           Recording
                   SETUP            Recording (changed transport)

    A.2 Server State Machine

       The server can assume the following states:

       Init:
              The initial state, no valid SETUP has been received yet.

       Ready:
              Last SETUP received was successful, reply sent or after
              playing, last PAUSE received was successful, reply sent.

       Playing:
              Last PLAY received was successful, reply sent. Data is being
              sent.

       Recording:
              The server is recording media data.

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       In general, the server changes state on receiving requests. If the
       server is in state Playing or Recording and in unicast mode, it MAY
       revert to Init and tear down the RTSP session if it has not received
       "wellness" information, such as RTCP reports or RTSP commands, from
       the client for a defined interval, with a default of one minute. The
       server can declare another timeout value in the Session response
       header (Section 12.37). If the server is in state Ready, it MAY
       revert to Init if it does not receive an RTSP request for an interval
       of more than one minute. Note that some requests (such as PAUSE) may
       be effective at a future time or position, and server state changes
       at the appropriate time. The server reverts from state Playing or
       Recording to state Ready at the end of the range requested by the
       client.

       The REDIRECT message, when sent, is effective immediately unless it
       has a Range header specifying when the redirect is effective. In such
       a case, server state will also change at the appropriate time.

       If no explicit SETUP is required for the object, the state starts at
       Ready and there are only two states, Ready and Playing.

       The "next state" column indicates the state assumed after sending a
       success response (2xx). If a request results in a status code of 3xx,
       the state becomes Init. A status code of 4xx results in no change.

         state           message received  next state
         Init            SETUP             Ready
                         TEARDOWN          Init
         Ready           PLAY              Playing
                         SETUP             Ready
                         TEARDOWN          Init
                         RECORD            Recording
         Playing         PLAY              Playing
                         PAUSE             Ready
                         TEARDOWN          Init
                         SETUP             Playing
         Recording       RECORD            Recording
                         PAUSE             Ready
                         TEARDOWN          Init
                         SETUP             Recording

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    Appendix B: Interaction with RTP

       RTSP allows media clients to control selected, non-contiguous
       sections of media presentations, rendering those streams with an RTP
       media layer[24]. 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 12.34), 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 12.35) 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 12.33) header provides
       the first sequence number of the next segment.

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    Appendix C: Use of SDP for RTSP Session Descriptions

       The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
       describe streams or presentations in RTSP. Such usage is limited to
       specifying means of access and encoding(s) for:

       aggregate control:
              A presentation composed of streams from one or more servers
              that are not available for aggregate control. Such a
              description is typically retrieved by HTTP or other non-RTSP
              means. However, they may be received with ANNOUNCE methods.

       non-aggregate control:
              A presentation composed of multiple streams from a single
              server that are available for aggregate control. Such a
              description is typically returned in reply to a DESCRIBE
              request on a URL, or received in an ANNOUNCE method.

       This appendix describes how an SDP file, retrieved, for example,
       through HTTP, determines the operation of an RTSP session. It also
       describes how a client should interpret SDP content returned in reply
       to a DESCRIBE request. SDP 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).

    C.1 Definitions

       The terms "session-level", "media-level" and other key/attribute
       names and values used in this appendix are to be used as defined in
       SDP (RFC 2327 [6]):

    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.

       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 1808 [25].
       Implementations should look for a base URL in the following order:

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       1.     The RTSP Content-Base field
       2.     The RTSP Content-Location field
       3.     The RTSP request URL

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

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

       Example:
         m=audio 0 RTP/AVP 31

    C.1.3 Payload type(s)

       The payload type(s) are specified in the "m=" field. In case the
       payload type is a static payload type from RFC 1890 [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 1890 (Sections 5 and 6), or an experimental encoding
       with a "X-" prefix as specified in SDP (RFC 2327 [6]).  Codec-
       specific parameters are not specified in this field, but rather in
       the "fmtp" attribute described below. Implementors seeking to
       register new encodings should follow the procedure in RFC 1890 [1].
       If the media type is not suited to the RTP AV profile, then it is
       recommended that a new profile be created and the appropriate profile
       name be used in lieu of "RTP/AVP" in the "m=" field.

    C.1.4 Format-specific parameters

       Format-specific parameters are conveyed using the "fmtp" media
       attribute. The syntax of the "fmtp" attribute is specific to the
       encoding(s) that the attribute refers to. Note that the packetization
       interval is conveyed using the "ptime" attribute.

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    C.1.5 Range of presentation

       The "a=range" attribute defines the total time range of the stored
       session. (The length of live sessions can be deduced from the "t" and
       "r" parameters.) Unless the presentation contains media streams of
       different durations, the range attribute is a session-level
       attribute. The unit is specified first, followed by the value range.
       The units and their values are as defined in Section 3.5, 3.6 and
       3.7.

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

    C.1.6 Time of availability

       The "t=" field MUST contain suitable values for the start and stop
       times for both aggregate and non-aggregate stream control. With
       aggregate control, the server SHOULD indicate a stop time value for
       which it guarantees the description to be valid, and a start time
       that is equal to or before the time at which the DESCRIBE request was
       received. It MAY also indicate start and stop times of 0, meaning
       that the session is always available. With non-aggregate control, the
       values should reflect the actual period for which the session is
       available in keeping with SDP semantics, and not depend on other
       means (such as the life of the web page containing the description)
       for this purpose.

    C.1.7 Connection Information

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

      C.1.8 Entity Tag

       The optional "a=etag" attribute identifies a version of the session
       description. It is opaque to the client. SETUP requests may include
       this identifier in the If-Match field (see section 12.22) 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

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

    C.2 Aggregate Control Not Available

       If a presentation does not support aggregate control and multiple
       media sections are specified, each section MUST have the control URL
       specified via the "a=control:" attribute.

       Example:
         v=0
         o=- 2890844256 2890842807 IN IP4 204.34.34.32
         s=I came from a web page
         t=0 0
         c=IN IP4 0.0.0.0
         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 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.

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

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       Example:
         v=0
         o=- 2890844256 2890842807 IN IP4 204.34.34.32
         s=I contain
         i=<more info>
         t=0 0
         c=IN IP4 0.0.0.0
         a=control:rtsp://example.com/movie/
         m=video 8002 RTP/AVP 31
         a=control:trackID=1
         m=audio 8004 RTP/AVP 3
         a=control:trackID=2

       In this example, the client is required to establish a single RTSP
       session to the server, and uses the URLs
       rtsp://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/ controls the
       whole movie.

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    Appendix D: Minimal RTSP implementation

    D.1 Client

       A client implementation MUST be able to do the following :

         * Generate the following requests: SETUP, TEARDOWN, and one of PLAY
           (i.e., a minimal playback client) or RECORD (i.e., a minimal
           recording client). If RECORD is implemented, ANNOUNCE must be
           implemented as well.
         * Include the following headers in requests: CSeq, Connection,
           Session, Transport. If ANNOUNCE is implemented, the capability to
           include headers Content-Language, Content-Encoding, Content-
           Length, and Content-Type should be as well.
         * Parse and understand the following headers in responses: CSeq,
           Connection, Session, Transport, Content-Language, Content-
           Encoding, Content-Length, Content-Type. If RECORD is implemented,
           the Location header must be understood as well.  RTP-compliant
           implementations should also implement RTP-Info.
         * Understand the class of each error code received and notify the
           end-user, if one is present, of error codes in classes 4xx and
           5xx. The notification requirement may be relaxed if the end-user
           explicitly does not want it for one or all status codes.
         * Expect and respond to asynchronous requests from the server, such
           as ANNOUNCE. This does not necessarily mean that it should
           implement the ANNOUNCE method, merely that it MUST respond
           positively or negatively to any request received from the server.

       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".

         * Implement RTP/AVP/UDP as a valid transport.
         * Inclusion of the User-Agent header.
         * Understand SDP session descriptions as defined in Appendix C
         * Accept media initialization formats (such as SDP) from standard
           input, command line, or other means appropriate to the operating
           environment to act as a "helper application" for other
           applications (such as web browsers).

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

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    D.1.1 Basic Playback

       To support on-demand playback of media streams, the client MUST
       additionally be able to do the following:
         * generate the PAUSE request;
         * implement the REDIRECT method, and the Location header.

    D.1.2 Authentication-enabled

       In order to access media presentations from RTSP servers that require
       authentication, the client MUST additionally be able to do the
       following:
         * recognize the 401 status code;
         * parse and include the WWW-Authenticate header;
         * implement Basic Authentication and Digest Authentication.

    D.2 Server

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

         * Implement the following methods: SETUP, TEARDOWN, OPTIONS and
           either PLAY (for a minimal playback server) or RECORD (for a
           minimal recording server).  If RECORD is implemented, ANNOUNCE
           should be implemented as well.
         * Include the following headers in responses: Connection,
           Content-Length, Content-Type, Content-Language, Content-Encoding,
           Transport, Public. The capability to include the Location header
           should be implemented if the RECORD method is. RTP-compliant
           implementations should also implement the RTP-Info field.
         * Parse and respond appropriately to the following headers in
           requests: Connection, Session, Transport, 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".

         * Implement RTP/AVP/UDP as a valid transport.
         * Inclusion of the Server header.
         * Implement the DESCRIBE method.
         * Generate SDP session descriptions as defined in Appendix C

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

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    D.2.1 Basic Playback

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

         * Recognize the Range header, and return an error if seeking is not
           supported.
         * Implement the PAUSE method.

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

         * Include and parse the Range header, with NPT units.
           Implementation of SMPTE units is recommended.
         * Include the length of the media presentation in the media
           initialization information.
         * Include mappings from data-specific timestamps to NPT. When RTP
           is used, the rtptime portion of the RTP-Info field may be used to
           map RTP timestamps to NPT.

         Client implementations may use the presence of length information
         to determine if the clip is seekable, and visibly 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.

    D.2.2 Authentication-enabled

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

         * Generate the 401 status code when authentication is required for
           the resource.
         * Parse and include the WWW-Authenticate header
         * Implement Basic Authentication and Digest Authentication

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    Appendix E: Authors' Addresses

       Henning Schulzrinne
       Dept. of Computer Science
       Columbia University
       1214 Amsterdam Avenue
       New York, NY 10027
       USA

       EMail: schulzrinne@cs.columbia.edu

       Anup Rao
       Netscape Communications Corp.
       501 E. Middlefield Road
       Mountain View, CA 94043
       USA

       EMail: anup@netscape.com

       Robert Lanphier
       RealNetworks
       1111 Third Avenue Suite 2900
       Seattle, WA 98101
       USA

       EMail: robla@real.com

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    Appendix F: Acknowledgements

       This memo is based on the functionality of the original RTSP document
       submitted in October 96. 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, 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, John K. Ho, Philipp Hoschka,
       Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan
       Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Papadopouli,
       Sujal Patel, Ema Patki, Alagu Periyannan, Igor Plotnikov, Pinaki
       Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and
       John Francis Stracke.

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    References

       1      Schulzrinne, H., "RTP profile for audio and video conferences
              with minimal control", RFC 1890, January 1996.

       2      Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
              Berners-Lee, "Hypertext transfer protocol - HTTP/1.1", RFC
              2068, January 1997.

       3      Yergeau, F., Nicol, G., Adams, G., and M. Duerst,
              "Internationalization of the hypertext markup language", RFC
              2070, January 1997.

       4      Bradner, S., "Key words for use in RFCs to indicate
              requirement levels", BCP 14, RFC 2119, March 1997.

       5      ISO/IEC, "Information technology - generic coding of moving
              pictures and associated audio information - 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      Handley, M., and V. Jacobson, "SDP: Session Description
              Protocol", RFC 2327, April 1998.

       7      Franks, J., Hallam-Baker, P., and J. Hostetler, "An extension to
              HTTP: digest access authentication", RFC 2069, January 1997.

       8      Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
              1980.

       9      Hinden, B. and C. Partridge, "Version 2 of the reliable data
              protocol (RDP)", RFC 1151, April 1990.

       10     Postel, J., "Transmission control protocol", STD 7, RFC 793,
              September 1981.

       11     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.

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

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    RFC 2326              Real Time Streaming Protocol            April 1998

       13     McMahon, P., "GSS-API authentication method for SOCKS version
              5", RFC 1961, June 1996.

       14     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.

       15     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.

       16     Crocker, D. and P. Overell, "Augmented BNF for syntax
              specifications: ABNF", RFC 2234, November 1997.

       17     Braden, B., "Requirements for internet hosts - application and
              support", STD 3, RFC 1123, October 1989.

       18     Elz, R., "A compact representation of IPv6 addresses", RFC
              1924, April 1996.

       19     Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform
              resource locators (URL)", RFC 1738, December 1994.

       20     Yergeau, F., "UTF-8, a transformation format of ISO 10646",
              RFC 2279, January 1998.

       22     Braden, B., "T/TCP - TCP extensions for transactions
              functional specification", RFC 1644, July 1994.

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

       23     Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
              "RTP: a transport protocol for real-time applications", RFC
              1889, January 1996.

       24     Fielding, R., "Relative uniform resource locators", RFC 1808,
              June 1995.

    Schulzrinne, et. al.        Standards Track                    [Page 91]

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    Full Copyright Statement

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

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