Internet Engineering Task Force                                   MMUSIC WG
Internet Draft                          H. Schulzrinne, A. Rao, R. Lanphier
ietf-mmusic-rtsp-02.txt
draft-ietf-mmusic-rtsp-03.txt     Columbia U./Netscape/Progressive Networks
March 27,
July 30, 1997                                     Expires: September 26, 1997 January 30, 1998

                  Real Time Streaming Protocol (RTSP)

STATUS OF THIS MEMO

   This document is an Internet-Draft. Internet-Drafts are working
   documents of the Internet Engineering Task Force (IETF), its areas,
   and its working groups.  Note that other groups may also distribute
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   ftp.isi.edu (US West Coast).

   Distribution of this document is unlimited.

                                 ABSTRACT

  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 delivery mechanisms based upon RTP (RFC
   1889).

H. Schulzrinne, A. Rao, R. Lanphier                            Page  1

Contents

     * 1 Introduction
          + 1.1 Purpose

   The Real-Time Streaming
          + 1.2 Requirements
          + 1.3 Terminology
          + 1.4 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 9.11).
   In other words, Properties
          + 1.5 Extending 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
          + 1.6 Overall Operation
          + 1.7 RTSP connection; instead, a server maintains
   a session labeled by an identifier. An States
          + 1.8 Relationship with Other Protocols
     * 2 Notational Conventions
     * 3 Protocol Parameters
          + 3.1 RTSP session is in no way tied
   to a transport-level connection such as a TCP connection. During an Version
          + 3.2 RTSP session, an URL
          + 3.3 Conference Identifiers
          + 3.4 Session Identifiers
          + 3.5 SMPTE Relative Timestamps
          + 3.6 Normal Play Time
          + 3.7 Absolute Time
     * 4 RTSP client may open Message
          + 4.1 Message Types
          + 4.2 Message Headers
          + 4.3 Message Body
          + 4.4 Message Length
     * 5 General Header Fields
     * 6 Request
          + 6.1 Request Line
          + 6.2 Request Header Fields
     * 7 Response
          + 7.1 Status-Line
               o 7.1.1 Status Code 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, 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: Reason Phrase
               o RTSP introduces a number of new methods 7.1.2 Response Header Fields
     * 8 Entity
          + 8.1 Entity Header Fields
          + 8.2 Entity Body
     * 9 Connections
          + 9.1 Pipelining
          + 9.2 Reliability and has a different
         protocol identifier. Acknowledgements
     * 10 Method Definitions
          + 10.1 OPTIONS
          + 10.2 DESCRIBE
          + 10.3 ANNOUNCE
          + 10.4 SETUP
          + 10.5 PLAY
          + 10.6 PAUSE
          + 10.7 TEARDOWN
          + 10.8 GET_PARAMETER
          + 10.9 SET_PARAMETER
          + 10.10 REDIRECT

H. Schulzrinne, A. Rao, R. Lanphier                            Page  2
          + 10.11 RECORD
          + 10.12 Embedded (Interleaved) Binary Data
     * 11 Status Code Definitions
          + 11.1 Redirection 3xx
          + 11.2 Client Error 4xx
               o An RTSP server needs to maintain state by default in almost
         all cases, as opposed to the stateless nature of HTTP. (RTSP
         servers and clients MAY use the HTTP state maintenance
         mechanism [2].) 11.2.1 405 Method Not Allowed
               o Both an RTSP server and client can issue requests. 11.2.2 451 Parameter Not Understood
               o Data is carried out-of-band, by a different protocol. (There
         is an exception to this.) 11.2.3 452 Conference Not Found
               o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
         8859-1, consistent with current HTML internationalization
         efforts [3]. 11.2.4 453 Not Enough Bandwidth
               o The Request-URI always contains the absolute URI. Because of
         backward compatibility with a historical blunder, HTTP/1.1
         carries only the absolute path 11.2.5 45x Session Not Found
               o 11.2.6 45x Method Not Valid in the request This makes virtual hosting easier. However, this is
             incompatible with HTTP/1.1, which may be a bad idea.

   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 State
               o 11.2.7 45x Header Field Not Valid for
        security reasons.

   Invitation Resource
               o 11.2.8 45x Invalid Range
               o 11.2.9 45x Parameter Is Read-Only
               o 11.2.10 45x Aggregate operation not allowed
               o 11.2.11 45x Only aggregate operation allowed
     * 12 Header Field Definitions
          + 12.1 Accept
          + 12.2 Accept-Encoding
          + 12.3 Accept-Language
          + 12.4 Allow
          + 12.5 Authorization
          + 12.6 Bandwidth
          + 12.7 Blocksize
          + 12.8 C-PEP
          + 12.9 C-PEP-Info
          + 12.10 Cache-Control
          + 12.11 Conference
          + 12.12 Connection
          + 12.13 Content-Encoding
          + 12.14 Content-Language
          + 12.15 Content-Length
          + 12.16 Content-Type
          + 12.17 Date
          + 12.18 Expires
          + 12.19 From
          + 12.20 Host
          + 12.21 If-Modified-Since
          + 12.22 Last-Modified
          + 12.23 Location
          + 12.24 PEP
          + 12.25 PEP-Info
          + 12.26 Proxy-Authenticate
          + 12.27 Public
          + 12.28 Range
          + 12.29 Referer
          + 12.30 Retry-After
          + 12.31 Scale
          + 12.32 Speed
          + 12.33 Server

H. Schulzrinne, A. Rao, R. Lanphier                            Page  3
          + 12.34 Session
          + 12.35 Transport
          + 12.36 Transport-Info
          + 12.37 User-Agent
          + 12.38 Vary
          + 12.39 Via
          + 12.40 WWW-Authenticate
     * 13 Caching
     * 14 Examples
          + 14.1 Media on Demand (Unicast)
          + 14.2 Streaming of a Container file
          + 14.3 Live Media Presentation Using Multicast
          + 14.4 Playing media server to a conference: A media server can be
        "invited" to join into an existing conference, either to play back
        media into the presentation or to record all or session
          + 14.5 Recording
     * 15 Syntax
          + 15.1 Base Syntax
     * 16 Security Considerations
     * A RTSP Protocol State Machines
          + A.1 Client State Machine
          + A.2 Server State Machine
     * B Open Issues
     * C Changes
     * D Author Addresses
     * E Acknowledgements
     * References

1 Introduction

1.1 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls
   either a subset single or several time-synchronized streams of the continuous
   media in a presentation. This mode is useful for distributed
        teaching applications. Several parties in the conference may
        take turns "pushing such as audio and video. It does not typically deliver the remote control buttons".

   Addition
   continuous streams itself, although interleaving of media to an existing presentation: Particularly for live
        presentations, it is useful if the server can tell the client
        about additional continuous
   media becoming available. stream with the control stream is possible (see Section 10.12).
   In other words, RTSP requests may be handled by proxies, tunnels and caches acts as in
   HTTP/1.1.

1.2 Requirements a ``network remote control'' for
   multimedia servers.

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are set of streams to be interpreted as described in RFC xxxx [4].

1.3 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [5].  Terms
   not listed here are controlled is defined as in HTTP/1.1.

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

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

   Connection: A transport layer virtual circuit established between two
        programs presentation
   description. This memorandum does not define a format for the purpose of communication.

   Continuous media: Data where there is a timing relationship between
        source and sink, that is, the sink must reproduce the timing
        relationshop that existed at the source. The most common
        examples of continuous media are audio and motion video.
        Continuous media can be realtime (interactive) , where there is
        a "tight" timing relationship between source and sink, or
        streaming (playback) , where the relationship
   presentation description.

H. Schulzrinne, A. Rao, R. Lanphier                            Page  4
   There is less strict.

   Participant: Participants are members no notion of conferences. A participant
        may be a machine, e.g., a media record or playback server.

   Media server: The network entity providing playback or recording
        services for one or more media streams. Different media streams
        within an RTSP connection; instead, a presentation may originate from different media
        servers. A media server may reside on the same or maintains
   a different
        host as the web server the presentation session labeled by an identifier. An RTSP session is invoked from.

   Media parameter: Parameter specific in no way tied
   to a media type that transport-level connection such as a TCP connection. During an
   RTSP session, an RTSP client may be
        changed while open and close many reliable
   transport connections to the stream is being played or prior server to it.

   (Media) stream: A single media instance, e.g., an audio stream or issue RTSP requests.
   Alternatively, it may use a
        video stream as well connectionless transport protocol such as a single whiteboard or shared
        application group. When using RTP, a stream consists of all RTP
        and RTCP packets created
   UDP.

   The streams controlled by a source within an RTSP may use RTP session. This
        is equivalent to [1], but the definition of a DSM-CC stream.

   Message: The basic unit operation of
   RTSP communication, consisting of a
        structured sequence of octets matching does not depend on the syntax defined transport mechanism used to carry
   continuous media.

   The protocol is intentionally similar in
        Section 14 syntax and transmitted via operation to
   HTTP/1.1, so that extension mechanisms to HTTP can in most cases also
   be added to RTSP. However, RTSP differs in a connection or number of important
   aspects from HTTP:

     * RTSP introduces a connectionless
        protocol.

   Presentation: A set number of one or more streams which the server allows
        the client to manipulate together. A presentation new methods and has a single
        time axis for all streams belonging different
       protocol identifier.
     * An RTSP server needs to it. Presentations are
        defined maintain state by presentation descriptions (see below). A presentation
        description contains RTSP URIs that define which streams can be
        controlled individually and an RTSP URI default in almost all
       cases, as opposed to control the whole
        presentation. A movie or live concert consisting stateless nature of one or more
        audio HTTP.
     * Both an RTSP server and video streams client can issue requests.
     * Data is be an example of carried out-of-band, by a presentation.

   Presentation description: A presentation description 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
        information about one or more media streams within a
        presentation, such as the set absolute URI. Because of encodings, network addresses
        and information about
       backward compatibility with a historical blunder, HTTP/1.1 carries
       only the content. Other IETF protocols such as
        SDP [6] use absolute path in the term "session" for request and puts the host name in a live presentation. The
        presentation description may take
       separate header field.

     This makes ``virtual hosting'' easier, where a single host with one
     IP address hosts several different formats,
        including but not limited to document trees.

   The protocol supports the session following operations:

   Retrieval of media from media server:
          The client can request a presentation description format SDP.

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

   Request: An RTSP request. or
          some other method. 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 consist of a client setting up a
        transport mechanism for presentation is being multicast, the continuous media stream ( SETUP),
        starting
          presentation description contains the stream with  PLAY or  RECORD multicast addresses and closing
          ports to be used for the stream
        with  TEARDOWN.

1.4 Protocol Properties

   RTSP has continuous media. If the following properties:

   Extendable: New methods and parameters can presentation
          is to be easily added sent only to RTSP.

   Easy the client via unicast, the client
          provides the destination for security reasons.

H. Schulzrinne, A. Rao, R. Lanphier                            Page  5
   Invitation of a media server to parse: RTSP a conference:
          A media server can be parsed by standard HTTP or MIME parsers.

   Secure: RTSP re-uses web security mechanisms, ``invited'' to join an existing
          conference, either at to play back media into the transport
        level (TLS [7]) presentation or
          to record all or within the protocol itself.  All HTTP
        authentication mechanisms such as basic [5] and digest
        authentication [8] are directly applicable.

   Transport-independent: RTSP may use either an unreliable datagram
        protocol (UDP) [9], a reliable datagram protocol (RDP, not
        widely used [10]) or a reliable stream protocol such as TCP [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 subset of the
        different media servers.  Media synchronization in a presentation. This
          mode 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 useful for distributed teaching applications. Several
          parties in the two modes ("VCR").

   Separation of stream control and conference initiation: Stream may take turns ``pushing the remote
          control is divorced from inviting a buttons''.

   Addition of media server to a
        conference. The only requirement an existing presentation:
          Particularly for live presentations, it is that useful if the conference
        initiation protocol either provides or can be used to create a
        unique conference identifier. In particular, SIP [12] or H.323
        may be used to invite a server to a conference.

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

   Presentation description neutral: The protocol does not impose a
        particular presentation description or metafile format and
          can
        convey the type of format to be used. However, tell the presentation
        description must contain at least one client about additional media becoming available.

   RTSP URI.

   Proxy and firewall friendly: The protocol should requests may be readily handled by both application proxies, tunnels and transport-layer (SOCKS [13]) firewalls.
        A firewall may need to understand the  SETUP method caches as in
   HTTP/1.1.

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 open a
        "hole" for the UDP media stream.

   HTTP-friendly: Where sensible, RTSP re-uses HTTP concepts, so that
        the existing infrastructure can be re-used. This infrastructure
        includes JEPI (the Joint Electronic Payment Initiative) for
        electronic payments and PICS (Platform for Internet Content
        Selection) for associating labels with content. However, RTSP
        does not just add methods to HTTP, since the controlling
        continuous media requires server state interpreted as described in most cases.

   Appropriate server control: If a client can start a stream, it must
        be able to stop a stream. Servers should
   RFC 2119 [4].

1.3 Terminology

   Some of the terminology has been adopted from HTTP/1.1 [5]. Terms not start streaming to
        clients
   listed here are defined as in such HTTP/1.1.

   Conference:
          a way that clients cannot stop the stream.

   Transport negotiation: multiparty, multimedia presentation, where ``multi'' implies
          greater than or equal to one.

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

   Connection:
          A transport method
        prior to actually needing to process layer virtual circuit established between two
          programs for the purpose of communication.

H. Schulzrinne, A. Rao, R. Lanphier                            Page  6
   Continuous media:
          Data where there is a timing relationship between source and
          sink, that is, the sink must reproduce the timing relationshop
          that existed at the source. The most common examples of
          continuous media stream.

   Capability negotiation: If basic features are disabled, there must audio and motion video. Continuous media
          can be
        some clean mechanism for realtime (interactive), where there is a ``tight''
          timing relationship between source and sink, or streaming
          (playback), where the client to determine which methods relationship is less strict.

   Participant:
          Participants are not going to members of conferences. A participant may 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
          machine, e.g., 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. media record or playback server.

   Media server:
          The protocol would not
        address how several clients negotiate access; this is left
        to either a "social protocol" network entity providing playback or some other floor control
        mechanism.

1.5 Extending RTSP

   Since not all recording services for
          one or more media servers have the same functionality, streams. Different media
   servers by necessity will support streams within a
          presentation may originate from different sets of requests. For
   example:

        o media servers. A
          media server may only be capable of playback, not recording and
         thus has no need to support reside on the  RECORD request.

        o A same or a different host as the
          web server may not be capable of seeking (absolute positioning),
         say, if it the presentation is invoked from.

   Media parameter:
          Parameter specific to support live events only.

        o Some servers a media type that may not support setting be changed while
          the stream parameters and
         thus not support  GET_PARAMETER and  SET_PARAMETER.

   A server SHOULD implement all header fields described in Section 11.

   It is up to the creators of presentation descriptions not being played or prior to ask the
   impossible of it.

   (Media) stream:
          A single media instance, e.g., an audio stream or a server. This situation is similar in HTTP/1.1, where
   the methods described in [H19.6] are not likely to be supported
   across all servers.

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

        o Existing methods can be extended with new parameters, video
          stream as long well as these parameters can be safely ignored a single whiteboard or shared application
          group. When using RTP, a stream consists of all RTP and RTCP
          packets created by the recipient.
         (This a source within an RTP session. This is
          equivalent to adding new parameters to an HTML tag.)

        o New methods can be added. If the recipient definition of the message does
         not understand the request, it responds with error code  501
         (Not implemented) and the sender can then attempt an earlier,
         less functional version.

        o A new version a DSM-CC stream([18]).

   Message:
          The basic unit of the protocol can be defined, allowing almost
         all aspects (except the position RTSP communication, consisting of a
          structured sequence of octets matching 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 syntax defined in
          Section 15 and the properties transmitted via a connection or a connectionless
          protocol.

   Presentation:
          A set of one or more streams which the media server allows the client
          to manipulate together. A presentation is made up of has a single time axis
          for all streams belonging to it. Presentations are defined by a
          presentation description
   file, the format of which is outside the scope of this specification.
   The descriptions (see below). A presentation
          description file may contains RTSP URIs that define which streams can be obtained by the client using
   HTTP or other means such as email
          controlled individually and may not necessarily be stored
   on the media server.

   For an RTSP URI to control the purposes whole
          presentation. A movie or live concert consisting 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
          audio and
   without loss of generality, it video streams is assumed that the presentation
   description contains exactly one such an example of a presentation.

H. Schulzrinne, A. Rao, R. Lanphier                            Page  7
   Presentation description:
          A presentation
   may contain several media streams.

   The presentation description file contains a description of the information about one or
          more media streams making up the within a presentation, including their such as the set of
          encodings,
   language, network addresses and other parameters that enable information about the client to choose content.
          Other IETF protocols such as SDP [6] use the
   most appropriate combination of media. In this term ``session''
          for a live presentation. The presentation
   description, each media stream 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 individually controllable by
          indicated explicitly.

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

   RTSP URL, which points to session:
          A complete RTSP ``transaction'', e.g., the media server
   handling that particular viewing of a movie.
          A session typically consists of a client setting up a transport
          mechanism for the continuous media stream and names (SETUP), starting the
          stream stored on
   that server.  Several media streams can be located on different
   servers; for example, audio with PLAY or RECORD and video streams can be split across
   servers for load sharing.  The description also enumerates which
   transport methods the server is capable of.

   Besides closing the media parameters, stream with
          TEARDOWN.

1.4 Protocol Properties

   RTSP has the network destination address following properties:

   Extendable:
          New methods and
   port need to be determined. Several modes of operation parameters can be
   distinguished:

   Unicast: The media is transmitted easily added to the source of the RTSP.

   Easy to parse:
          RTSP request,
        with the port number chosen can be parsed by standard HTTP or MIME parsers.

   Secure:
          RTSP re-uses web security mechanisms, either at the client. Alternatively, the
        media is transmitted on transport
          level (TLS [7]) or within the same protocol itself. All HTTP
          authentication mechanisms such as basic [5, Section 11.1] and
          digest authentication [8] are directly applicable.

   Transport-independent:
          RTSP may use either an unreliable datagram protocol (UDP) [9],
          a reliable datagram protocol (RDP, not widely used [10]) or a
          reliable stream protocol such as RTSP.

   Multicast, server chooses address: The TCP [11] as it implements
          application-level reliability.

H. Schulzrinne, A. Rao, R. Lanphier                            Page  8
   Multi-server capable:
          Each media server picks the
        multicast address and port. This is the typical case for stream within a live
        or near-media-on-demand transmission.

   Multicast, presentation can reside on a
          different server. The client chooses address: If automatically establishes several
          concurrent control sessions with the server different media servers.
          Media synchronization is to participate in
        an existing multicast conference, performed at the multicast address, port transport level.

   Control of recording devices:
          The protocol can control both recording and encryption key are given by the conference description,
        established by means outside playback devices,
          as well as devices that can alternate between the scope two modes
          (``VCR'').

   Separation 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 and conference initiation:
          Stream control may
   occur on a TCP connection while the data flows via UDP. Thus, data
   delivery continues even if no RTSP requests are received by the media
   server.  Also, during its lifetime, is divorced from inviting a single media stream may be
   controlled by RTSP requests issued sequentially on different TCP
   connections.  Therefore, the server needs to maintain "session state" 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
          may be able used to correlate RTSP requests with invite a stream. The state
   transitions are described in Section A.

   Many methods in server to a conference.

   Suitable for professional applications:
          RTSP do not contribute supports frame-level accuracy through SMPTE time stamps to state. However, the
   following play
          allow remote digital editing.

   Presentation description neutral:
          The protocol does not impose a central role in defining the allocation particular presentation
          description or metafile format and usage can convey the type of
   stream resources on
          format to be used. However, the server:  SETUP,  PLAY,  RECORD, PAUSE, presentation description must
          contain at least one RTSP URI.

   Proxy and
   TEARDOWN.

   SETUP: Causes firewall friendly:
          The protocol should be readily handled by both application and
          transport-layer (SOCKS [13]) firewalls. A firewall may need to
          understand the server SETUP method to allocate resources for open a stream and start
        an ``hole'' for the UDP
          media stream.

   HTTP-friendly:
          Where sensible, 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 re-uses HTTP concepts, so that the stream. The
          existing infrastructure can be re-used. This infrastructure
          includes PICS (Platform for Internet Content Selection [20])
          for associating labels with content. However, RTSP
        session ceases does not
          just add methods to exist on HTTP, since the server.

1.8 Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP. It also may
   interact with HTTP controlling continuous
          media requires server state in that the initial contact with 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 content
   is often to be made through clients in such a web page.
          way that clients cannot stop the stream.

H. Schulzrinne, A. Rao, R. Lanphier                            Page  9
   Transport negotiation:
          The current protocol
   specification aims client can negotiate the transport method prior to allow different hand-off points between actually
          needing to process a web
   server and the continuous media server implementing RTSP. For example, the
   presentation description can stream.

   Capability negotiation:
          If basic features are disabled, there must be retrieved using HTTP or RTSP. Having some clean
          mechanism for the presentation description client to determine which methods are not
          going to be returned by implemented. This allows clients to present the web server makes
          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
   possible was determined that a better approach was to have make sure
     that the web server take care of authentication and
   billing, protocol is easily extensible to the multi-client
     scenario. Stream identifiers can be used by handing out a presentation description whose media
   identifier includes an encrypted version of several control
     streams, so that ``passing the requestor's IP remote'' would be possible. The
     protocol would not address and a timestamp, with how several clients negotiate access;
     this is left to either a shared secret between web and media
   server.

   However, ``social protocol'' or some other floor
     control mechanism.

1.5 Extending 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

   Since not all media servers have the same functionality, media client servers
   by necessity will support different sets of requests. For example:
     * A server may only be capable of playback, not recording and media thus
       has no need to support the RECORD request.
     * A server can issue
   requests. RTSP requests are also may not stateless, in that they be capable of seeking (absolute positioning),
       say, if it is to support live events only.
     * Some servers may set not support setting stream parameters and continue thus
       not support GET_PARAMETER and SET_PARAMETER.

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

   It is up to control the creators of presentation descriptions not to ask the
   impossible of a media stream long after server. This situation is similar in HTTP/1.1, where
   the
   request has been acknowledged.

        Re-using HTTP functionality has advantages methods described in at least two
        areas, namely security and proxies. The requirements [H19.6] 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 likely to RTP. be supported across
   all servers.

   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 be extended in three ways, listed in order of the definitions and syntax are identical to HTTP/1.1,
   this specification only points to magnitude
   of changes supported:

     * Existing methods can be extended with new parameters, as long as
       these parameters can be safely ignored by the section where they are defined
   rather than copying it. For brevity, [HX.Y] recipient. (This is
       equivalent to be taken to refer adding new parameters to Section X.Y an HTML tag.)
     * New methods can be added. If the recipient of the current HTTP/1.1 specification (RFC 2068).

   All message does not
       understand the mechanisms specified in this document are described in both
   prose request, it responds with error code 501 (Not
       implemented) and an augmented Backus-Naur form (BNF) similar the sender should not attempt to that used in
   RFC 2068 [H2.1]. It is described in detail in [14].

   In use this draft, we method
       again. A client may also use indented and smaller-type paragraphs the OPTIONS method to provide
   background and motivation. Some of these paragraphs are marked with
   HS, AR and RL, designating opinions and comments inquire about
       methods supported by the individual
   authors which may not be shared by server. The server SHOULD list the co-authors and require
   resolution.

3 Protocol Parameters

3.1 RTSP Version

   applies, with HTTP replaced
       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 RTSP.

3.2 an RTSP URL URL.
   The "rtsp" overall presentation and "rtspu" schemes are used to refer to network resources
   via the RTSP protocol. This section defines properties of the scheme-specific
   syntax and semantics for RTSP URLs.

     rtsp_URL = ( "rtsp:" | "rtspu:" ) "//" host [ ":" port ] [abs_path]
     host     = <A legal Internet host domain name media the
   presentation is made up of IP address
                (in dotted decimal form), as are defined by Section 2.1
                of RFC 1123>
     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 presentation description
   file, the interpretation
        left to format of which is outside the RTSP server. scope of this specification.
   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 presentation description file may be obtained by the port is empty client using
   HTTP or other means such as email and may not given, port 554 is assumed. The semantics
   are that the identified resource can be controlled necessarily be RTSP at the
   server listening for TCP (scheme "rtsp") connections or UDP (scheme
   "rtspu") packets stored on that port of host , and
   the Request-URI for media server.

   For the
   resource is rtsp_URL

   The use purposes of IP addresses in URLs SHOULD be avoided whenever possible
   (see RFC 1924 [15]).

   A this specification, a presentation description is
   assumed to describe one or more presentations, each of which maintains
   a stream is identified by an textual media
   identifier, using the character set common time axis. For simplicity of exposition and escape conventions [H3.2] without loss of
   URLs [16]. Requests described in Section 9 can refer to either
   generality, it is assumed that the
   whole presentation or an individual stream within the description contains
   exactly one such presentation.
   Note that some methods can only be applied to streams, not
   presentations and vice versa. A specific instance of a presentation
   or stream, e.g., one of may contain several concurrent transmissions
   media streams.

   The presentation description file contains a description of the same
   content, an 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 session , is indicated identified by the Session header field
   (Section 11.26) where needed.

   For example, the an RTSP URL

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

   identifies URL, which points to the audio media server
   handling that particular media stream within and names the presentation "twister", which stream stored on
   that server. Several media streams can be controlled via RTSP requests issued over a TCP connection to
   port 554 of host media.example.com

        This does not imply a standard way to reference located on different
   servers; for example, audio and video streams in
        URLs. can be split across
   servers for load sharing. The presentation description defines also enumerates which
   transport methods the hierarchical
        relationships in server is capable of.

   Besides the presentation and media parameters, the URLs for the
        individual streams. A presentation description may name a
        stream 'a.mov' network destination address and the whole presentation 'b.mov'.

   The path components port
   need to be determined. Several modes of the RTSP URL are opaque operation can be
   distinguished:

   Unicast:
          The media is transmitted to the client and do
   not imply any particular file system structure for source of the server.

        This decoupling also allows presentation descriptions to be
        used RTSP request,
          with non-RTSP media control protocols, simply the port number chosen by
        replacing the scheme in client. Alternatively, the URL.

3.3 Conference Identifiers

   Conference identifiers are opaque to RTSP and are encoded using
   standard URI encoding methods (i.e., LWS
          media is escaped with %). They can
   contain any octet value. transmitted on the same reliable stream as RTSP.

   Multicast, server chooses address:
          The conference identifier MUST be globally
   unique. For H.323, media server picks the conferenceID value multicast address and port. This is to be used.

     conference-id = 1*OCTET  ; LWS must be URL-escaped

        Conference identifiers are used to allow to allow RTSP
        sessions to obtain parameters from multimedia conferences
          the media typical case for a live or near-media-on-demand
          transmission.

   Multicast, client chooses address:
          If the server is participating in. These conferences to participate in an existing multicast
          conference, the multicast address, port and encryption key are
        created
          given by protocols the conference description, established by means
          outside the scope of this
        specification, e.g., H.323 [17] or SIP [12]. Instead specification.

1.7 RTSP States

     RTSP controls a stream which may be sent via a separate protocol,
   independent of the
        RTSP client explicitly providing transport information, for control channel. For example, it asks RTSP control may
   occur on a TCP connection while the data flows via UDP. Thus, data
   delivery continues even if no RTSP requests are received by the media
   server. Also, during its lifetime, a single media stream may be
   controlled by RTSP requests issued sequentially on different TCP
   connections. Therefore, the server needs to use 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 values
   following play a central role in defining the
        conference description instead. If allocation and usage of
   stream resources on the conference
        participant inviting server: SETUP, PLAY, RECORD, PAUSE, and
   TEARDOWN.

   SETUP:
          Causes the media server would only supply a
        conference identifier which is unique to allocate resources for that inviting
        party, the media server could add a stream and start
          an internal identifier
        for that party, e.g., its Internet address. However, this
        would prevent that the conference participant 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
        initiator of the stream. The RTSP commands are two different entities.

3.4 SMPTE Relative Timestamps

   A SMPTE relative time-stamp expresses time relative session
          ceases to exist on the start of
   the clip. Relative timestamps are expressed as SMPTE time codes for
   frame-level access accuracy. The time code server.

1.8 Relationship with Other Protocols

   RTSP has the format
                       hours:minutes:seconds.frames
                                     , some overlap in functionality with HTTP. It also may interact
   with HTTP in that the origin at the start of the clip. For NTSC, the frame rate initial contact with streaming content is
   29.97 frames per second. This is handled by dropping 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 first frame
   index of every minute, except every tenth minute. If media
   server implementing RTSP. For example, the frame value
   is zero, it may presentation description
   can be omitted.

     smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ]
     smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ "." 1*2DIGIT ]

   Examples:

     smpte=10:12:33.40-
     smpte=10:7:33-
     smpte=10:7:0-10:7:33

3.5 Normal Play Time

   Normal play time (NPT) indicates retrieved using HTTP or RTSP. Having the stream absolute position
   relative presentation
   description be returned by the web server makes it possible to have
   the beginning web server take care of the presentation, measured in seconds authentication and microseconds. The beginning of billing, by handing out
   a presentation corresponds to 0
   seconds description whose media identifier includes an
   encrypted version of the requestor's IP address and 0 microseconds. Negative values are not defined. The
   microsecond field is always less than 1,000,000. NPT is defined as a timestamp, with
   a shared secret between web and media server.

   However, RTSP differs fundamentally from HTTP in
   DSM-CC:  "Intuitively, NPT that data delivery
   takes place out-of-band, in a different protocol. HTTP is an
   asymmetric protocol, where the clock client issues requests and the viewer associates with a
   program.  It is often digitally displayed on a VCR. NPT advances
   normally when server
   responds. In RTSP, both the media client and media server can issue
   requests. RTSP requests are also not stateless, in normal play mode (scale = 1), advances at that they may set
   parameters and continue to control a faster
   rate when in fast scan forward (high positive scale ratio),
   decrements when media stream long after the
   request has been acknowledged.

     Re-using HTTP functionality has advantages in scan reverse (high negative scale ratio) at least two areas,
     namely security and is
   fixed in pause mode. NPT is [logically] equivalent proxies. The requirements are very similar, so
     having the ability to SMPTE time
   codes." [18]

     npt-range = "npt" "=" npt-time "-" [ npt-time ]
     npt-time  = 1*DIGIT [ ":" *DIGIT ]

   Examples:

     npt=123:45-125

3.6 Absolute Time

   Absolute time adopt HTTP work on caches, proxies and
     authentication is expressed valuable.

   While most real-time media will use RTP 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

   Example

4 RTSP Message transport protocol, RTSP
   is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2044). Lines are terminated by CRLF, but
   receivers should be prepared not tied to also interpret CR RTP.

   RTSP assumes the existence of a presentation description format that
   can express both static and LF by
   themselves as line terminators.

        Text-based protocols make it easier to add optional
        parameters in temporal properties of a self-describing manner. presentation
   containing several media streams.

2 Notational Conventions

   Since the number many of
        parameters the definitions and syntax are identical to HTTP/1.1,
   this specification only points to the frequency of commands is low, processing
        efficiency section where they are defined
   rather than copying it. For brevity, [HX.Y] is not a concern. Text-based protocols, if done
        carefully, also allow easy implementation to be taken to refer to
   Section X.Y of research
        prototypes the current HTTP/1.1 specification (RFC 2068).

   All the mechanisms specified in scripting languages such as Tcl, Visual Basic this document are described in both
   prose and Perl.

   The 10646 character set avoids tricky character set switching, but is
   invisible an augmented Backus-Naur form (BNF) similar to the application as long as US-ASCII is being used. This
   is also the encoding that used for RTCP. ISO 8859-1 translates directly
   into Unicode, with a high-order octet in
   RFC 2068 [H2.1]. It is described in detail in [14].

   In this draft, we use indented and smaller-type paragraphs to provide
   background and motivation. Some of zero. ISO 8859-1 characters these paragraphs are marked with
   HS, AR and RL, designating opinions and comments by the most-significant bit set are represented as 1100001x
   10xxxxxx.

   RTSP messages can individual
   authors which may not be carried over any lower-layer transport protocol
   that is 8-bit clean.

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

4.1 Message Types

   See [H4.1]

4.2 Message Headers

   See [H4.2]

4.3 Message Body

   See [H4.3]

4.4 Message Length

   When a message-body is included
   resolution.

3 Protocol Parameters

3.1 RTSP Version

   [H3.1] applies, with a message, the length of that
   body is determined HTTP replaced by one of RTSP.

3.2 RTSP URL

     The ``rtsp'', ``rtspu'' and ``rtsps'' schemes are used to refer to
   network resources via the following (in order of precedence):

        1.   Any response message which MUST NOT include a message-body
             (such as RTSP protocol. This section defines the 1xx, 204,
   scheme-specific syntax and 304 responses) is always
             terminated semantics for RTSP URLs.

  rtsp_URL = ( "rtsp:" | "rtspu:" | "rtsps:" )
                "//" host [ ":" port ] [abs_path]
  host     = <A legal Internet host domain name of IP address
             (in dotted decimal form), as defined by the first empty line after the header fields,
             regardless Section 2.1
             of the entity-header fields present RFC 1123>
  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
             message. (Note: An empty line consists of only CRLF.)

        2.   If interpretation left to the RTSP
     server.

   The scheme rtsp requires that commands are issued via a  Content-Length header field (section 11.12) is
             present, its value in bytes represents reliable
   protocol (within the length of Internet, TCP), while the
             message-body. scheme rtspu identifies
   an unreliable protocol (within the Internet, UDP). The scheme rtsps
   indicates that a TCP connection secured by TLS [7] must be used.

   If this header field the port is empty or not present, a value
             of zero given, port 554 is assumed.

        3.   By the server closing the connection. (Closing The semantics
   are that the
             connection cannot identified resource can be used to indicate controlled be RTSP at the end of a request
             body, since
   server listening for TCP (scheme ``rtsp'') connections or UDP (scheme
   ``rtspu'') packets on that would leave no possibility port of host, and the Request-URI for the server
             to send back
   resource is rtsp_URL.

   The use of IP addresses in URLs SHOULD be avoided whenever possible
   (see RFC 1924 [15]).

   A presentation or a response.)

   Note that RTSP does not (at present) support stream is identified by an textual media
   identifier, using the HTTP/1.1 "chunked"
   transfer coding character set and requires the presence escape conventions [H3.2] of the  Content-Length
   header field.

        Given the moderate length
   URLs [16]. URLs may refer to a stream or an aggregate of streams ie. a
   presentation. Accordingly, requests described in Section 10 can apply
   to either the whole presentation descriptions
        returned, or an individual stream within the server should always
   presentation. Note that some request methods can only be able applied to determine its
        length, even if it is generated dynamically, making
   streams, not presentations and vice versa.

   For example, the
        chunked transfer encoding unnecessary. Even though
        Content-Length must be present if there is any entity body, RTSP URL
  rtsp://media.example.com:554/twister/audiotrack

   identifies the rules ensure reasonable behavior even if audio stream within the length is
        not given explicitly.

5 Request

   A request message from presentation ``twister'', which
   can be controlled via RTSP requests issued over a client TCP connection to a server or vice versa includes,
   within the first line
   port 554 of that message, host media.example.com.

   Also, the method to RTSP URL
  rtsp://media.example.com:554/twister

   identifies the presentation ``twister'', which may be applied composed of
   audio and video streams.

     This does not imply a standard way to reference streams in URLs.
     The presentation description defines the resource, hierarchical relationships
     in the identifier of presentation and the resource, URLs for the individual streams. A
     presentation description may name a stream 'a.mov' and the protocol
   version in use.

     Request = Request-line CRLF
               *request-header
               CRLF
               [ message-body ]

     Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no CRLF

     Method = "DESCRIBE"        ; Section
            | "GET_PARAMETER"   ; Section
            | "OPTIONS"         ; Section
            | "PAUSE"           ; Section
            | "PLAY"            ; Section
            | "RECORD"          ; Section
            | "REDIRECT"        ; Section
            | "SETUP"           ; Section
            | "SET_PARAMETER"   ; Section
            | "TEARDOWN"        ; Section
            | extension-method

     extension-method = token

     Request-URI = "*" | absolute_URI

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

     seq-no = 1*DIGIT

   Note that in contrast to HTTP/1.1, RTSP requests always contain whole
     presentation 'b.mov'.

   The path components of the
   absolute RTSP URL (that is, including are opaque to the scheme, host client and port) rather
   than just the absolute path.

   The asterisk "*" in the Request-URI means that the request does do
   not
   apply to a imply any particular resource, but to the server itself, and is only
   allowed when file system structure for the method used does not necessarily apply server.

     This decoupling also allows presentation descriptions to a
   resource.  One example would be

     OPTIONS * RTSP/1.0

6 Response

   [H6] applies except that HTTP-Version is replaced by RTSP-Version
   define some HTTP codes. The valid response codes and the methods they
   can be used
     with are defined non-RTSP media control protocols, simply by replacing the
     scheme in the table 1.

   After receiving URL.

3.3 Conference Identifiers

     Conference identifiers are opaque to RTSP and interpreting a request message, the recipient
   responds are encoded using
   standard URI encoding methods (i.e., LWS is escaped with an RTSP response message.

     Response %). 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 = Status-Line             ; Section
                *( general-header       ; Section
                 | response-header      ; Section
                 | entity-header )      ; Section
                CRLF
                [ message-body ] 1*OCTET  ; Section

6.1 Status-Line

   The first line of a Response message is the Status-Line , consisting
   of LWS must be URL-escaped

     Conference identifiers are used to allow to allow RTSP sessions to
     obtain parameters from multimedia conferences the protocol version followed media server is
     participating in. These conferences are created by a numeric status code, protocols
     outside the
   sequence number scope of the corresponding request and the textual phrase
   associated with the status code, with each element separated by SP
   characters. No CR this specification, e.g., H.323 [17] or LF is allowed except in the final CRLF sequence.
   Note that the addition of a

     Status-Line = RTSP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF

6.1.1 Status Code and Reason Phrase

   The Status-Code element is a 3-digit integer result code SIP
     [12]. Instead of the
   attempt RTSP client explicitly providing transport
     information, for example, it asks the media server to understand and satisfy use the request. These codes are fully
   defined
     values in section10. The Reason-Phrase is intended to give a short
   textual the conference description of instead. If the Status-Code. The Status-Code conference
     participant inviting the media server would only supply a
     conference identifier which is intended unique for use by automata and that inviting party, the Reason-Phrase is intended
     media server could add an internal identifier for that party, e.g.,
     its Internet address. However, this would prevent that the human
   user. The client is not required to examine or display
     conference participant and the Reason-
   Phrase

   The first digit initiator of the Status-Code defines the class of response. The
   last RTSP commands are
     two digits do not have any categorization role. There different entities.

3.4 Session Identifiers

     Session identifiers are 5
   values for the first digit:

        o 1xx: Informational - Request received, continuing process

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

        o 3xx: Redirection - Further action opaque strings of arbitrary length. Linear
   white space must be taken in order URL-escaped. A session identifier SHOULD be chosen
   randomly and SHOULD be at least eight octets long to
         complete the request

        o 4xx: Client Error - The request contains bad syntax or cannot make guessing it
   more difficult. (See Section 16).

  session-id = 1*OCTET      ; LWS must be fulfilled

        o 5xx: Server Error - The server failed URL-escaped

3.5 SMPTE Relative Timestamps

     A SMPTE relative time-stamp expresses time relative to fulfill an apparently
         valid request

   The individual values the start of
   the numeric status clip. Relative timestamps are expressed as SMPTE time codes defined for
   RTSP/1.0, and an example set of corresponding Reason-Phrase below.
   frame-level access accuracy. The reason phrases listed here are only recommended -- they may be
   replaced time code has the format
   hours:minutes:seconds:frames.subframes, with the origin at the start
   of the clip. For NTSC, the frame rate is 29.97 frames per second. This
   is handled by local equivalents without affecting dropping the protocol. Note
   that RTSP adopts most HTTP/1.1 status codes first two frame indices (values 00 and adds RTSP-specific
   status codes in 01)
   of every minute, except every tenth minute. If the starting at 450 to avoid conflicts with newly
   defined HTTP status codes.

      Status-Code frame value is
   zero, it may be omitted. Subframes are measured in one-hundredth of a
   frame.

  smpte-range = "100"   ; Continue
                     | "200"   ; OK
                     | "201"   ; Created
                     | "smpte" "=" smpte-time "-" [ smpte-time ]
  smpte-time = 2DIGIT ":" 2DIGIT ":" 2DIGIT [ ":" 2DIGIT ] [ "." 2DIGIT]

   Examples:
  smpte=10:12:33:20-
  smpte=10:07:33-
  smpte=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, measured in seconds and
   microseconds. The beginning of a presentation corresponds to 0 seconds
   and 0 microseconds. Negative values are not defined. The microsecond
   field is always less than 1,000,000. NPT is defined as in DSM-CC [18]:
   ``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.'' [18]

  npt-range = "npt" "=" npt-time "-" [ npt-time ]
  npt-time  = 1*DIGIT [ ":" *DIGIT ]

   Examples:
  npt=123:45-125

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

   Example

4 RTSP Message

     RTSP is a text-based protocol and uses the ISO 10646 character set
   in UTF-8 encoding (RFC 2044). 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.

   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]

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 12.15) 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 12.10
                         |     Connection        ; Section 12.12
                         |     Date              ; Section 12.17
                         |     Via               ; Section 12.39

6 Request

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

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

6.1 Request Line

  Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no 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

  seq-no = 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 12.19
                  |          If-Modified-Since        ; Section 12.21
                  |          Range                    ; Section 12.28
                  |          Referer                  ; Section 12.29
                  |          User-Agent               ; Section 12.37
   Note that in contrast to HTTP/1.1, RTSP requests always contain the
   absolute URL (that is, including the scheme, host and port) rather
   than just the absolute path.

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

   The asterisk "*" in the Request-URI means that the request does not
   apply to a particular resource, but to the server 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 the 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, the
   sequence number of the corresponding request 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 seq-no 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 section11. The Reason-Phrase is intended to give a short
   textual description of the Status-Code. The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human
   user. The client is not required to examine or display the
   Reason-Phrase.

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

     * 1xx: Informational - Request received, continuing process
     * 2xx: Success - The action was successfully received, understood,
       and accepted
     * 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 status codes and adds
   RTSP-specific status codes in the starting at 450 to avoid conflicts
   with newly defined HTTP status codes.

   Status-Code    = "100"   ; Continue
                  | "200"   ; OK
                  | "201"   ; Created
                  | "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
                  | "409"   ; Conflict
                  | "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
                  | "45x"   ; Session Not Found
                  | "45x"   ; Method Not Valid in This State
                  | "45x"   ; Header Field Not Valid for Resource
                  | "45x"   ; Invalid Range
                  | "45x"   ; Parameter Is Read-Only
                  | "45x"   ; Aggregate operation not allowed
                  | "45x"   ; Only aggregate operation allowed
                  | "500"   ; Internal Server Error
                  | "501"   ; Not Implemented
                  | "502"   ; Bad Gateway
                  | "503"   ; Service Unavailable
                  | "504"   ; Gateway Time-out
                  | "505"   ; HTTP RTSP Version not supported
                  | extension-code
   extension-code = 3DIGIT

   Reason-Phrase  = *<TEXT, excluding CR, LF>

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

6.1.2 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 server and about further access to the resource
   identified by the Request-URI

     response-header = Location              ; Section
                       | Proxy-Authenticate  ; Section
                       | Public              ; Section
                       | Retry-After         ; Section
                       | Server              ; Section
                       | Vary                ; Section
                       | WWW-Authenticate    ; Section

   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.

7 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 agents SHOULD
   present to the entity-headers.

   In this section, both sender and recipient refer user the entity returned with the response, since that
   entity is likely to either include human-readable information which will
   explain the client unusual status.

   Code    reason
      _______________________________________________________________          Reason
   100           Continue                         all

_______________________________________________________________
   200           OK                               all
   201           Created                          RECORD
      _______________________________________________________________
   300           Multiple Choices                 all
   301           Moved Permanently                all
   302           Moved Temporarily                all
   303           See Other                        all
   305           Use Proxy                        all

_______________________________________________________________
   400           Bad Request                      all
   401           Unauthorized                     all
   402           Payment Required                 all
   403           Forbidden                        all
   404           Not Found                        all
   405           Method Not Allowed               all
   406           Not Acceptable                   all
   407           Proxy Authentication Required    all
   408           Request Timeout                  all
   409           Conflict                         RECORD
   410           Gone                             all
   411           Length Required                  SETUP
   412           Precondition Failed              DESCRIBE, SETUP
   413           Request Entity Too Large         SETUP
   414           Request-URI Too Long             all
   415           Unsupported Media Type           SETUP
   45x     Only Valid for Stream            SETUP           Session not found                all
   45x           Invalid parameter                SETUP
   45x           Not Enough Bandwidth             SETUP
   45x           Illegal Conference Identifier    SETUP
   45x           Illegal Session Identifier       PLAY, RECORD, TEARDOWN
   45x           Parameter Is Read-Only           SET_PARAMETER
   45x           Header Field Not Valid           all
      _______________________________________________________________
   45x           Method Not Valid In This State   all
   45x           Aggregate operation not allowed  all
   45x           Only aggregate operation allowed  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
!
   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.

   response-header   =     Location            ; Section 12.23
                     |     Proxy-Authenticate  ; Section 12.26
                     |     Public              ; Section 12.27
                     |     Retry-After         ; Section 12.30
                     |     Server              ; Section 12.33
                     |     Vary                ; Section 12.38
                     |     WWW-Authenticate    ; Section 12.40

   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.

7.1

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 14.7 12.4
                         |    Content-Encoding    ; Section 14.12 12.13
                         |    Content-Language    ; Section 14.13 12.14
                         |    Content-Length      ; Section 14.14 12.15
                         |    Content-Type        ; Section 14.18 12.16
                         |    Expires             ; Section 14.21 12.18
                         |    Last-Modified       ; Section 14.29 12.22
                         |    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.

7.2

8.2 Entity Body

   See [H7.2]

8

9 Connections

     RTSP requests can be transmitted in several different ways:

        o

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

        o
     * one connection per request/response transaction;

        o
     * connectionless mode.

   The type of transport connection is defined by the RTSP URI
   (Section 3.2). For the scheme "rtsp", ``rtsp'', a persistent connection is
   assumed, while the scheme "rtspu" ``rtspu'' calls for RTSP requests to be send
   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.

8.1

9.1 Pipelining

   A client that supports persistent connections or connectionless mode
   MAY "pipeline" ``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.

8.2

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 TBD), 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, the timeout value MAY be set
   to an arbitrarily large value.

     This can greatly increase responsiveness for proxies operating in
     local-area networks with small RTTs. The mechanism is defined such
     that the client implementation does not have be aware of whether a
     reliable or unreliable transport protocol is being used. It is
     probably a bad idea to have two reliability mechanisms on top of
     each other, although the RTSP RTT estimate is likely to be larger
     than the TCP estimate.

   Each request carries a sequence number, which is incremented by one
   for each request transmitted. If a request is repeated because of lack
   of acknowledgement, the sequence number is incremented.

     This avoids ambiguities when computing round-trip time estimates.

   [TBD: An initial sequence number negotiation needs to be added for
   UDP; otherwise, a new stream connection may see a request be
   acknowledged by a delayed response from an earlier "connection". ``connection''.
   This handshake can be avoided with a sequence number containing a
   timestamp of sufficiently high resolution.]

   The reliability mechanism described here does not protect against
   reordering. This may cause problems in some instances. For example, a
   TEARDOWN followed by a PLAY has quite a different effect than the
   reverse. Similarly, if a PLAY request arrives before all parameters
   are set due to reordering, the media server would have to issue an
   error indication. Since sequence numbers for retransmissions are
   incremented (to allow easy RTT estimation), the receiver cannot just
   ignore out-of-order packets. [TBD: This problem could be fixed by
   including both a sequence number that stays the same for
   retransmissions and a timestamp for RTT estimation.]

   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 method header message MUST contain a Content-Length header whenever
   that method message contains a payload. Otherwise, an RTSP packet is
   terminated with an empty line immediately following the method last message
   header.

9

10 Method Definitions

     The method token indicates the method to be performed on the
   resource identified by the Request-URI 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 token. Methods are
   summarized in Table 2.

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

9.1

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 non-standard request. It does not influence server state.

   In addition, if the optional Require header is present, option tags
   within the header indicate features needed by the requestor that are
   not required at the version level of the protocol.

   Example 1: :
  C->S:  OPTIONS * RTSP/1.0 1
            Require: implicit-play, record-feature
            Transport-Require: switch-to-udp-control, gzipped-messages

   Note that these are fictional features (though we may want to make
   them real one day).

   Example 2 (using RFC2069-style authentication only as an example):

     S->C: OPTIONS * RTSP/1.0 1
           Authenticate: Digest realm="testrealm@host.com",
             nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
             opaque="5ccc069c403ebaf9f0171e9517f40e41"
         PEP: {{map "http://www.iana.org/rtsp/implicit-play"}}
              {{map "http://www.iana.org/rtsp/record-feature"}}
         C-PEP: {{map "http://www.iana.org/rtsp/udp-control"}}
                {{map "http://www.iana.org/rtsp/gzipped-messages"}}

  S->C:  RTSP/1.0 200 1 2 OK
           Date: 23 Jan 1997 15:35:06 GMT
           Nack-Transport-Require: switch-to-udp-control
         PEP-Info: {{map "http://www.iana.org/rtsp/implicit-play"}
                    {for "/" *}}
                   {{map "http://www.iana.org/rtsp/record-feature"}
                    {for "/" *}}
         C-PEP-Info: {{map "http://www.iana.org/rtsp/udp-control"}
                      {for "/" *}}
                     {{map "http://www.iana.org/rtsp/gzipped-messages"}
                      {for "/" *}}
         Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE

   Note that these are fictional features (though we may want to make
   them real one day).

   Example 2 (using RFC2069-style authentication only as an example):

     C->S: RTSP/1.0 401 1 Unauthorized
           Authorization: Digest username="Mufasa",
               realm="testrealm@host.com",
               nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
               uri="/dir/index.html",
               response="e966c932a9242554e42c8ee200cec7f6",
               opaque="5ccc069c403ebaf9f0171e9517f40e41"

9.2

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. Alternatively,

   Example:

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

  S->C: RTSP/1.0 200 312 OK
        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 server may "push" a new 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

ANNOUNCE

     The ANNOUNCE method serves two purposes:

   When sent from client to server, ANNOUNCE posts the client, for example, if description of a new stream has become available.
   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: DESCRIBE ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0 312
           Accept: application/sdp, application/rtsl, application/mheg

     S->C: RTSP/1.0 200 312 OK
        Date: 23 Jan 1997 15:35:06 GMT
        Content-Type: application/sdp
        Content-Length: 376 332
        v=0
        o=mhandley 2890844526 2890842807 2890845468 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

   or 31

  S->C: RTSP/1.0 200 312 OK
           Date: 23 Jan 1997 15:35:06 GMT
           Content-Type: application/rtsl
           Content-Length: 2782

           <2782 octets of data containing stream description>

   Server to client example:

     S->C: DESCRIBE /twister RTSP/1.0 902
           Session: 1234
           Content-Type: application/rtsl

           new RTSL presentation description

9.3

SETUP

     The SETUP request for a URI specifies the transport mechanism to be
   used for the streamed media. A client can issue a SETUP request for a
   stream that is already playing to change transport parameters. parameters, which a
   server MAY allow(If it does not allow it, it must respond with error
   ``45x 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.

        This

     Segregating content desciption into a DESCRIBE message and
     transport information in SETUP avoids having firewall to parse
     numerous different presentation description formats, formats for information
     which is
        irrelevant.

   If the optional  Require header is present, option tags within the
   header indicate features needed by the requestor that are not
   required at the version level of the protocol. The  Transport-Require
   header is used to indicate proxy-sensitive features that MUST be
   stripped by the proxy to the server if not supported. Furthermore,
   any Transport-Require header features that are not supported by the
   proxy MUST be negatively acknowledged by the proxy irrelevant to the client if
   not supported.

        HS: In my opinion, the Require header should be replaced by
        PEP since PEP is standards-track, has more functionality
        and somebody already did the work. transport.

   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 foo/bar/baz.rm rtsp://example.com/foo/bar/baz.rm RTSP/1.0 302
        Transport: rtp/udp;port=458 RTP/AVP;port=4588

  S->C: RTSP/1.0 200 302 OK
        Date: 23 Jan 1997 15:35:06 GMT
        Transport: cush/udp;port=458

9.4 RTP/AVP;port=4588

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 commands in
   the example below arrive, the server will play first second 10 through
   15 and then, immediately following, seconds 20 to 25 and finally
   seconds 30 through the end.

  C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 835
        Range: npt=10-15

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

  C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 837
        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.

   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 synchronisation
   of streams obtained from different sources.

   For a on-demand stream, the server replies back 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/1.0 833
        Range: smpte=0:10:20-;time=19970123T153600Z

  S->C: RTSP/1.0 200 833 OK
        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/1.0 835
        Range: clock=19961108T142300Z-19961108T143520Z

  S->C: RTSP/1.0 200 833 OK
        Date: 23 Jan 1997 15:35:06 GMT

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

9.5 and smpte formats.

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.

   The PAUSE request may contain a Range header specifying when the
   stream or presentation is to be halted. The header must contain
   exactly one value rather than a time range. The normal play time for
   the stream is set to that value. The pause request becomes effective
   the first time the server is encountering the time point specified. If
   this header is missing, stream delivery is interrupted immediately on
   receipt of the message.

   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, it stops immediately. If the pause request is for
   NPT 16, it stops after completing the first 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 playing 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.

   Example:

  C->S: PAUSE /fizzle/foo rtsp://example.com/fizzle/foo RTSP/1.0 834
        Session: 1234

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

9.6

TEARDOWN

     Stop the stream delivery for the given URI, freeing the resources
   associated with it. If the URI is the root node 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 /fizzle/foo rtsp://example.com/fizzle/foo RTSP/1.0 892
        Session: 1234

  S->C: RTSP/1.0 200 892 OK

9.7

GET_PARAMETER

     The requests retrieves the value of a parameter of a presentation or
   stream specified in the URI. Multiple parameters can be requested in
   the message body using the content type text/rtsp-parameters text/rtsp-parameters. Note
   that parameters include server and client statistics. IANA registers
   parameter names for statistics and other purposes. GET_PARAMETER with
   no entity body may be used to test client or server liveness
   ("ping").
   (``ping'').

   Example:

  S->C: GET_PARAMETER /fizzle/foo rtsp://example.com/fizzle/foo RTSP/1.0 431
        Content-Type: text/rtsp-parameters
        Session: 1234
        Content-Length: 15

        packets_received
        jitter

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

        packets_received: 10
        jitter: 0.3838

9.8

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. 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.

   A SET_PARAMETER request without parameters can be used as a way to
   detect client or server liveness.

   Example:

  C->S: SET_PARAMETER /fizzle/foo rtsp://example.com/fizzle/foo RTSP/1.0 421
        Content-type: text/rtsp-parameters

           fooparam: foostuff

        barparam: barstuff

  S->C: RTSP/1.0 450 421 Invalid Parameter
        Content-Length: 6

        barparam

9.9

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 a  DESCRIBE requests for that URL. It
   may contain the parameter Range, which indicates when the redirection
   takes effect.

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

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

9.10

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  Conference header is
   mandatory.

   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,
   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.

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

  C->S: RECORD /meeting/audio.en rtsp://example.com/meeting/audio.en RTSP/1.0 954
        Session: 1234
        Conference: 128.16.64.19/32492374

9.11

10.12 Embedded (Interleaved) Binary Data

   Binary packets

     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 data are packets is encapsulated by an ASCII dollar
   sign (24 decimal), followed by a one-byte session channel identifier, followed
   by the length of the encapsulated binary data as a binary, two-byte
   integer in network byte order. The binary data follows
   immediately afterwards, without a CRLF.

10 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 12.35.

  C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 2
        Transport: RTP/AVP/TCP;channel=0
  S->C: RTSP/1.0 200 2 OK
        Date: 05 Jun 1997 18:57:18 GMT
        Transport: RTP/AVP/TCP;channel=0
        Session: 12345

  C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0 3
        Session: 12345

  S->C: RTSP/1.0 200 3 OK
        Session: 12345
        Date: 05 Jun 1997 18:59:15 GMT

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

11 Status Code Definitions

     Where applicable, HTTP status [H10] codes are re-used. 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 request.

10.1

11.1 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.

10.2

11.2 Client Error 4xx

10.2.1

  11.2.1 405 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.

  11.2.2 451 Parameter Not Understood

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

10.2.2

  11.2.3 452 Conference Not Found

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

10.2.3

  11.2.4 453 Not Enough Bandwidth

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

10.2.4

  11.2.5 45x Session Not Found

   The RTSP session identifier is invalid or has timed out.

10.2.5

  11.2.6 45x Method Not Valid in This State

   The client or server cannot process this request in its current state.

10.2.6

  11.2.7 45x 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.

10.2.7

  11.2.8 45x Invalid Range

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

10.2.8

  11.2.9 45x Parameter Is Read-Only

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

11

  11.2.10 45x 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.

  11.2.11 45x 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.

12 Header Field Definitions

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

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

   If the field content does not apply to the particular resource, the
   server MUST return status 45x (Header Field Not Valid for Resource).

11.1 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] particular resource, the
   server MUST return status 45x (Header Field Not Valid for syntax.

   Example of use:

     Accept: application/rtsl, application/sdp;level=2 Resource).

    Header              type   support   methods
     _________________________________________________________________
    Accept              R      opt.      entity
    Accept-Encoding     R      opt.      entity
    Accept-Language     R      opt.      all
    Authorization       R      opt.      all
    Bandwidth           R      opt.      SETUP      all
    Blocksize           R      opt.      all but OPTIONS, TEARDOWN
    Cache-Control         Rr       g      opt.      SETUP
    Conference          R      opt.      SETUP
    Connection            Rr          g      req.      all
    Content-Encoding      R    e      req.      SET_PARAMETER
    Content-Encoding      r    e      req.      DESCRIBE      DESCRIBE, ANNOUNCE
    Content-Language    e      req.      DESCRIBE, ANNOUNCE
    Content-Length        R      e      req.      SET_PARAMETER      SET_PARAMETER, ANNOUNCE
    Content-Length        r      e      req.      entity
    Content-Type          R        e      req.      SET_PARAMETER      SET_PARAMETER, ANNOUNCE
    Content-Type        r      req.      entity
    Date                  Rr                g      opt.      all
    Expires               r             e      opt.      DESCRIBE      DESCRIBE, ANNOUNCE
    From                R      opt.      all
    If-Modified-Since   R      opt.      DESCRIBE, SETUP
    Last-Modified         r       e      opt.      entity
    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
    Scale               Rr     opt.      PLAY, RECORD
    Session             Rr     req.      all but SETUP, OPTIONS
    Server              r      opt.      all
    Speed               Rr     opt.      PLAY
    Transport           Rr     req.      SETUP
     Transport-Require     R       xeq.      all
    Transport-Info      r      req.      PLAY
    User-Agent          R      opt.      all
    Via                   Rr                 g      opt.      all
    WWW-Authenticate    r      opt.      all
!
            Table 3: Overview of RTSP header fields

11.2

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]

11.3

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.

11.4

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

11.5

12.5 Authorization

     See [H14.8]

11.6

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.

  Bandwidth  = "Bandwidth" ":" 1*DIGIT

   Example:
  Bandwidth: 4000

11.7

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 overrides override it with
   the media specific size if necessary. The block size is a strictly
   positive decimal number and measured in octets. The server only
   returns an error (416) if the value is syntactically invalid.

11.8

12.8 C-PEP

     This corresponds to the C-PEP: header in the ``Protocol Extension
   Protocol'' defined in RFC XXXX [21]. This field differs from the PEP
   field (Section 12.24) only in that it is hop-by-hop rather than
   end-to-end as PEP is. Servers and proxies MUST parse this field and
   MUST return "420 Bad Extension" when there is a PEP extension of
   strength "must". See RFC XXXX for more details on this.

12.9 C-PEP-Info

     This corresponds to the C-PEP-Info: header in the ``Protocol
   Extension Protocol'' defined in RFC XXXX [21].

12.10 Cache-Control

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

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

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

   [HS: Should there be an exception cacheable, except for DESCRIBE?] 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"
      | "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 cachable 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)
          (non-shared) cache may cache the media stream.

   no-transform:
          An intermediate cache (proxy) may find it useful to convert the
          media type of 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.

   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. (I.e.,
          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.)

11.9

12.11 Conference

     This request header field establishes a logical connection between a
   conference, established using non-RTSP means, 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

11.10 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr

12.12 Connection

     See [H14.10].

11.11

12.13 Content-Encoding

     See [H14.12]

11.12

12.14 Content-Language

     See [H14.13]

12.15 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. It is interpreted according to [H14.14].

11.13

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.

11.14

12.17 Date

     See [H14.19].

11.15

12.18 Expires

     The Expires entity-header field gives the date/time after which the
   media-stream should be considered stale. A stale cache entry may not
   normally be returned by a cache (either a proxy cache or an user agent
   cache) unless it is first validated with the origin server (or with an
   intermediate cache that has a fresh copy of the entity). See section
   13.2 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 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 cachable, unless
   indicated otherwise by a Cache-Control header field (Section 11.8.

11.16 12.10).

12.19 From

     See [H14.22].

12.20 Host

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

12.21 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) (DESCRIBE) or a
   stream will not be setup ( SETUP); (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

11.17 Last-modified

12.22 Last-Modified

     The Last-Modified entity-header field indicates the date and time at
   which the origin server believes the variant was last modified. See
   [H14.29]. If the request URI refers to an aggregate, the field
   indicates the last modification time across all leave nodes of that
   aggregate.

11.18

12.23 Location

     See [H14.30].

11.19 Nack-Transport-Require

   Negative acknowledgement of features not supported by

12.24 PEP

     This corresponds to the server. If PEP: header in the ``Protocol Extension
   Protocol'' defined in RFC XXXX. Servers MUST parse this field and MUST
   return ``420 Bad Extension'' when there is a proxy on the path between the client and PEP extension of strength
   ``must'' (see RFC XXXX).

12.25 PEP-Info

     This corresponds to the server, PEP-Info: header in the
   proxy MUST insert a message reply with an error message 506 (Feature
   not supported).

        HS: Same caveat as for Require applies.

11.20 ``Protocol Extension
   Protocol'' defined in RFC XXXX.

12.26 Proxy-Authenticate

     See [H14.33].

12.27 Public

   See [H14.35].

12.28 Range

     This request header field specifies a range of time. The range can
   be specified in a number of units. This specification defines the
   smpte (see Section 3.4) 3.5) and clock (see Section 3.6) 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 and SMPTE range
   formats.

  Range = "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ]

  ranges-specifier = npt-range | utc-range | smpte-range

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

     The notation is similar to that used for the HTTP/1.1 header. It
     allows to select a clip from the media object, to play from a given
     point to the end and from the current location to a given point.

11.21 Require

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

        HS: Naming start of features -- yet another name space. I believe
        this header field to be redundant. PEP should playback can be used
        instead.

   For example

   C->S:   SETUP /foo/bar/baz.rm RTSP/1.0 302
           Require: funky-feature
           Funky-Parameter: funkystuff

   S->C:   RTSP/1.0 200 506 Option not supported
           Unsupported: funky-feature

   C->S:   SETUP /foo/bar/baz.rm RTSP/1.0 303

   S->C:   RTSP/1.0 200 303 OK
   This is to make sure that the client-server interaction will proceed
   optimally when all options are understood by both sides, and only
   slow down if options aren't understood (as scheduled for at any time in the case above). For a
   well-matched client-server pair, the interaction proceeds quickly,
   saving
     future, although a round-trip often required by negotiation mechanisms. In
   addition, it also removes state ambiguity when the client requires
   features server may refuse to keep server resources for
     extended idle periods.

12.29 Referer

     See [H14.37]. The URL refers to that of the server doesn't understand.

11.22 presentation
   description, typically retrieved via HTTP.

12.30 Retry-After

     See [H14.38].

11.23

12.31 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") (``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

11.24

12.32 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. A negative value indicates that the stream
   is to be played back in reverse direction.

        HS: With 'Scale', the negative value is redundant and
        should probably be removed since it only leads to possible
        conflicts when Scale is positive and Speed negative. 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

11.25

   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.

12.33 Server

     See [H14.39]

11.26

12.34 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 and has the same syntax as a conference identifier. (see Section 3.4). Once a client receives a Session
   identifier, it MUST return it for any request related to that session.

        HS: This

  Session  = "Session" ":" session-id

   Note that a session identifier identifies a RTSP session across
   transport sessions or connections. Control messages for more than one
   RTSP URL may be redundant with the standards-track HTTP
        state maintenance mechanism [2]. The equivalent way of
        doing this would be for sent within a single RTSP session. Hence, it is
   possible that clients use the server to send Set-Cookie:
        Session="123"; Version=1; Path = "/twister" and same session for controlling many
   streams comprising a presentation, as long as all the
        client to return later Cookie: Session = "123"; $Version=1;
        $Path = "/twister" response to the TEARDOWN message, streams come
   from the
        server would simply send Set-Cookie: Session="123";
        Version=1; Max-Age=0 to get rid of same server. (See example in Section 14). However, multiple
   ``user'' sessions for the cookie on same URL from the same client
        side. Cookies also have a time-out, so that a server may
        limit the lifetime of a MUST use
   different session at will. Unlike a web
        browser, a client would not store these states on disk. To
        avoid privacy issues, we should prohibit identifiers.

     The session identifier is needed to distinguish several delivery
     requests for the Host
        parameter.

11.27 same URL coming from the same client.

12.35 Transport

     This request header indicates which transport protocol is to be used
   and configures its parameters such as multicast, 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. In some cases, the presentation description contains all
   necessary information.  In those cases,

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

   The Transport header field
   (and the  SETUP request containing it) are not needed.

   in whatever protocol is being MAY also be used by the control to change certain transport
   parameters. A server MAY refuse to change parameters of an existing
   stream. Currently,

   The server MAY return a Transport response header in the response to
   indicate the next-layer protocols RTP is defined. Parameters values actually chosen.

   A Transport request header field may be added contain a list of transport
   options acceptable to
   each protocol, separated by the client. In that case, the server MUST return
   a semicolon. For RTP, single option which was actually chosen.

   The syntax for the boolean
   parameter compressed transport specifier is defined, indicating compressed RTP according
   transport/profile/lower-transport. Defaults for "lower-transport" are
   specific to RFC XXXX. the profile. For multicast UDP, RTP/AVP, the integer parameter  ttl defines default is UDP.

   Below are the time-to-live value configuration parameters associated with transport:

   General parameters:

   destination:
          The address to which a stream will be used. sent. The client may
          specify the multicast address with the  multicast destination parameter. A
          server SHOULD authenticate the client and SHOULD log such
          attempts before allowing the client to direct a media stream to a multicast
          an address not chosen by the server server to avoid becoming the
          unwitting perpetrator of a remote-controlled denial-of-service
          attack. This is particularly important if RTSP commands are
          issued via UDP, but TCP cannot be relied upon 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.

   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. For RECORD, the append flag
          indicates that the media data should be appended to the
          existing resource rather than overwriting it. If appending is
          requested and the server does not support this, it MUST refuse
          the request rather than overwrite the resouce 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. Currently, the next-layer protocols RTP is
          defined. The `channel' parameter defines the channel number to
          be used in the $ statement (see section 10.12).

   Multicast specific:

   ttl:
          multicast time-to-live

   RTP Specific:

   compressed:
          Boolean parameter indicating compressed RTP according to avoid
   becoming RFC
          XXXX.

   port:
          RTP/RTCP destination ports on client. The client receives RTCP
          reports on the unwitting perpetrator value of a denial-of-service attack. For
   UDP and TCP, the parameter port defines plus one, as is standard RTP
          convention.

   cport:
          the control port that the data is server wishes the client to be sent send
          its RTCP reports to.

   The  SSRC parameter indicates

   ssrc:
          Indicates the RTP SSRC [19, 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.

   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.

   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  Transport
   header field makes sense only for an individual media stream, not a
   presentation.

  Transport = "Transport" ":"
                 1#transport-protocol/upper-layer
              1#transport-protocol/profile[/lower-transport] *parameter
  transport-protocol  = "UDP" | "TCP"
     upper-layer  = "RTP"
     parameters
  profile     = "AVP"
  lower-transport = "TCP" | "UDP"
  parameter   = ";" "multicast" "destination" [ "=" mca address ]
              | ";" "compressed"
              | ";" "interleaved" "channel" "=" channel
              | ";" "append"
              | ";" "ttl" "=" ttl
              | ";" "port" "=" port
              | ";" "cport" "=" port
              | ";" "ssrc" "=" ssrc
              | ";" "mode" = <"> 1#mode <">
  ttl         = 1*3(DIGIT)
  port        = 1*5(DIGIT)
  ssrc        = 8*8(HEX)
     mca
  channel     = 1*3(DIGIT)
  address     = host
  mode        = "PLAY" | "RECORD" *parameter

   Example:
  Transport: udp/rtp;compressed;ttl=127;port=3456

11.28 Transport-Require RTP/AVP;compressed;ttl=127;port=3456;
    mode="PLAY,RECORD;append"

     The Transport-Require Transport header is used restricted to indicate proxy-sensitive
   features that MUST be stripped by the proxy 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.

12.36 Transport-Info

     This field is used to set Transport specific parameters in the server if not
   supported.  Furthermore, any Transport-Require header features that
   are not supported by the proxy MUST be negatively acknowledged by the
   proxy to PLAY
   response.

   seq:
          Indicates the client if not supported.

   See Section 11.21 for more details on sequence number of the mechanics first packet of the
          stream. This allows clients to gracefully deal with packets
          when seeking. The client uses this message
   and a usage example.

        HS: Same caveat as for Require applies.

11.29 Unsupported

   See Section 11.21 for a usage example.

        HS: same caveat as for Require applies.

11.30 value to differentiate
          packets that originated before the seek from packets that
          originated after the seek.

  Transport-Info = "Transport-Info" ":"
              1#transport-protocol/profile[/lower-transport] ";"
              streamid
              *parameter
  transport-protocol  = "RTP"
  profile     = "AVP"
  lower-transport = "TCP" | "UDP"
  stream-id = "streamid" "=" streamid
  parameter   = ";" "seq" "=" sequence number
  sequence-number = 1*16(DIGIT)

   Example:
Transport-Info: RTP/AVP;streamid=0;seq=43754027,
                RTP/AVP;streamid=1;seq=34834738

12.37 User-Agent

     See [H14.42]

11.31

12.38 Vary

     See [H14.43]

12.39 Via

     See [H14.44].

11.32

12.40 WWW-Authenticate

     See [H14.46].

12

13 Caching

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

   On receiving a SETUP or PLAY request, the proxy would ascertain as to
   whether it has an up-to-date copy of the continuous media content. If
   not, it would modify the SETUP transport parameters as appropriate and
   forward the request to the origin server. Subsequent control commands
   such as PLAY or PAUSE would pass the proxy unmodified. The proxy would
   then pass the continuous media data to the client, while possibly
   making a local copy for later re-use. The exact behavior allowed to
   the cache is given by the cache-response directives described in
   Section 11.8. 12.10. A cache MUST answer any DESCRIBE requests if it is
   currently serving the stream to the requestor, as it is possible that
   low-level details of the stream description may have changed on the
   origin-server.

   Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
   through"
   ``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 would appear like a regular media
   server, to the media origin server like a client. Just like an HTTP
   cache has to store HTTP
   cache has to store the content type, content language, etc. for the
   objects it caches, a media cache has to store the presentation
   description. Typically, a cache would eliminate 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 reference stream description formats that are
   not finalized, such as RTSL and SDP. Please do not use these examples
   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 content type, content language, etc. for time line of the
   objects it caches, a media cache has to store movie.

   In our example, the presentation
   description. Typically, a cache would eliminate all transport-
   references (that is, multicast information) from client is only interested in the presentation
   description, since these are independent last part of the data delivery from
   movie. The server requires authentication for this 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

      v=0
      o=- 2890844526 2890842807 IN IP4 192.16.24.202
      s=RTSP Session
      m=audio 0 RTP/AVP 0
      a=murl:rtsp://audio.example.com/twister/audio.en
      m=video 0 RTP/AVP 31
      a=murl:rtsp://audio.example.com/twister/video

C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 1
      Transport: rtp/udp;port=3056

A->C: RTSP/1.0 200 1 OK
      Session: 1234

C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 1
      Transport: rtp/udp;port=3058

V->C: RTSP/1.0 200 1 OK
      Session: 1235

C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 2
      Session: 1235
      Range: smpte=0:10:00-

V->C: RTSP/1.0 200 2 OK

C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 2
      Session: 1234
      Range: smpte=0:10:00-

A->C: RTSP/1.0 200 2 OK

C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 3
      Session: 1234

A->C: RTSP/1.0 200 3 OK

C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 3
      Session: 1235

V->C: RTSP/1.0 200 3 OK

   Even though the cache 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. Information on client can synchronize the encodings remains two using standard RTP
   methods, in particular the
   same. If time scale contained in the cache RTCP sender
   reports.

14.2 Streaming of a Container file

   For purposes of this example, a container file is able a storage entity in
   which multiple continuous media types pertaining to translate the cached media data, it
   would create a new same end-user
   presentation description with all are present. In effect, the encoding
   possibilities it can offer.

13 Examples

   The following examples reference stream description formats that container file represents a
   RTSP presentation, with each of its components being RTSP streams.
   Container files are
   not finalized, a widely used means to store such presentations.
   While the components are essentially transported as RTSL and SDP. Please do not use these examples
   as independant
   streams, it is desirable to maintain a reference common context for those formats.

13.1 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
   streams at the server W. The media description contains descriptions of end.

     This enables the
   presentation and server to keep a single storage handle open
     easily. It also allows treating all its streams, including the codecs that are
   available, dynamic RTP payload types, streams equally in case of
     any prioritization of streams by the protocol stack and content
   information such as language or copyright restrictions. server.

   It may is also
   give an indication about possible that the time line presentation author may wish to prevent
   selective retreival of the movie.

   In our example, the streams by client is only interested in order to preserve the last part
   artistic effect of the
   movie. The server requires authentication for this movie. The audio
   track can combined media presentation. Similarly, in such
   a tightly bound presentation, it is desirable to be dynamically switched between between two sets of
   encodings.  The URL with scheme rtpsu indicates able to control
   all the media servers
   want 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 UDP for exchanging of aggregate URLs.

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

   C->W: URL to the
   container file.

C->M: DESCRIBE /twister HTTP/1.1
         Host: www.example.com
         Accept: application/rtsl; application/sdp

   W->C: rtsp://foo/twister  RTSP/1.0 1

M->C: RTSP/1.0 200 1 OK
      Content-Type: application/rtsl

         <session>
           <group language=en lipsync>
             <switch>
               <track type=audio
                 e="PCMU/8000/1"
                 src="rtsp://audio.example.com/twister/audio.en/lofi">
               <track type=audio
                 e="DVI4/16000/2" pt="90 DVI4/8000/1"
                 src="rtsp://audio.example.com/twister/audio.en/hifi">
             </switch>
             <track type="video/jpeg"
               src="rtspu://video.example.com/twister/video">
           </group>
         </session>

   C->A: application/sdp
      Content-Length: 64
      s= sample rtsp presentation
      r = rtsp://foo/twister   /* aggregate URL*/
      m= audio 0 RTP/AVP 0
      r = rtsp://foo/twister/audio
      m=video 0 RTP/AVP 26
      r = rtsp://foo/twister/video

C->M: SETUP rtsp://audio.example.com/twister/audio.en/lofi rtsp://foo/twister/audio RTSP/1.0 1 2
      Transport: rtp/udp;compression;port=3056

   A->C: RTP/AVP;port=8000

M->C: RTSP/1.0 200 1 2 OK
      Session: 1234

   C->V:

C->M: SETUP rtsp://video.example.com/twister/video rtsp://foo/twister/video RTSP/1.0 1 3
      Transport: rtp/udp;compression;port=3058

   V->C: RTP/AVP;port=8002
      Session: 1234

M->C: RTSP/1.0 200 1 3 OK
      Session: 1235

   C->V: 1234

C->M: PLAY rtsp://video.example.com/twister/video  rtsp://foo/twister  RTSP/1.0 2
         Session: 1235 4
      Range: smpte=0:10:00-

   V->C: npt=0-
      Session: 1234

M->C: RTSP/1.0 200 2 4 OK

   C->A: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 2
      Session: 1234
         Range: smpte=0:10:00-

   A->C: 200 2 OK

   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi

C->M: PAUSE rtsp://foo/twister/video RTSP/1.0 3 5
      Session: 1234

   A->C: 200 3 OK

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

M->C: RTSP/1.0 3 4xx 5 Only aggregate operation allowed

C->M: PAUSE rtsp://foo/twister RTSP/1.0 6
      Session: 1235

   V->C: 1234

M->C: RTSP/1.0 200 3 6 OK

   Even though
      Session: 1234

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

M->C: RTSP/1.0 4xx 7 Aggregate operation not allowed

   In the audio and video track are on two different servers,
   may start at slightly different times and may drift with respect to
   each other, first instance of failure, the client can synchronize tries to pause one
   stream(in this case video) of the two using standard RTP
   methods, in particular presentation which is disallowed for
   that presentation by the time scale contained in server. In the RTCP sender
   reports.

13.2 second instance, the aggregate
   URL may not be used for SETUP and one control message is required per
   stream to setup transport parameters.

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

14.3 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.
   During the RTSP session, a new subtitling stream is added.

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

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

      <session>
        <track id=17 src="rtsp://live.example.com/concert/audio">
      </session>

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

M->C: RTSP/1.0 200 1 OK
      Content-Type: application/rtsl

         <track id=17 type=audio address=224.2.0.1 port=3456 ttl=16> application/sdp

      v=0
      o=- 2890844526 2890842807 IN IP4 192.16.24.202
      s=RTSP Session
      m=audio 3456 RTP/AVP 0
      c=IN IP4 224.2.0.1/16

C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 2
      Transport: multicast=224.2.0.1; port=3456; ttl=16

C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 3
         Range: smpte 1:12:0

M->C: RTSP/1.0 405 200 3 No positioning possible

   M->C: DESCRIBE concert RTSP/1.0
         Content-Type: application/rtsl

         <session>
           <track id=17
             media=audio/g.728 src="rtsp://live.example.com/concert/audio">
           <track id=18
             media=text/html src="rtsp://live.example.com/concert/lyrics">
         </session>

   C->M: PLAY rtsp://live.example.com/concert/lyrics RTSP/1.0 OK

   The attempt to position the stream fails since this is a live
   presentation.

13.3

14.4 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. When retrieving the
   presentation description, 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: GET /demo HTTP/1.1
         Host: www.example.com DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0 1
      Accept: application/rtsl, application/sdp

M->C: HTTP/1.1 RTSP/1.0 200 1 OK
      Content-type: application/rtsl

         <session>
           <track type=audio/g.723.1
             src="rtsp://server.example.com/demo/548/sound">
         </session>

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

C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0 2
      Conference: 218kadjk

13.4

14.5 Recording

   The conference participant C asks the media server M to record a
   meeting. If the presentation description contains any alternatives,
   the server records them all.

C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 89 90
      Content-Type: application/sdp

      v=0
      s=Mbone Audio
      i=Discussion of Mbone Engineering Issues

M->C: 415 89 Unsupported Media Type
         Accept: application/rtsl

   C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 90
         Content-Type: application/rtsl

   M->C: 200 90 OK

C->S: SETUP rtsp://server.example.com/meeting RTSP/1.0 91
      Transport: RTP/AVP;mode=record

S->C: RTSP/1.0 200 91 OK
      Transport: RTP/AVP;port=3244;mode=record
      Session: 508876

C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 91 92
      Session: 508876
      Range: clock 19961110T1925-19961110T2015

14

15 Syntax

     The RTSP syntax is described in an augmented Backus-Naur form (BNF)
   as used in RFC 2068 (HTTP/1.1).

14.1

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
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>

15

16 Security Considerations

   The protocol offers the opportunity for a remote-control denial-of-
   service remote-controlled
   denial-of-service attack.

   The attacker, using a forged source IP address, can ask for a stream
   to be played back to that forged IP address. Thus, an RTSP server
   SHOULD only allow client-specified destinations for RTSP-initiated
   traffic flows if the server has verified the client's identity, e.g.,
   using the RTSP authentication mechanisms.

   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. This does not require spoofing of network packet addresses.
   The server SHOULD use a large random session identifier to make this
   attack more difficult.

   Both problems can be be prevented by appropriate authentication.

   Servers SHOULD implement both basic and digest [8] authentication.

   In addition, the security considerations outlined in [H15] apply.

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.

   [TBD: should we allow for the trivial case of a server that only
   implements the  PLAY message, with no control.]

   State is defined on a per object basis. An object is uniquely
   identified by the stream URL and the RTSP session identifier. (A
   server may choose to generate dynamic presentation descriptions where
   the URL is unique for a particular RTSP session and thus may not need
   an explicit RTSP session identifier in the request header.) Any
   request/reply using aggregate URLs denoting an RTSP session presentations
   comprised of multiple streams will have an effect on the individual
   states of all the substreams. streams. For example, if the stream presentation /movie
   contains two
   substreams streams /movie/audio and /movie/video, then the following
   command:

  PLAY /movie rtsp://foo.com/movie RTSP/1.0 559
  Session: 12345

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

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

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

   Client and servers MUST disregard messages with a sequence number
   less than the last one. If no message has been received, the first
   received message's sequence number will be the starting point.

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 after playing, PAUSE reply received.

   Playing:
          PLAY reply received

   Recording:
          RECORD reply received

   The

   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
   available via a multicast group), state begins at READY. In this case,
   there are only two states, READY and PLAYING.

   The "next state" 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 greater or
   equal to 300, the client state becomes Init, with the exception of a status codes 401 (Unauthorized) and 402 (Payment Required), where code of 3xx,
   the state remains unchanged becomes Init, and the request should be re-issued with the
   appropriate authentication or payment information. 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.

        HS: Depends on allowing PLAY without SETUP. After 4xx or
        5xx error, do we go back to Init?

     state        message           next state
   _______________________________________________________
     Init         SETUP       Ready
                  TEARDOWN    Init
     Ready        PLAY        Playing
                  RECORD      Recording
                  TEARDOWN    Init
     Playing      PAUSE       Ready
                  TEARDOWN    Init
                  PLAY        Playing
                 RECORD      Recording
                  SETUP       Playing (changed transport)
     Recording    PAUSE       Init       Ready
                  TEARDOWN    Init
                 PLAY        Playing
                  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 received.

   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.

   The

   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"
   ``wellness'' information, such as RTCP reports, from the client for a
   defined interval, with a default of one minute. 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 transitions 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. If a
   similar change of location occurs at immediately unless it
   has a certain time in Range: header specifying when the future,
   this redirect is assumed to be indicated by effective. In such
   a case, server state will also change at the presentation description.

   SETUP is valid in states Init and Ready only. An error message should
   be returned in other cases. appropriate time.

   If no explicit SETUP is required for the object, state starts at
   READY, 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             next state
   ___________________________________
   Init           SETUP               Ready
                  TEARDOWN            Init
   Ready          PLAY                Playing
                  SETUP               Ready
                  TEARDOWN    Ready            Init
                  RECORD              Recording
   Playing        PLAY                Playing
                  PAUSE               Ready
                  TEARDOWN    Ready
                RECORD      Recording            Init
                  SETUP               Playing
   Recording      RECORD              Recording
                  PAUSE               Ready
                  TEARDOWN    Ready
                PLAY        Playing            Init
                  SETUP               Recording

B Open Issues

        o

   1.
          Define text/rtsp-parameter MIME type.

        o
   2.
          Reverse: Scale: -1, with reversed start times, or both?
   3.
          HS believes that RTSP should only control individual media
          objects rather than aggregates. This avoids disconnects between
          presentation descriptions and streams and avoids having to deal
          separately with single-host and multi-host case. Cost: several
          PLAY/PAUSE/RECORD in one packet, one for each stream.

        o
   4.
          Allow changing of transport for a stream that's playing? May
          not be a great idea since the same can be accomplished by tear
          down and re-setup.

        o Allow fragment (#) identifiers for controlling substreams Exception: near-video-on-demand, where the
          server changes the address in
         Quicktime, AVI a PLAY response. Servers may not
          be able to reliably send TEARDOWN to clients and ASF files?

        o the client
          wouldn't know why this happened in any event.
   5.
          How does the server get back to the client unless a persistent
          connection is used? Probably cannot, in general.

        o Cache and proxy behavior?

        o Session: or Set-Cookie: ?

        o When do relative RTSP URLs make sense?

        o Nack-require, etc. are dubious. This is getting awfully close
   6.
          Server issues TEARDOWN and other 'event' notifications to
          client? This raises the HTTP extension mechanisms [19] problem discussed in complexity, the previous open
          issue, but is
         different.

        o Use HTTP absolute path + Host field or do useful for the right thing and
         carry full URL, including host in request? client if the data stream contains
          no end indication.

C Changes

   Since the February March 1997 version, the following changes were made:

        o Various editorial changes and clarifications.

        o Removed references

     * Allowing the Transport header to direct media streams to SDF unicast
       and replaced by RTSL.

        o Added  Scale general header.

        o Clarify behavior of multicast addresses, with an appropriate warning about
       denial-of-service attacks.
     * Add mode parameter to Transport header to allow RECORD or PLAY.

        o Rename GET
     * The Embedded binary data section was modified to DESCRIBE.

        o Removed SESSION since it is just DESCRIBE in clearly indicate
       the other
         direction.

        o Rename CLOSE stream the data corresponds to, and a reference to TEARDOWN, in symmetry with SETUP.

        o Terminology adjusted the
       Transport header was added.
     * The Transport header format has been changed to use a more general
       means to "presentation" specify data channel and "stream".

        o Redundant syntax BNF application level protocol. It
       also conveys the port to be used at the server for RTCP messages,
       and the start sequence number that will be used in appendix removed since it just
         duplicates HTTP spec.

        o Beginnings the RTP
       packets.
     * The use of cache control.

   Changes are marked by changebars the Session: header has been enhanced. Requests for
       multiple URLs may be sent in a single session.
     * There is a distinction between aggregate(presentation) URLs and
       stream URLs. Error codes have been added to reflect the fact that
       some methods may be allowed only on a particular type of URL.
     * Example showing the use of aggregate/presentation control using a
       single RTSP session has been added.
     * Support for the PEP(Protocol Extension Protocol) headers has been
       added.
     * Server-Client DESCRIBE messages have been renamed to ANNOUNCE for
       better clarity and differentiation.

   Note that this list does not reflect minor changes in the margins wording or
   correction of the PostScript
   version. typographical errors.

D Author Addresses

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   electronic mail: schulzrinne@cs.columbia.edu
   Anup Rao
   Netscape Communications Corp.
   501 E. Middlefield Road
   Mountain View, CA 94043
   USA
   electronic mail: anup@netscape.com

   Robert Lanphier
   Progressive Networks
   1111 Third Avenue Suite 2900
   Seattle, WA 98101
   USA
   electronic mail: robla@prognet.com

E Acknowledgements

   This draft is based on the functionality of the original RTSP draft. draft
   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             Eduardo F. Llach
            Bruce Butterfield         Rob McCool
             Steve Casner             David Oran
            Martin Dunsmuir           Sujal Patel
            Eric Fleischman
            Mark Handley              Igor Plotnikov
            Peter Haight              Pinaki Shah
            Brad Hefta-Gaub           Jeff Smith
            John K. Ho           Alexander Sokolsky
   Ruth Lang            Dale Stammen
   Stephanie Leif       John Francis Stracke

F Bibliography

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

   [2] D. Kristol and L. Montulli, "HTTP state management mechanism,"
   RFC 2109, Internet Engineering Task Force, Feb. 1997.

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

   [4] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," Internet Draft, Internet Engineering Task Force, Jan. 1997.
   Work in progress.

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

   [6] M. Handley, "SDP: Session description protocol," Internet Draft,
   Internet Engineering Task Force, Nov. 1996.  Work in progress.

   [7] A. Freier, P. Karlton, and P. Kocher, "The TLS protocol,"
   Internet Draft, Internet Engineering Task Force, Dec. 1996.  Work in
   progress.

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

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

   [10] R. Hinden and C. Partridge, "Version 2 of the reliable data
   protocol (RDP),"  RFC 1151, Internet Engineering Task Force, Apr.
   1990.

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

   [12] M. Handley, H. Schulzrinne, and E. Schooler, "SIP: Session
   initiation protocol," Internet Draft, Internet Engineering Task
   Force, Dec. 1996.  Work in progress.

   [13] P. McMahon, "GSS-API authentication method                Alexander Sokolsky
            Ruth Lang                 Dale Stammen
            Stephanie Leif            John Francis Stracke

References

   1
          H. Schulzrinne, ``RTP profile for SOCKS version 5," audio and video conferences
          with minimal control,'' RFC 1961, 1890, Internet Engineering Task
          Force, June Jan. 1996.

   [14]
   2
          D. Crocker, "Augmented BNF for syntax specifications: ABNF,"
   Internet Draft, Internet Engineering Task Force, Oct. 1996.  Work in
   progress.

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

   [16] T. Berners-Lee, L. Masinter, Kristol and M. McCahill, "Uniform resource
   locators (URL)," L. Montulli, ``HTTP state management
          mechanism,'' RFC 1738, 2109, Internet Engineering Task Force, Dec.
   1994.

   [17] International Telecommunication Union, "Visual telephone systems Feb.
          1997.
   3
          F. Yergeau, G. Nicol, G. Adams, 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.

   [18] ISO/IEC, "Information technology -- generic coding M. Duerst,
          ``Internationalization of moving
   pictures and associated audio informaiton -- part 6: extension for
   digital storage media and control," Draft International Standard ISO
   13818-6, International Organization for Standardization ISO/IEC
   JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.

   [19] D. Connolly, "PEP: an extension mechanism for http," Internet
   Draft, the hypertext markup language,'' RFC
          2070, Internet Engineering Task Force, Jan. 1997.  Work in progress.

                           Table of Contents

   1          Introduction ........................................    1
   1.1        Purpose .............................................    1
   1.2        Requirements ........................................    3
   1.3        Terminology .........................................    3
   1.4        Protocol Properties .................................    5
   1.5        Extending RTSP ......................................    6
   1.6        Overall Operation ...................................    7
   1.7        RTSP States .........................................    8
   1.8        Relationship with Other Protocols ...................    9
   2          Notational Conventions ..............................   10
   3          Protocol Parameters .................................   10
   3.1        H3.1 ................................................   10
   3.2        RTSP URL ............................................   10
   3.3        Conference Identifiers ..............................   11
   3.4        SMPTE Relative Timestamps ...........................   12
   3.5        Normal Play Time ....................................   13
   3.6        Absolute Time .......................................   13
   4          RTSP Message ........................................   13
   4.1        Message Types .......................................   14
   4.2        Message Headers .....................................   14
   4.3        Message Body ........................................   14
   4.4        Message Length ......................................   14
          S. Bradner, ``Key words for use in RFCs to indicate requirement
          levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997.
   5          Request .............................................   15
   6          Response ............................................   16
   6.1        Status-Line .........................................   17
   6.1.1      Status Code
          R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and Reason Phrase .......................   17
   6.1.2      Response Header Fields ..............................   19 T.
          Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC
          2068, Internet Engineering Task Force, Jan. 1997.
   6
          M. Handley, ``SDP: Session description protocol,'' Internet
          Draft, Internet Engineering Task Force, Nov. 1996.
          Work in progress.
   7          Entity ..............................................   19
   7.1        Entity Header Fields ................................   21
   7.2        Entity Body .........................................   21
          A. Freier, P. Karlton, and P. Kocher, ``The TLS protocol,''
          Internet Draft, Internet Engineering Task Force, Dec. 1996.
          Work in progress.
   8          Connections .........................................   21
   8.1        Pipelining ..........................................   22
   8.2        Reliability
          J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and Acknowledgements ....................   22
          E. L. Stewart, ``An extension to HTTP: digest access
          authentication,'' RFC 2069, Internet Engineering Task Force,
          Jan. 1997.
   9          Method Definitions ..................................   23
   9.1        OPTIONS .............................................   24
   9.2         DESCRIBE ...........................................   25
   9.3         SETUP ..............................................   26
   9.4         PLAY ...............................................   27
   9.5         PAUSE ..............................................   28
   9.6         TEARDOWN ...........................................   30
   9.7         GET_PARAMETER ......................................   30
   9.8         SET_PARAMETER ......................................   31
   9.9         REDIRECT ...........................................   31
   9.10        RECORD .............................................   32
   9.11       Embedded Binary Data ................................   32
          J. Postel, ``User datagram protocol,'' STD 6, RFC 768, Internet
          Engineering Task Force, Aug. 1980.
   10         Status Code Definitions .............................   33
   10.1       Redirection 3xx .....................................   33
   10.2       Client Error 4xx ....................................   33
   10.2.1     451 Parameter Not Understood ........................   33
   10.2.2     452 Conference Not Found ............................   33
   10.2.3     453 Not Enough Bandwidth ............................   33
   10.2.4     45x Session Not Found ...............................   33
   10.2.5     45x Method Not Valid in This State ..................   33
   10.2.6     45x Header Field Not Valid for Resource .............   33
   10.2.7     45x Invalid Range ...................................   33
   10.2.8     45x Parameter Is Read-Only ..........................   34
   11         Header Field Definitions ............................   34
   11.1       Accept ..............................................   34
   11.2       Accept-Encoding .....................................   35
   11.3       Accept-Language .....................................   35
   11.4       Allow ...............................................   36
   11.5       Authorization .......................................   36
   11.6       Bandwidth ...........................................   36
   11.7       Blocksize ...........................................   36
   11.8       Cache-Control .......................................   37
   11.9       Conference ..........................................   39
   11.10      Connection ..........................................   39
   11.11      Content-Encoding ....................................   39
   11.12      Content-Length ......................................   39
   11.13      Content-Type ........................................   39
   11.14      Date ................................................   40
   11.15      Expires .............................................   40
   11.16      If-Modified-Since ...................................   41
   11.17      Last-modified .......................................   41
   11.18      Location ............................................   41
   11.19      Nack-Transport-Require ..............................   41
   11.20      Range ...............................................   41
   11.21      Require .............................................   42
   11.22      Retry-After .........................................   43
   11.23      Scale ...............................................   43
   11.24      Speed ...............................................   44
   11.25      Server ..............................................   44
   11.26      Session .............................................   44
   11.27      Transport ...........................................   45
   11.28      Transport-Require ...................................   46
   11.29      Unsupported .........................................   46
   11.30      User-Agent ..........................................   47
   11.31      Via .................................................   47
   11.32      WWW-Authenticate ....................................   47
          R. Hinden and C. Partridge, ``Version 2 of the reliable data
          protocol (RDP),'' RFC 1151, Internet Engineering Task Force,
          Apr. 1990.
   11
          J. Postel, ``Transmission control protocol,'' STD 7, RFC 793,
          Internet Engineering Task Force, Sept. 1981.
   12         Caching .............................................   47
          M. Handley, H. Schulzrinne, and E. Schooler, ``SIP: Session
          initiation protocol,'' Internet Draft, Internet Engineering
          Task Force, Dec. 1996.
          Work in progress.
   13         Examples ............................................   48
   13.1       Media on Demand (Unicast) ...........................   48
   13.2       Live Media Presentation Using Multicast .............   49
   13.3       Playing media into an existing session ..............   50
   13.4       Recording ...........................................   51
          P. McMahon, ``GSS-API authentication method for SOCKS version
          5,'' RFC 1961, Internet Engineering Task Force, June 1996.
   14         Syntax ..............................................   52
   14.1       Base Syntax .........................................   52
          D. Crocker, ``Augmented BNF for syntax specifications: ABNF,''
          Internet Draft, Internet Engineering Task Force, Oct. 1996.
          Work in progress.
   15         Security Considerations .............................   52
   A          RTSP Protocol State Machines ........................   53
   A.1        Client State Machine ................................   54
   A.2        Server State
          R. Elz, ``A compact representation of IPv6 addresses,'' RFC
          1924, Internet Engineering Task Force, Apr. 1996.
   16
          T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform
          resource locators (URL),'' RFC 1738, Internet Engineering Task
          Force, Dec. 1994.
   17
          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.
   18
          ISO/IEC, ``Information technology - generic coding of moving
          pictures and associated audio informaiton - part 6: extension
          for digital storage media and control,'' Draft International
          Standard ISO 13818-6, International Organization for
          Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
          Nov. 1995.
   19
          H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
          ``RTP: a transport protocol for real-time applications,'' RFC
          1889, Internet Engineering Task Force, Jan. 1996.
   20
          J. Miller, P. Resnick, and D. Singer, ``Rating Services and
          Rating Systems(and Their Machine ................................   55
   B          Open Issues .........................................   56
   C          Changes .............................................   56
   D          Author Addresses ....................................   57
   E          Acknowledgements ....................................   57
   F          Bibliography ........................................   58 Readable Descriptions), ''
          REC-PICS-services-961031, Worldwide Web Consortium, Oct. 1996.
   21
          D. Connolly, R. Khare, H.F. Nielsen, ``PEP - an Extension
          Mechanism for HTTP", Internet draft, work-in-progress. W3C
          Draft WD-http-pep-970714
          http://www.w3.org/TR/WD-http-pep-970714, July, 1996.