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


                  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
   working documents as Internet-Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as ``work in progress''.

   To learn the current status of any Internet-Draft, please check the
   ``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
   Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
   munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or
   ftp.isi.edu (US West Coast).

   Distribution of this document is unlimited.

                                 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).

1 Introduction

1.1 Purpose

   The Real-Time Streaming Protocol (RTSP) establishes and controls



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   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, RTSP acts as a "network remote control" for
   multimedia servers.

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

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

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

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

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

        o 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].)

        o Both an RTSP server and client can issue requests.

        o Data is carried out-of-band, by a different protocol. (There
         is an exception to this.)

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

        o The Request-URI always contains the absolute URI. Because of
         backward compatibility with a historical blunder, HTTP/1.1
         carries only the absolute path in the request



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             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 for
        security reasons.

   Invitation of a media server to a conference: A media server can be
        "invited" to join an existing conference, either to play back
        media into the presentation or to record all or a subset of the
        media in a presentation. This mode is useful for distributed
        teaching applications. Several parties in the conference may
        take turns "pushing the remote control buttons".

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

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

1.2 Requirements

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

1.3 Terminology

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

   Conference: 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 for the purpose of communication.

   Continuous media: Data where there is a timing relationship between



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        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 is less strict.

   Participant: Participants are members 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 a presentation may originate from different media
        servers. A media server may reside on the same or a different
        host as the web server the presentation is invoked from.

   Media parameter: Parameter specific to a media type that may be
        changed while the stream is being played or prior to it.

   (Media) stream: A single media instance, e.g., an audio stream or a
        video stream as well as a single whiteboard or shared
        application group. When using RTP, a stream consists of all RTP
        and RTCP packets created by a source within an RTP session. This
        is equivalent to the definition of a DSM-CC stream.

   Message: The basic unit of RTSP communication, consisting of a
        structured sequence of octets matching the syntax defined in
        Section 14 and transmitted via a connection or a connectionless
        protocol.

   Presentation: A set of one or more streams which the server allows
        the client to manipulate together. A presentation has a single
        time axis for all streams belonging to it. Presentations are
        defined by presentation descriptions (see below). A presentation
        description contains RTSP URIs that define which streams can be
        controlled individually and an RTSP URI to control the whole
        presentation. A movie or live concert consisting of one or more
        audio and video streams is be an example of a presentation.

   Presentation description: A presentation description contains
        information about one or more media streams within a
        presentation, such as the set of encodings, network addresses
        and information about the content. Other IETF protocols such as
        SDP [6] use the term "session" for a live presentation. The
        presentation description may take several different formats,
        including but not limited to the session description format SDP.

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



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        indicated explicitly.

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

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

1.4 Protocol Properties

   RTSP has the following properties:

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

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

   Secure: RTSP re-uses web security mechanisms, either at the transport
        level (TLS [7]) 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 the
        different media servers.  Media synchronization is performed at
        the transport level.

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

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



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        editing.

   Presentation description neutral: The protocol does not impose a
        particular presentation description or metafile format and can
        convey the type of format to be used. However, the presentation
        description must contain at least one RTSP URI.

   Proxy and firewall friendly: The protocol should be readily handled
        by both application and transport-layer (SOCKS [13]) firewalls.
        A firewall may need to understand the  SETUP method 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 in most cases.

   Appropriate server control: If a client can start a stream, it must
        be able to stop a stream. Servers should not start streaming to
        clients in such a way that clients cannot stop the stream.

   Transport negotiation: The client can negotiate the transport method
        prior to actually needing to process a continuous media stream.

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


        An earlier requirement in RTSP' was multi-client
        capability.  However, it was determined that a better
        approach was to make sure that the protocol is easily
        extensible to the multi-client scenario. Stream identifiers
        can be used by several control streams, so that "passing
        the remote" would be possible. The protocol would not
        address how several clients negotiate access; this is left
        to either a "social protocol" or some other floor control
        mechanism.

1.5 Extending RTSP

   Since not all media servers have the same functionality, media



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   servers by necessity will support different sets of requests. For
   example:

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

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

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

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

   It is up to the creators of presentation descriptions not to ask the
   impossible of a server. This situation is similar in HTTP/1.1, 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, as long
         as these parameters can be safely ignored by the recipient.
         (This is equivalent to adding new parameters to an HTML tag.)

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

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

1.6 Overall Operation

   Each presentation and media stream may be identified by an RTSP URL.
   The overall presentation and the properties of the media the
   presentation is made up of are defined by a presentation description
   file, the format of which is outside the scope of this specification.
   The presentation description file may be obtained by the client using
   HTTP or other means such as email and may not necessarily be stored
   on the media server.

   For the purposes of this specification, a presentation description is
   assumed to describe one or more presentations, each of which
   maintains a common time axis. For simplicity of exposition and



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

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

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

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

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

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

1.7 RTSP States

   RTSP controls a stream which may be sent via a separate protocol,
   independent of the control channel. For example, RTSP control may
   occur on a TCP connection while the data flows via UDP. Thus, data
   delivery continues even if no RTSP requests are received by the media
   server.  Also, during its lifetime, a single media stream may be
   controlled by RTSP requests issued sequentially on different TCP
   connections.  Therefore, the server needs to maintain "session state"
   to be able to correlate RTSP requests with a stream. The state
   transitions are described in Section A.

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



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   TEARDOWN.

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

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

   PAUSE: Temporarily halts a stream, without freeing server resources.

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

1.8 Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP. It also may
   interact with HTTP in that the initial contact with streaming content
   is often to be made through a web page. The current protocol
   specification aims to allow different hand-off points between a web
   server and the media server implementing RTSP. For example, the
   presentation description can be retrieved using HTTP or RTSP. Having
   the presentation description be returned by the web server makes it
   possible to have the web server take care of authentication and
   billing, by handing out a presentation description whose media
   identifier includes an encrypted version of the requestor's IP
   address and a timestamp, with a shared secret between web and media
   server.

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


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

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

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



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2 Notational Conventions

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

   All the mechanisms specified in this document are described in both
   prose and an augmented Backus-Naur form (BNF) similar to that used 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 these paragraphs are marked with
   HS, AR and RL, designating opinions and comments by the individual
   authors which may not be shared by the co-authors and require
   resolution.

3 Protocol Parameters

3.1 RTSP Version

   applies, with HTTP replaced by RTSP.

3.2 RTSP URL

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


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

   The scheme  rtsp requires that commands are issued via a reliable
   protocol (within the Internet, TCP), while the scheme  rtspu
   identifies an unreliable protocol (within the Internet, UDP).



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

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

   A presentation or a stream is identified by an textual media
   identifier, using the character set and escape conventions [H3.2] of
   URLs [16]. Requests described in Section 9 can refer to either the
   whole presentation or an individual stream within the 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 several concurrent transmissions of the same
   content, an RTSP session , is indicated by the Session header field
   (Section 11.26) where needed.

   For example, the RTSP URL

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


   identifies the audio stream within the presentation "twister", which
   can be controlled via RTSP requests issued over a TCP connection to
   port 554 of host media.example.com


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

   The path components of the RTSP URL are opaque to the client and do
   not imply any particular file system structure for the server.


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

3.3 Conference Identifiers

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



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   unique. For H.323, the conferenceID value 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 server is participating in. These conferences are
        created by protocols outside the scope of this
        specification, e.g., H.323 [17] or SIP [12]. Instead of the
        RTSP client explicitly providing transport information, for
        example, it asks the media server to use the values in the
        conference description instead. If the conference
        participant inviting the media server would only supply a
        conference identifier which is unique for that inviting
        party, the media server could add an internal identifier
        for that party, e.g., its Internet address. However, this
        would prevent that the conference participant and the
        initiator of the RTSP commands are two different entities.

3.4 SMPTE Relative Timestamps

   A SMPTE relative time-stamp expresses time relative to the start of
   the clip. Relative timestamps are expressed as SMPTE time codes for
   frame-level access accuracy. The time code has the format
                       hours:minutes:seconds.frames
                                     ,
   with the origin at the start of the clip. For NTSC, the frame rate is
   29.97 frames per second. This is handled by dropping the first frame
   index of every minute, except every tenth minute. If the frame value
   is zero, it may 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





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3.5 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:  "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



3.6 Absolute Time

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


     utc-range = "clock" "=" utc-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



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   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



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             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 11.12) 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 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 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 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



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            | "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 the
   absolute URL (that is, including the scheme, host and port) rather
   than just the absolute path.

   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



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 in the table 1.

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


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





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6.1 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.
   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 of the
   attempt to understand and satisfy the request. These codes are fully
   defined in section10. 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:

        o 1xx: Informational - Request received, continuing process

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

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

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

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

   The individual values of the numeric status codes defined for
   RTSP/1.0, and an example set of corresponding Reason-Phrase 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



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   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
                     | "500"   ; Internal Server Error
                     | "501"   ; Not Implemented
                     | "502"   ; Bad Gateway
                     | "503"   ; Service Unavailable
                     | "504"   ; Gateway Time-out
                     | "505"   ; HTTP Version not supported
                     | extension-code

      extension-code = 3DIGIT

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




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   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 the entity-headers.

   In this section, both sender and recipient refer to either the client



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      Code    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
      410     Gone                             all
      411     Length Required                  SETUP
      412     Precondition Failed
      413     Request Entity Too Large         SETUP
      414     Request-URI Too Long             all
      415     Unsupported Media Type           SETUP
      45x     Only Valid for Stream            SETUP
      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
      _______________________________________________________________
      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






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   or the server, depending on who sends and who receives the entity.

7.1 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
                      | Content-Encoding         ; Section 14.12
                      | Content-Language         ; Section 14.13
                      | Content-Length           ; Section 14.14
                      | Content-Type             ; Section 14.18
                      | Expires                  ; Section 14.21
                      | Last-Modified            ; Section 14.29
                      | 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 Entity Body

   See [H7.2]

8 Connections

   RTSP requests can be transmitted in several different ways:

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

        o one connection per request/response transaction;

        o connectionless mode.

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



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   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 Pipelining

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

8.2 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". 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



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

9 Method Definitions

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


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


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


   Notes on Table 2:  PAUSE is recommend, 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


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   SHOULD not try this method again for this server.

9.1 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"

     S->C: RTSP/1.0 200 1 OK
           Date: 23 Jan 1997 15:35:06 GMT
           Nack-Transport-Require: switch-to-udp-control



   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",



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               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, the server may "push" a new description to
   the client, for example, if a new stream has become available. 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 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 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

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



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           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. 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 avoids having firewall to parse numerous different
        presentation description 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 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.

   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.




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     C->S: SETUP foo/bar/baz.rm RTSP/1.0 302
           Transport: rtp/udp;port=458

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



9.4  PLAY

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

   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.



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   The  Range header may also contain a  time parameter. This parameter
   specifies a time in UTC at which the playback should start.  If the
   message is received after the specified time, playback is started
   immediately. The  time parameter may be used to aid in
   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 format
   and MAY support the clock format.

9.5  PAUSE




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   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/1.0 834

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






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9.6  TEARDOWN

   Stop the stream delivery for the given URI, freeing the resources
   associated with it. If the URI is the root node 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/1.0 892

     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 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").

   Example:


     S->C: GET_PARAMETER /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






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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/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 for that URL.  It



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   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/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, 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/1.0 954
            Session: 1234
            Conference: 128.16.64.19/32492374



9.11 Embedded Binary Data

   Binary packets such as RTP data are encapsulated by an ASCII dollar
   sign (24 decimal), followed by a one-byte session 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.




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10 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 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 Client Error 4xx

10.2.1 451 Parameter Not Understood

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

10.2.2 452 Conference Not Found

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

10.2.3 453 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 45x Session Not Found

   The RTSP session identifier is invalid or has timed out.

10.2.5 45x Method Not Valid in This State

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

10.2.6 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 45x Invalid Range



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

10.2.8 45x Parameter Is Read-Only

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

11 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"
   designates request headers, type "r" response headers.  Fields marked
   with "req." in the column labeled "support" MUST be implemented by
   the recipient for a particular method, while fields marked "opt." are
   optional. Note that not all fields marked '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" 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] for syntax.

   Example of use:

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






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     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
     Blocksize             R       opt.      all but OPTIONS, TEARDOWN
     Cache-Control         Rr      opt.      SETUP
     Conference            R       opt.      SETUP
     Connection            Rr      req.      all
     Content-Encoding      R       req.      SET_PARAMETER
     Content-Encoding      r       req.      DESCRIBE
     Content-Length        R       req.      SET_PARAMETER
     Content-Length        r       req.      entity
     Content-Type          R       req.      SET_PARAMETER
     Content-Type          r       req.      entity
     Date                  Rr      opt.      all
     Expires               r       opt.      DESCRIBE
     If-Modified-Since     R       opt.      DESCRIBE, SETUP
     Last-Modified         r       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
     User-Agent            R       opt.      all
     Via                   Rr      opt.      all
     WWW-Authenticate      r       opt.      all


   Table 3: Overview of RTSP header fields

11.2 Accept-Encoding

   See [H14.3]

11.3 Accept-Language

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


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   content.

11.4 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 Authorization

   See [H14.8]

11.6 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.


     Bandwidth  = "Bandwidth" ":" 1*DIGIT



   Example:

     Bandwidth: 4000



11.7 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 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.



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11.8 Cache-Control

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

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

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

   [HS: Should there be an exception for 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



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        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) 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



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        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 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 Connection

   See [H14.10].

11.11 Content-Encoding

   See [H14.12]

11.12 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 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.



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11.14 Date

   See [H14.19].

11.15 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.



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11.16 If-Modified-Since

   The  If-Modified-Since request-header field is used with the DESCRIBE
   and  SETUP methods to make them conditional: if the requested variant
   has not been modified since the time specified in this field, a
   description will not be returned from the server ( DESCRIBE) or a
   stream will not be 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

   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 Location

   See [H14.30].

11.19 Nack-Transport-Require

   Negative acknowledgement of features not supported by the server. If
   there is a proxy on the path between the client and the server, the
   proxy MUST insert a message reply with an error message 506 (Feature
   not supported).


        HS: Same caveat as for Require applies.

11.20 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) and  clock (see Section 3.6) range units. Within



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   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.


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

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



   Example:

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




        The notation is similar to that used for the HTTP/1.1
        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 of features -- yet another name space. I believe
        this header field to be redundant. PEP should 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





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   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 in the case above). For a
   well-matched client-server pair, the interaction proceeds quickly,
   saving a round-trip often required by negotiation mechanisms. In
   addition, it also removes state ambiguity when the client requires
   features that the server doesn't understand.

11.22 Retry-After

   See [H14.38].

11.23 Scale

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

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

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

   If the request contains a  Range parameter, the new scale value will
   take effect at that time.


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



   Example of playing in reverse at 3.5 times normal rate:


     Scale: -3.5





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11.24 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 Server

   See [H14.39]

11.26 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.
   Once a client receives a Session identifier, it MUST return it for
   any request related to that session.


        HS: This may be redundant with the standards-track HTTP
        state maintenance mechanism [2]. The equivalent way of



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        doing this would be for the server to send Set-Cookie:
        Session="123"; Version=1; Path = "/twister" and for the
        client to return later Cookie: Session = "123"; $Version=1;
        $Path = "/twister" response to the TEARDOWN message, the
        server would simply send Set-Cookie: Session="123";
        Version=1; Max-Age=0 to get rid of the cookie on the client
        side. Cookies also have a time-out, so that a server may
        limit the lifetime of a session at will. Unlike a web
        browser, a client would not store these states on disk. To
        avoid privacy issues, we should prohibit the Host
        parameter.

11.27 Transport

   This request header indicates which transport protocol is to be used
   and configures its parameters such as multicast, 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, a  Transport header field
   (and the  SETUP request containing it) are not needed.

   in whatever protocol is being used by the control stream. Currently,
   the next-layer protocols RTP is defined. Parameters may be added to
   each protocol, separated by a semicolon. For RTP, the boolean
   parameter compressed is defined, indicating compressed RTP according
   to RFC XXXX. For multicast UDP, the integer parameter  ttl defines
   the time-to-live value to be used. The client may specify the
   multicast address with the  multicast parameter. A server SHOULD
   authenticate the client before allowing the client to direct a media
   stream to a multicast address not chosen by the server to avoid
   becoming the unwitting perpetrator of a denial-of-service attack. For
   UDP and TCP, the parameter  port defines the port data is to be sent
   to.

   The  SSRC parameter indicates the RTP SSRC 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



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   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 *parameter
     transport-protocol = "UDP" | "TCP"
     upper-layer  = "RTP"
     parameters = ";" "multicast" [ "=" mca ]
                | ";" "compressed"
                | ";" "interleaved"
                | ";" "ttl" "=" ttl
                | ";" "port" "=" port
                | ";" "ssrc" "=" ssrc
     ttl        = 1*3(DIGIT)
     port       = 1*5(DIGIT)
     ssrc       = 8*8(HEX)
     mca        = host



   Example:

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



11.28 Transport-Require

   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 to the client if not supported.

   See Section 11.21 for more details on the mechanics of 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.



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11.30 User-Agent

   See [H14.42]

11.31 Via

   See [H14.44].

11.32 WWW-Authenticate

   See [H14.46].

12 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. A cache MUST answer any
   DESCRIBE requests if it is currently serving the stream to the
   requestor, as it is possible that low-level details of the stream
   description may have changed on the origin-server.

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

   To the client, an RTSP proxy cache would appear like a regular media
   server, to the media origin server like a client. Just like an 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



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   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.

13 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.

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
   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 time line of the movie.

   In our example, the client is only interested in the last part of the
   movie. The server requires authentication for this movie. The audio
   track can be dynamically switched between between two sets of
   encodings.  The URL with scheme rtpsu indicates the media servers
   want to use UDP for exchanging RTSP messages.


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

   W->C: 200 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>



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         </session>

   C->A: SETUP rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 1
         Transport: rtp/udp;compression;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;compression;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/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 RTSP/1.0 3
         Session: 1234

   A->C: 200 3 OK

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

   V->C: 200 3 OK



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

13.2 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



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   description, while the media server M maintains the full description.
   During the RTSP session, a new subtitling stream is added.


   C->W: GET /concert 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>

   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 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



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

13.3 Playing media into an existing session




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   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
         Accept: application/rtsl, application/sdp

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

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

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



13.4 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
         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->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 91



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         Range: clock 19961110T1925-19961110T2015



14 Syntax

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

14.1 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 = "

   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 Security Considerations




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   The protocol offers the opportunity for a remote-control 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.

   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.

   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 URLs denoting an RTSP session comprised of
   multiple streams will have an effect on the individual states of all
   the substreams. For example, if the stream /movie contains two
   substreams /movie/audio and /movie/video, then the following command:


     PLAY /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 in URLs or a relation to the filesystem. See
        Section 3.2.

   The requests  OPTIONS,  DESCRIBE,  GET_PARAMETER, SET_PARAMETER do



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   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 client changes state on receipt of replies to requests. 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" 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
   status codes 401 (Unauthorized) and 402 (Payment Required), where the
   state remains unchanged and the request should be re-issued with the
   appropriate authentication or payment information. 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


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                 PLAY        Playing
                 RECORD      Recording
                 SETUP       Playing (changed transport)
   Recording     PAUSE       Init
                 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 server changes state on receiving requests. If the server is in
   state Playing or Recording and in unicast mode, it MAY revert to Init
   and tear down the RTSP session if it has not received "wellness"
   information, such as RTCP reports, 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.

   The  REDIRECT message, when sent, is effective immediately. If a
   similar change of location occurs at a certain time in the future,
   this is assumed to be indicated by the presentation description.

   SETUP is valid in states Init and Ready only. An error message should
   be returned in other cases. If no explicit SETUP is required for the
   object, state starts at READY, there are only two states READY and
   PLAYING.


   state        message     next state
   ___________________________________
   Init         SETUP       Ready
                TEARDOWN    Init
   Ready        PLAY        Playing
                SETUP       Ready
                TEARDOWN    Ready


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   Playing      PLAY        Playing
                PAUSE       Ready
                TEARDOWN    Ready
                RECORD      Recording
                SETUP       Playing
   Recording    RECORD      Recording
                PAUSE       Ready
                TEARDOWN    Ready
                PLAY        Playing
                SETUP       Recording


B Open Issues

        o Define text/rtsp-parameter MIME type.

        o 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 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 in
         Quicktime, AVI and ASF files?

        o 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
         to the HTTP extension mechanisms [19] in complexity, but is
         different.

        o Use HTTP absolute path + Host field or do the right thing and
         carry full URL, including host in request?

C Changes

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



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        o Various editorial changes and clarifications.

        o Removed references to SDF and replaced by RTSL.

        o Added  Scale general header.

        o Clarify behavior of PLAY.

        o Rename GET to DESCRIBE.

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

        o Rename CLOSE to TEARDOWN, in symmetry with SETUP.

        o Terminology adjusted to "presentation" and "stream".

        o Redundant syntax BNF in appendix removed since it just
         duplicates HTTP spec.

        o Beginnings of cache control.

   Changes are marked by changebars in the margins of the PostScript
   version.

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.
   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



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   This draft is based on the functionality of the RTSP draft. 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
   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.



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   [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 for SOCKS version 5,"
   RFC 1961, Internet Engineering Task Force, June 1996.

   [14] 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, 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] D. Connolly, "PEP: an extension mechanism for http," Internet
   Draft, Internet Engineering Task Force, Jan. 1997.  Work in progress.






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                           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
   5          Request .............................................   15
   6          Response ............................................   16
   6.1        Status-Line .........................................   17
   6.1.1      Status Code and Reason Phrase .......................   17
   6.1.2      Response Header Fields ..............................   19
   7          Entity ..............................................   19
   7.1        Entity Header Fields ................................   21
   7.2        Entity Body .........................................   21
   8          Connections .........................................   21
   8.1        Pipelining ..........................................   22
   8.2        Reliability and Acknowledgements ....................   22
   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



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   9.11       Embedded Binary Data ................................   32
   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
   12         Caching .............................................   47
   13         Examples ............................................   48
   13.1       Media on Demand (Unicast) ...........................   48



H. Schulzrinne, A. Rao, R. Lanphier                          [Page 61]


Internet Draft                    RTSP                    March 27, 1997


   13.2       Live Media Presentation Using Multicast .............   49
   13.3       Playing media into an existing session ..............   50
   13.4       Recording ...........................................   51
   14         Syntax ..............................................   52
   14.1       Base Syntax .........................................   52
   15         Security Considerations .............................   52
   A          RTSP Protocol State Machines ........................   53
   A.1        Client State Machine ................................   54
   A.2        Server State Machine ................................   55
   B          Open Issues .........................................   56
   C          Changes .............................................   56
   D          Author Addresses ....................................   57
   E          Acknowledgements ....................................   57
   F          Bibliography ........................................   58





































H. Schulzrinne, A. Rao, R. Lanphier                          [Page 62]


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