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

Versions: (RFC 2326) 00 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 34 35 36 37 38 39 40 RFC 7826

Internet Engineering Task Force                                MMUSIC WG
Internet Draft                                            H. Schulzrinne
                                                             Columbia U.
                                                                  A. Rao
                                                                   Cisco
                                                             R. Lanphier
                                                            RealNetworks
draft-ietf-mmusic-rfc2326bis-00.txt
February 22, 2002
Expires: July 2002


                  Real Time Streaming Protocol (RTSP)

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

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

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

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.

Abstract

   This memorandum is a revision of RFC 2326, which is currently a
   Proposed Standard.

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



H. Schulzrinne et. al.                                        [Page 1]


Internet Draft                    RTSP                 February 22, 2002


   mechanisms based upon RTP (RFC 1889).


















































H. Schulzrinne et. al.                                        [Page 2]


Internet Draft                    RTSP                 February 22, 2002


1 Introduction

1.1 Purpose

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

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

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

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

   The protocol is intentionally similar in syntax and operation to
   HTTP/1.1 [2] so that extension mechanisms to HTTP can in most cases
   also be added to RTSP. However, RTSP differs in a number of important
   aspects from HTTP:

        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.

        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].




H. Schulzrinne et. al.                                        [Page 3]


Internet Draft                    RTSP                 February 22, 2002


        o The Request-URI always contains the absolute URI. Because of
          backward compatibility with a historical blunder, HTTP/1.1 [2]
          carries only the absolute path in the request and puts the
          host name in a separate header field.


             This makes "virtual hosting" easier, where a single
             host with one IP address hosts several document trees.

   The protocol supports the following operations:

        Retrieval of media from media server: The client can request a
             presentation description via HTTP or some other method. If
             the presentation is being multicast, the presentation
             description contains the multicast addresses and ports to
             be used for the continuous media.  If the presentation is
             to be sent only to the client via unicast, the client
             provides the destination for security reasons.

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

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

   RTSP requests may be handled by proxies, tunnels and caches as in
   HTTP/1.1 [2].

1.2 Requirements

   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
   document are to be interpreted as described in RFC 2119 [4].

1.3 Terminology

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

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



H. Schulzrinne et. al.                                        [Page 4]


Internet Draft                    RTSP                 February 22, 2002


             message to control both the audio and video feeds.

        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.

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

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

        Entity: The information transferred as the payload of a request
             or response. An entity consists of metainformation in the
             form of entity-header fields and content in the form of an
             entity-body, as described in Section 8.

        Media initialization: Datatype/codec specific initialization.
             This includes such things as clockrates, color tables, etc.
             Any transport-independent information which is required by
             a client for playback of a media stream occurs in the media
             initialization phase of stream setup.

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

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

        Media server indirection: Redirection of a media client to a



H. Schulzrinne et. al.                                        [Page 5]


Internet Draft                    RTSP                 February 22, 2002


             different media server.

        (Media) stream: A single media instance, e.g., an audio stream
             or a video stream as well as a single whiteboard or shared
             application group. When using RTP, a stream consists of all
             RTP and RTCP packets created by a source within an RTP
             session. This is equivalent to the definition of a DSM-CC
             stream([5]).

        Message: The basic unit of RTSP communication, consisting of a
             structured sequence of octets matching the syntax defined
             in Section 15 and transmitted via a connection or a
             connectionless protocol.

        Participant: Member of a conference. A participant may be a
             machine, e.g., a media record or playback server.

        Presentation: A set of one or more streams presented to the
             client as a complete media feed, using a presentation
             description as defined below. In most cases in the RTSP
             context, this implies aggregate control of those streams,
             but does not have to.

        Presentation description: A presentation description contains
             information about one or more media streams within a
             presentation, such as the set of encodings, network
             addresses and information about the content. Other IETF
             protocols such as SDP (RFC 2327 [6]) use the term "session"
             for a live presentation. The presentation description may
             take several different formats, including but not limited
             to the session description format SDP.

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

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

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

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




H. Schulzrinne et. al.                                        [Page 6]


Internet Draft                    RTSP                 February 22, 2002


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, RFC 2246 [7]) or within the protocol
             itself. All HTTP authentication mechanisms such as basic
             (RFC 2068 [2]) and digest authentication (RFC 2069 [8]) are
             directly applicable.

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

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

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

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

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

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



H. Schulzrinne et. al.                                        [Page 7]


Internet Draft                    RTSP                 February 22, 2002


             URI.

        Proxy and firewall friendly: The protocol should be readily
             handled by both application and transport-layer (SOCKS
             [14]) firewalls. A firewall may need to understand the
             SETUP method to open a "hole" for the UDP media stream.

        HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so
             that the existing infrastructure can be reused. This
             infrastructure includes PICS (Platform for Internet Content
             Selection [15,16]) for associating labels with content.
             However, RTSP does not just add methods to HTTP since the
             controlling continuous media requires server state in most
             cases.

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

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

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


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

1.5 Extending RTSP

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

        o A server may only be capable of playback thus has no need to



H. Schulzrinne et. al.                                        [Page 8]


Internet Draft                    RTSP                 February 22, 2002


          support the  RECORD request.

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

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

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

   It is up to the creators of presentation descriptions not to ask the
   impossible of a server. This situation is similar in HTTP/1.1 [2],
   where the methods described in [H19.6] are not likely to be supported
   across all servers.

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

        o Existing methods can be extended with new parameters, as long
          as these parameters can be safely ignored by the recipient.
          (This is equivalent to adding new parameters to an HTML tag.)
          If the client needs negative acknowledgement when a method
          extension is not supported, a tag corresponding to the
          extension may be added in the  Require:  field (see Section
          12.33).

        o New methods can be added. If the recipient of the message does
          not understand the request, it responds with error code 501
          (Not Implemented) and the sender should not attempt to use
          this method again.  A client may also use the  OPTIONS method
          to inquire about methods supported by the server. The server
          SHOULD list the methods it supports using the  Public response
          header.

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

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.




H. Schulzrinne et. al.                                        [Page 9]


Internet Draft                    RTSP                 February 22, 2002


   For the purposes of this specification, a presentation description is
   assumed to describe one or more presentations, each of which
   maintains a common time axis. For simplicity of exposition and
   without loss of generality, it is assumed that the presentation
   description contains exactly one such presentation. A presentation
   may contain several media streams.

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

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

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

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

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

1.7 RTSP States

   RTSP controls a stream which may be sent via a separate protocol,
   independent of the control channel. For example, RTSP control may
   occur on a TCP connection while the data flows via UDP. Thus, data
   delivery continues even if no RTSP requests are received by the media
   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



H. Schulzrinne et. al.                                       [Page 10]


Internet Draft                    RTSP                 February 22, 2002


   transitions are described in Section A.

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

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

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

        PAUSE: Temporarily halts a stream without freeing server
             resources.

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

             RTSP methods that contribute to state use the  Session
             header field (Section 12.38) to identify the RTSP session
             whose state is being manipulated. The server generates
             session identifiers in response to  SETUP requests (Section
             10.4).

1.8 Relationship with Other Protocols

   RTSP has some overlap in functionality with HTTP. It also may
   interact with HTTP in that the initial contact with streaming content
   is often to be made through a web page. The current protocol
   specification aims to allow different hand-off points between a web
   server and the media server implementing RTSP. For example, the
   presentation description can be retrieved using HTTP or RTSP, which
   reduces roundtrips in web-browser-based scenarios, yet also allows
   for standalone RTSP servers and clients which do not rely on HTTP at
   all.

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


        Re-using HTTP functionality has advantages in at least two
        areas, namely security and proxies. The requirements are



H. Schulzrinne et. al.                                       [Page 11]


Internet Draft                    RTSP                 February 22, 2002


        very similar, so having the ability to adopt HTTP work on
        caches, proxies and authentication is valuable.

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

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

2 Notational Conventions

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

   All the mechanisms specified in this document are described in both
   prose and an augmented Backus-Naur form (BNF) similar to that used in
   [H2.1]. It is described in detail in RFC 2234 [17], with the
   difference that this RTSP specification maintains the "1#" notation
   for comma-separated lists.

   In this draft, we use indented and smaller-type paragraphs to provide
   background and motivation. This is intended to give readers who were
   not involved with the formulation of the specification an
   understanding of why things are the way that they are in RTSP.

3 Protocol Parameters

3.1 RTSP Version

   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 [18]>
   port                                                _  *DIGIT




H. Schulzrinne et. al.                                       [Page 12]


Internet Draft                    RTSP                 February 22, 2002


   abs_path is defined in [H3.2.1].


        Note that fragment and query identifiers do not have a
        well-defined meaning at this time, with the interpretation
        left to the RTSP server.

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

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

   The use of IP addresses in URLs SHOULD be avoided whenever possible
   (see RFC 1924 [19]).

   A presentation or a stream is identified by a textual media
   identifier, using the character set and escape conventions [H3.2] of
   URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of
   streams, i.e., a presentation. Accordingly, requests described in
   Section 10 can apply to either the whole presentation or an
   individual stream within the presentation. Note that some request
   methods can only be applied to streams, not presentations and vice
   versa.

   For example, the RTSP URL:

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


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

   Also, the RTSP URL:

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


   identifies the presentation "twister", which may be composed of audio
   and video streams.


        This does not imply a standard way to reference streams in



H. Schulzrinne et. al.                                       [Page 13]


Internet Draft                    RTSP                 February 22, 2002


        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
   unique. For H.323, the conferenceID value is to be used.


   conference-id _  1*xchar



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

3.4 Session Identifiers

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


   session-id _  8*( ALPHA | DIGIT | safe )


3.5 SMPTE Relative Timestamps

   A SMPTE relative timestamp expresses time relative to the start of



H. Schulzrinne et. al.                                       [Page 14]


Internet Draft                    RTSP                 February 22, 2002


   the clip. Relative timestamps are expressed as SMPTE time codes for
   frame-level access accuracy. The time code has the format
                  hours:minutes:seconds:frames.subframes
                                     ,
   with the origin at the start of the clip. The default smpte format
   is"SMPTE 30 drop" format, with frame rate is 29.97 frames per second.
   Other SMPTE codes MAY be supported (such as "SMPTE 25") through the
   use of alternative use of "smpte time". For the "frames" field in the
   time value can assume the values 0 through 29. The difference between
   30 and 29.97 frames per second is handled by dropping the first two
   frame indices (values 00 and 01) of every minute, except every tenth
   minute. If the frame value is zero, it may be omitted. Subframes are
   measured in one-hundredth of a frame.


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


   Examples:

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



3.6 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position
   relative to the beginning of the presentation. The timestamp consists
   of a decimal fraction. The part left of the decimal may be expressed
   in either seconds or hours, minutes, and seconds. The part right of
   the decimal point measures fractions of a second.

   The beginning of a presentation corresponds to 0.0 seconds. Negative
   values are not defined. The special constant  now is defined as the
   current instant of a live event. It may be used only for live events.

   NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the
   viewer associates with a program. It is often digitally displayed on
   a VCR.  NPT advances normally when in normal play mode (scale = 1),
   advances at a faster rate when in fast scan forward (high positive



H. Schulzrinne et. al.                                       [Page 15]


Internet Draft                    RTSP                 February 22, 2002


   scale ratio), decrements when in scan reverse (high negative scale
   ratio) and is fixed in pause mode. NPT is (logically) equivalent to
   SMPTE time codes." [5]

   npt-range                                            _  ["npt" "="] npt-range-spec
   ; implementations SHOULD use npt= prefix, but SHOULD
   ; be prepared to interoperate with RFC 2326
   ; implementations which don't use it
   npt-range-spec                                       _  ( npt-time "-" [ npt-time ] ) | ( "-" npt-time )
   npt-time                                             _  "now" | npt-sec | npt-hhmmss
   npt-sec                                              _  1*DIGIT [ "." *DIGIT ]
   npt-hhmmss                                           _  npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
   npt-hh                                               _  1*DIGIT                                          ; any positive number
   npt-mm                                               _  1*2DIGIT                                         ; 0-59
   npt-ss                                               _  1*2DIGIT                                         ; 0-59


   Examples:

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




        The syntax conforms to ISO 8601. The npt-sec notation is
        optimized for automatic generation, the ntp-hhmmss notation
        for consumption by human readers. The "now" constant allows
        clients to request to receive the live feed rather than the
        stored or time-delayed version. This is needed since
        neither absolute time nor zero time are appropriate for
        this case.

3.7 Absolute Time

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


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


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



H. Schulzrinne et. al.                                       [Page 16]


Internet Draft                    RTSP                 February 22, 2002


     19961108T143720.25Z



3.8 Option Tags

   Option tags are unique identifiers used to designate new options in
   RTSP. These tags are used in in  Require (Section 12.33) and  Proxy-
   Require (Section 12.28) header fields.

   Syntax:

   option-tag _  token


   The creator of a new RTSP option should either prefix the option with
   a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name
   for a feature whose inventor can be reached at "foo.com"), or
   register the new option with the Internet Assigned Numbers Authority
   (IANA).

3.8.1 Registering New Option Tags with IANA

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

        o Name and description of option. The name may be of any length,
          but SHOULD be no more than twenty characters long. The name
          MUST not contain any spaces, control characters or periods.

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

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

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

4 RTSP Message

   RTSP is a text-based protocol and uses the ISO 10646 character set in
   UTF-8 encoding (RFC 2279 [22]). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by



H. Schulzrinne et. al.                                       [Page 17]


Internet Draft                    RTSP                 February 22, 2002


   themselves as line terminators.


        Text-based protocols make it easier to add optional
        parameters in a self-describing manner. Since the number of
        parameters and the frequency of commands is low, processing
        efficiency is not a concern. Text-based protocols, if done
        carefully, also allow easy implementation of research
        prototypes in scripting languages such as Tcl, Visual Basic
        and Perl.

   The 10646 character set avoids tricky character set switching, but is
   invisible to the application as long as US-ASCII is being used. This
   is also the encoding used for RTCP. ISO 8859-1 translates directly
   into Unicode with a high-order octet of zero. ISO 8859-1 characters
   with the most-significant bit set are represented as 1100001x
   10xxxxxx.  (See RFC 2279 [22])

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

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

4.1 Message Types

   See [H4.1]

4.2 Message Headers

   See [H4.2]

4.3 Message Body

   See [H4.3]

4.4 Message Length

   When a message body is included with a message, the length of that
   body is determined by one of the following (in order of precedence):

        1.   Any response message which MUST NOT include a message body
             (such as the 1xx, 204, and 304 responses) is always
             terminated by the first empty line after the header fields,
             regardless of the entity-header fields present in the
             message. (Note: An empty line consists of only CRLF.)



H. Schulzrinne et. al.                                       [Page 18]


Internet Draft                    RTSP                 February 22, 2002


        2.   If a  Content-Length header field (section 12.15) is
             present, its value in bytes represents the length of the
             message-body. If this header field is not present, a value
             of zero is assumed.

   Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
   transfer coding(see [H3.6]) and requires the presence of the
   Content-Length header field.


        Given the moderate length of presentation descriptions
        returned, the server should always be able to determine its
        length, even if it is generated dynamically, making the
        chunked transfer encoding unnecessary.

5 General Header Fields

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


   general-header      =       Cache-Control      ; Section 12.9
   |               Connection   ; Section 12.11
   |                  CSeq      ; Section 12.18
   |                  Date      ; Section 12.19
   |                  Via       ; Section 12.44


6 Request

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


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


6.1 Request Line


   Request-Line _  Method SP Request-URI SP RTSP-Version CRLF



H. Schulzrinne et. al.                                       [Page 19]


Internet Draft                    RTSP                 February 22, 2002


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



   extension-method _  token
   Request-URI      _  "*" | absolute_URI
   RTSP-Version     _  "RTSP" "/" 1*DIGIT "." 1*DIGIT


6.2 Request Header Fields


   request-header          =          Accept             ; Section 12.1
   |                Accept-Encoding    ; Section 12.2
   |                Accept-Language    ; Section 12.3
   |                 Authorization     ; Section 12.6
   |                   Bandwidth       ; Section 12.7
   |                   Blocksize       ; Section 12.8
   |                  Conference       ; Section 12.10
   |                     From          ; Section 12.21
   |               If-Modified-Since   ; Section 12.24
   |                 Proxy-Require     ; Section 12.28
   |                     Range         ; Section 12.30
   |                    Referer        ; Section 12.31
   |                    Require        ; Section 12.33
   |                     Scale         ; Section 12.35
   |                    Session        ; Section 12.38
   |                     Speed         ; Section 12.36
   |                   Transport       ; Section 12.40
   |                  User-Agent       ; Section 12.42


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





H. Schulzrinne et. al.                                       [Page 20]


Internet Draft                    RTSP                 February 22, 2002


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

   The asterisk "*" in the Request-URI means that the request does not
   apply to a particular resource, but to the server itself, and is only
   allowed when the method used does not necessarily apply to a
   resource.  One example would be:


     OPTIONS * RTSP/1.0



7 Response

   [H6] applies except that  HTTP-Version is replaced by RTSP-Version.
   Also, RTSP defines additional status codes and does not define some
   HTTP codes. The valid response codes and the methods they can be used
   with are defined in Table 1.

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


   Response                 =         Status-Line       ;  Section 7.1
   *(                general-header   ;  Section 5
   |                 response-header  ;  Section 7.1.2
   |                 entity-header )  ;  Section 8.1
   CRLF
   [ message-body ]  ;  Section 4.3


7.1 Status-Line

   The first line of a Response message is the  Status-Line, consisting
   of the protocol version followed by a numeric status code, and the
   textual phrase associated with the status code, with each element
   separated by  SP characters. No  CR or LF is allowed except in the
   final CRLF sequence.


   Status-Line _  RTSP-Version SP Status-Code SP Reason-Phrase CRLF


7.1.1 Status Code and Reason Phrase



H. Schulzrinne et. al.                                       [Page 21]


Internet Draft                    RTSP                 February 22, 2002


   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request. These codes are fully
   defined in Section 11. The  Reason-Phrase is intended to give a short
   textual description of the Status-Code. The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human
   user. The client is not required to examine or display the  Reason-
   Phrase.

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



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



H. Schulzrinne et. al.                                       [Page 22]


Internet Draft                    RTSP                 February 22, 2002


        |                "402"       ;  Payment Required
        |                "403"       ;  Forbidden
        |                "404"       ;  Not Found
        |                "405"       ;  Method Not Allowed
        |                "406"       ;  Not Acceptable
        |                "407"       ;  Proxy Authentication Required
        |                "408"       ;  Request Time-out
        |                "410"       ;  Gone
        |                "411"       ;  Length Required
        |                "412"       ;  Precondition Failed
        |                "413"       ;  Request Entity Too Large
        |                "414"       ;  Request-URI Too Large
        |                "415"       ;  Unsupported Media Type
        |                "451"       ;  Parameter Not Understood
        |                "452"       ;  Conference Not Found
        |                "453"       ;  Not Enough Bandwidth
        |                "454"       ;  Session Not Found
        |                "455"       ;  Method Not Valid in This State
        |                "456"       ;  Header Field Not Valid for Resource
        |                "457"       ;  Invalid Range
        |                "458"       ;  Parameter Is Read-Only
        |                "459"       ;  Aggregate operation not allowed
        |                "460"       ;  Only aggregate operation allowed
        |                "461"       ;  Unsupported transport
        |                "462"       ;  Destination unreachable
        |                "500"       ;  Internal Server Error
        |                "501"       ;  Not Implemented
        |                "502"       ;  Bad Gateway
        |                "503"       ;  Service Unavailable
        |                "504"       ;  Gateway Time-out
        |                "505"       ;  RTSP Version not supported
        |                "551"       ;  Option not supported
        |            extension-code



        extension-code  =  3DIGIT
        Reason-Phrase   =  *<TEXT,  excluding CR, LF>


   RTSP status codes are extensible. RTSP applications are not required
   to understand the meaning of all registered status codes, though such
   understanding is obviously desirable. However, applications MUST
   understand the class of any status code, as indicated by the first
   digit, and treat any unrecognized response as being equivalent to the
   x00 status code of that class, with the exception that an
   unrecognized response MUST NOT be cached. For example, if an
   unrecognized status code of 431 is received by the client, it can



H. Schulzrinne et. al.                                       [Page 23]


Internet Draft                    RTSP                 February 22, 2002


   safely assume that there was something wrong with its request and
   treat the response as if it had received a 400 status code. In such
   cases, user agents SHOULD present to the user the entity returned
   with the response, since that entity is likely to include human-
   readable information which will explain the unusual status.

7.1.2 Response Header Fields

   The response-header fields allow the request recipient to pass
   additional information about the response which cannot be placed in
   the Status-Line. These header fields give information about the
   server and about further access to the resource identified by the
   Request-URI.


   response-header          =           Location           ; Section 12.26
   |                Proxy-Authenticate   ; Section 12.27
   |                      Public         ; Section 12.29
   |                      Range          ; Section 12.30
   |                   Retry-After       ; Section 12.32
   |                     RTP-Info        ; Section 12.34
   |                      Scale          ; Section 12.35
   |                     Session         ; Section 12.38
   |                      Server         ; Section 12.37
   |                      Speed          ; Section 12.36
   |                    Transport        ; Section 12.40
   |                   Unsupported       ; Section 12.41
   |                       Vary          ; Section 12.43
   |                 WWW-Authenticate    ; Section 12.45


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

8 Entity

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

   In this section, both sender and recipient refer to either the client
   or the server, depending on who sends and who receives the entity.




H. Schulzrinne et. al.                                       [Page 24]


Internet Draft                    RTSP                 February 22, 2002


8.1 Entity Header Fields

   Entity-header fields define optional metainformation about the
   entity-body or, if no body is present, about the resource identified
   by the request.


   entity-header            =          Allow              ; Section 12.5
   |                   Content-Base     ; Section 12.12
   |                 Content-Encoding   ; Section 12.13
   |                 Content-Language   ; Section 12.14
   |                  Content-Length    ; Section 12.15
   |                 Content-Location   ; Section 12.16
   |                   Content-Type     ; Section 12.17
   |                     Expires        ; Section 12.20
   |                  Last-Modified     ; Section 12.25
   |                 extension-header
   extension-header         =          message-header


   The extension-header mechanism allows additional entity-header fields
   to be defined without changing the protocol, but these fields cannot
   be assumed to be recognizable by the recipient. Unrecognized header
   fields SHOULD be ignored by the recipient and forwarded by proxies.

8.2 Entity Body

   See [H7.2]

9 Connections

   RTSP requests can be transmitted in several different ways:

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

        o one connection per request/response transaction;

        o connectionless mode.

   The type of transport connection is defined by the RTSP URI (Section
   3.2). For the scheme "rtsp", a persistent connection is assumed,
   while the scheme "rtspu" calls for RTSP requests to be sent without
   setting up a connection.

   Unlike HTTP, RTSP allows the media server to send requests to the
   media client. However, this is only supported for persistent
   connections, as the media server otherwise has no reliable way of



H. Schulzrinne et. al.                                       [Page 25]


Internet Draft                    RTSP                 February 22, 2002



          Code  reason
          _______________________________________________________
          100   Continue                          all

_______________________________________________________
          200   OK                                all
          201   Created                           RECORD
          250   Low on Storage Space              RECORD
          _______________________________________________________
          300   Multiple Choices                  all
          301   Moved Permanently                 all
          302   Moved Temporarily                 all
          303   See Other                         all
          305   Use Proxy                         all

_______________________________________________________
          400   Bad Request                       all
          401   Unauthorized                      all
          402   Payment Required                  all
          403   Forbidden                         all
          404   Not Found                         all
          405   Method Not Allowed                all
          406   Not Acceptable                    all
          407   Proxy Authentication Required     all
          408   Request Timeout                   all
          410   Gone                              all
          411   Length Required                   all
          412   Precondition Failed               DESCRIBE, SETUP
          413   Request Entity Too Large          all
          414   Request-URI Too Long              all
          415   Unsupported Media Type            all
          451   Parameter Not Understood          SETUP
          452   Illegal Conference Identifier     SETUP
          453   Not Enough Bandwidth              SETUP
          454   Session Not Found                 all
          455   Method Not Valid In This State    all
          456   Header Field Not Valid            all
          457   Invalid Range                     PLAY
          458   Parameter Is Read-Only            SET_PARAMETER
          459   Aggregate Operation Not Allowed   all
          460   Only Aggregate Operation Allowed  all
          461   Unsupported Transport             all
          462   Destination Unreachable           all
          _______________________________________________________
          500   Internal Server Error             all
          501   Not Implemented                   all
          502   Bad Gateway                       all
          503   Service Unavailable               all
          504   Gateway Timeout                   all
          505   RTSP Version Not Supported        all
          551   Option not support                all

   Table 1: Status codes and their usage with RTSP methods

H. Schulzrinne et. al.                                       [Page 26]


Internet Draft                    RTSP                 February 22, 2002


   reaching the client.  Also, this is the only way that requests from
   media server to client are likely to traverse firewalls.

9.1 Pipelining

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

9.2 Reliability and Acknowledgements

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

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


        If both the underlying reliable transport such as TCP and
        the RTSP application retransmit requests, it is possible
        that each packet loss results in two retransmissions. The
        receiver cannot typically take advantage of the
        application-layer retransmission since the transport stack
        will not deliver the application-layer retransmission
        before the first attempt has reached the receiver. If the
        packet loss is caused by congestion, multiple
        retransmissions at different layers will exacerbate the
        congestion.

   If RTSP is used over a small-RTT LAN, standard procedures for
   optimizing initial TCP round trip estimates, such as those used in
   T/TCP (RFC 1644) [23], can be beneficial.

   The  Timestamp header (Section 12.39) is used to avoid the
   retransmission ambiguity problem [24] and obviates the need for
   Karn's algorithm.

   Each request carries a sequence number in the  CSeq header (Section
   12.18), which is incremented by one for each distinct request
   transmitted. If a request is repeated because of lack of
   acknowledgement, the request MUST carry the original sequence number
   (i.e., the sequence number is not incremented).



H. Schulzrinne et. al.                                       [Page 27]


Internet Draft                    RTSP                 February 22, 2002


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

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

10 Method Definitions

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


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


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


   Notes on Table 2: PAUSE is recommended, but not required in that a
   fully functional server can be built that does not support this
   method, for example, for live feeds. If a server does not support a
   particular method, it MUST return 501 (Not Implemented) and a client
   SHOULD not try this method again for this server.

10.1 OPTIONS

   The behavior is equivalent to that described in [H9.2]. An OPTIONS
   request may be issued at any time, e.g., if the client is about to


H. Schulzrinne et. al.                                       [Page 28]


Internet Draft                    RTSP                 February 22, 2002


   try a nonstandard request. It does not influence server state.

   Example:


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

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



   Note that these are necessarily fictional features (one would hope
   that we would not purposefully overlook a truly useful feature just
   so that we could have a strong example in this section).

10.2 DESCRIBE

   The DESCRIBE method retrieves the description of a presentation or
   media object identified by the request URL from a server. It may use
   the  Accept header to specify the description formats that the client
   understands. The server responds with a description of the requested
   resource. The DESCRIBE reply-response pair constitutes the media
   initialization phase of RTSP.

   Example:


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

     S->C: RTSP/1.0 200 OK
           CSeq: 312
           Date: 23 Jan 1997 15:35:06 GMT
           Content-Type: application/sdp
           Content-Length: 376

           v=0
           o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
           s=SDP Seminar
           i=A Seminar on the session description protocol
           u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
           e=mjh@isi.edu (Mark Handley)



H. Schulzrinne et. al.                                       [Page 29]


Internet Draft                    RTSP                 February 22, 2002


           c=IN IP4 224.2.17.12/127
           t=2873397496 2873404696
           a=recvonly
           m=audio 3456 RTP/AVP 0
           m=video 2232 RTP/AVP 31
           m=whiteboard 32416 UDP WB
           a=orient:portrait



   The DESCRIBE response MUST contain all media initialization
   information for the resource(s) that it describes. If a media client
   obtains a presentation description from a source other than DESCRIBE
   and that description contains a complete set of media initialization
   parameters, the client SHOULD use those parameters and not then
   request a description for the same media via RTSP.

   Additionally, servers SHOULD NOT use the DESCRIBE response as a means
   of media indirection.


        By forcing a DESCRIBE response to contain all media
        initialization for the set of streams that it describes,
        and discouraging use of DESCRIBE for media indirection, we
        avoid looping problems that might result from other
        approaches.

   Media initialization is a requirement for any RTSP-based system, but
   the RTSP specification does not dictate that this must be done via
   the DESCRIBE method. There are three ways that an RTSP client may
   receive initialization information:

        o via RTSP's DESCRIBE method;

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

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

   It is RECOMMENDED that minimal servers support the DESCRIBE method,
   and highly recommended that minimal clients support the ability to
   act as a "helper application" that accepts a media initialization
   file from standard input, command line, and/or other means that are
   appropriate to the operating environment of the client.

10.3 ANNOUNCE




H. Schulzrinne et. al.                                       [Page 30]


Internet Draft                    RTSP                 February 22, 2002


   The ANNOUNCE method serves two purposes:

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

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

   Example:


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

           v=0
           o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
           s=SDP Seminar
           i=A Seminar on the session description protocol
           u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
           e=mjh@isi.edu (Mark Handley)
           c=IN IP4 224.2.17.12/127
           t=2873397496 2873404696
           a=recvonly
           m=audio 3456 RTP/AVP 0
           m=video 2232 RTP/AVP 31

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



10.4 SETUP

   The SETUP request for a URI specifies the transport mechanism to be
   used for the streamed media. A client can issue a SETUP request for a
   stream that is already playing to change transport parameters, which
   a server MAY allow. If it does not allow this, it MUST respond with
   error 455 (Method Not Valid In This State). For the benefit of any
   intervening firewalls, a client must indicate the transport
   parameters even if it has no influence over these parameters, for



H. Schulzrinne et. al.                                       [Page 31]


Internet Draft                    RTSP                 February 22, 2002


   example, where the server advertises a fixed multicast address.


        Since SETUP includes all transport initialization
        information, firewalls and other intermediate network
        devices (which need this information) are spared the more
        arduous task of parsing the DESCRIBE response, which has
        been reserved for media initialization.

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


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

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



   The server generates session identifiers in response to SETUP
   requests. If a SETUP request to a server includes a session
   identifier, the server MUST bundle this setup request into the
   existing session or return error 459 (Aggregate Operation Not
   Allowed) (see Section 11.4.10).

10.5 PLAY

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

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

        This allows precise editing.  For example, regardless of



H. Schulzrinne et. al.                                       [Page 32]


Internet Draft                    RTSP                 February 22, 2002


        how closely spaced the two PLAY requests in the example
        below arrive, the server will first play seconds 10 through
        15, then, immediately following, seconds 20 to 25, and
        finally seconds 30 through the end.


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

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

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



   See the description of the PAUSE request for further examples.

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

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

   For a on-demand stream, the server replies with the actual range that
   will be played back. This may differ from the requested range if
   alignment of the requested range to valid frame boundaries is
   required for the media source. If no range is specified in the
   request, the current position is returned in the reply. The unit of
   the range in the reply is the same as that in the request.

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

   The following example plays the whole presentation starting at SMPTE



H. Schulzrinne et. al.                                       [Page 33]


Internet Draft                    RTSP                 February 22, 2002


   time code 0:10:20 until the end of the clip. The playback is to start
   at 15:36 on 23 Jan 1997.


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

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



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


     C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
           CSeq: 835
           Session: 12345678
           Range: clock=19961108T142300Z-19961108T143520Z

     S->C: RTSP/1.0 200 OK
           CSeq: 835
           Date: 23 Jan 1997 15:35:06 GMT




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

10.6 PAUSE

   The PAUSE request causes the stream delivery to be interrupted
   (halted) temporarily. If the request URL names a stream, only
   playback and recording of that stream is halted. For example, for
   audio, this is equivalent to muting. If the request URL names a
   presentation or group of streams, delivery of all currently active
   streams within the presentation or group is halted. After resuming
   playback or recording, synchronization of the tracks MUST be
   maintained. Any server resources are kept, though servers MAY close
   the session and free resources after being paused for the duration
   specified with the timeout parameter of the  Session header in the
   SETUP message.



H. Schulzrinne et. al.                                       [Page 34]


Internet Draft                    RTSP                 February 22, 2002


   Example:


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

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



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

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

   For example, if the server has play requests for ranges 10 to 15 and
   20 to 29 pending and then receives a pause request for NPT 21, it
   would start playing the second range and stop at NPT 21. If the pause
   request is for NPT 12 and the server is playing at NPT 13 serving the
   first play request, the server stops immediately. If the pause
   request is for NPT 16, the server stops after completing the first
   play request and discards the second play request.

   As another example, if a server has received requests to play ranges
   10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
   request for NPT=14 would take effect while the server plays the first
   range, with the second PLAY request effectively being ignored,
   assuming the PAUSE request arrives before the server has started
   playing the second, overlapping range. Regardless of when the PAUSE
   request arrives, it sets the NPT to 14.

   If the server has already sent data beyond the time specified in the



H. Schulzrinne et. al.                                       [Page 35]


Internet Draft                    RTSP                 February 22, 2002


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

10.7 TEARDOWN

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

   Example:


     C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 892
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 892



10.8 GET_PARAMETER

   The GET_PARAMETER request retrieves the value of a parameter of a
   presentation or stream specified in the URI. The content of the reply
   and response is left to the implementation. GET_PARAMETER with no
   entity body may be used to test client or server liveness ("ping").

   Example:


     S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 431
           Content-Type: text/parameters
           Session: 12345678
           Content-Length: 15

           packets_received
           jitter

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



H. Schulzrinne et. al.                                       [Page 36]


Internet Draft                    RTSP                 February 22, 2002



           packets_received: 10
           jitter: 0.3838




        The "text/parameters" section is only an example type for
        parameter. This method is intentionally loosely defined
        with the intention that the reply content and response
        content will be defined after further experimentation.

10.9 SET_PARAMETER

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

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

   Note: transport parameters for the media stream MUST only be set with
   the SETUP command.

        Restricting setting transport parameters to SETUP is for
        the benefit of firewalls.


        The parameters are split in a fine-grained fashion so that
        there can be more meaningful error indications. However, it
        may make sense to allow the setting of several parameters
        if an atomic setting is desirable. Imagine device control
        where the client does not want the camera to pan unless it
        can also tilt to the right angle at the same time.

   Example:


     C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
           CSeq: 421
           Content-length: 20
           Content-type: text/parameters

           barparam: barstuff




H. Schulzrinne et. al.                                       [Page 37]


Internet Draft                    RTSP                 February 22, 2002


     S->C: RTSP/1.0 451 Parameter Not Understood
           CSeq: 421
           Content-length: 10
           Content-type: text/parameters

           barparam




        The "text/parameters" section is only an example type for
        parameter. This method is intentionally loosely defined
        with the intention that the reply content and response
        content will be defined after further experimentation.

10.10 REDIRECT

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

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


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



10.11 RECORD

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

   The server decides whether to store the recorded data under the
   request-URI or another URI. If the server does not use the request-
   URI, the response SHOULD be 201 (Created) and contain an entity which



H. Schulzrinne et. al.                                       [Page 38]


Internet Draft                    RTSP                 February 22, 2002


   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 rtsp://example.com/meeting/audio.en RTSP/1.0
           CSeq: 954
           Session: 12345678
           Conference: 128.16.64.19/32492374



10.12 Embedded (Interleaved) Binary Data

   Certain firewall designs and other circumstances may force a server
   to interleave RTSP methods and stream data. This interleaving should
   generally be avoided unless necessary since it complicates client and
   server operation and imposes additional overhead. Interleaved binary
   data SHOULD only be used if RTSP is carried over TCP.

   Stream data such as RTP packets is encapsulated by an ASCII dollar
   sign (24 decimal), followed by a one-byte channel identifier,
   followed by the length of the encapsulated binary data as a binary,
   two-byte integer in network byte order. The stream data follows
   immediately afterwards, without a CRLF, but including the upper-layer
   protocol headers. Each $ block contains exactly one upper-layer
   protocol data unit, e.g., one RTP packet.

   The channel identifier is defined in the  Transport header with the
   interleaved parameter(Section 12.40).

   When the transport choice is RTP, RTCP messages are also interleaved
   by the server over the TCP connection. As a default, RTCP packets are
   sent on the first available channel higher than the RTP channel. The
   client MAY explicitly request RTCP packets on another channel. This
   is done by specifying two channels in the  interleaved parameter of
   the Transport header(Section 12.40).


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



H. Schulzrinne et. al.                                       [Page 39]


Internet Draft                    RTSP                 February 22, 2002


        configuration and transfer them onto UDP when possible.


     C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
           CSeq: 2
           Transport: RTP/AVP/TCP;interleaved=0-1

     S->C: RTSP/1.0 200 OK
           CSeq: 2
           Date: 05 Jun 1997 18:57:18 GMT
           Transport: RTP/AVP/TCP;interleaved=0-1
           Session: 12345678

     C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
           CSeq: 3
           Session: 12345678

     S->C: RTSP/1.0 200 OK
           CSeq: 3
           Session: 12345678
           Date: 05 Jun 1997 18:59:15 GMT
           RTP-Info: url=rtsp://foo.com/bar.file;
             seq=232433;rtptime=972948234

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



11 Status Code Definitions

   Where applicable, HTTP status [H10] codes are reused. Status codes
   that have the same meaning are not repeated here. See Table 1 for a
   listing of which status codes may be returned by which requests.

11.1 Success 2xx

11.1.1 250 Low on Storage Space

   The server returns this warning after receiving a  RECORD request
   that it may not be able to fulfill completely due to insufficient
   storage space. If possible, the server should use the Range header to
   indicate what time period it may still be able to record. Since other
   processes on the server may be consuming storage space
   simultaneously, a client should take this only as an estimate.

11.2 Redirection 3xx



H. Schulzrinne et. al.                                       [Page 40]


Internet Draft                    RTSP                 February 22, 2002


   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.

11.3 Client Error 4xx

11.4 400 Bad Request

   The request could not be understood by the server due to malformed
   syntax. The client SHOULD NOT repeat the request without
   modifications [H10.4.1]. If the request does not have a  CSeq header,
   the server MUST not include a  CSeq in the response.

11.4.1 405 Method Not Allowed

   The method specified in the request is not allowed for the resource
   identified by the request URI. The response MUST include an Allow
   header containing a list of valid methods for the requested resource.
   This status code is also to be used if a request attempts to use a
   method not indicated during SETUP, e.g., if a RECORD request is
   issued even though the  mode parameter in the  Transport header only
   specified PLAY.

11.4.2 451 Parameter Not Understood

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

11.4.3 452 Conference Not Found

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

11.4.4 453 Not Enough Bandwidth

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

11.4.5 454 Session Not Found

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

11.4.6 455 Method Not Valid in This State



H. Schulzrinne et. al.                                       [Page 41]


Internet Draft                    RTSP                 February 22, 2002


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

11.4.7 456 Header Field Not Valid for Resource

   The server could not act on a required request header. For example,
   if PLAY contains the  Range header field but the stream does not
   allow seeking.

11.4.8 457 Invalid Range

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

11.4.9 458 Parameter Is Read-Only

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

11.4.10 459 Aggregate Operation Not Allowed

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

11.4.11 460 Only Aggregate Operation Allowed

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

11.4.12 461 Unsupported Transport

   The  Transport field did not contain a supported transport
   specification.

11.4.13 462 Destination Unreachable

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

11.5 Server Error 5xx

11.5.1 551 Option not supported




H. Schulzrinne et. al.                                       [Page 42]


Internet Draft                    RTSP                 February 22, 2002


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

12 Header Field Definitions

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

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


12.1 Accept

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

        The "level" parameter for presentation descriptions is
        properly defined as part of the MIME type registration, not
        here.

   See [H14.1] for syntax.

   Example of use:

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



12.2 Accept-Encoding

   See [H14.3]

12.3 Accept-Language



H. Schulzrinne et. al.                                       [Page 43]


Internet Draft                    RTSP                 February 22, 2002


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

12.4 Accept-Ranges

12.5 Allow

   The  Allow entity-header field lists the methods supported by the
   resource identified by the request-URI. The purpose of this field is
   to strictly inform the recipient of valid methods associated with the
   resource. An  Allow header field must be present in a 405 (Method Not
   Allowed) response.

   Example of use:

     Allow: SETUP, PLAY, RECORD, SET_PARAMETER



12.6 Authorization

   See [H14.8]

12.7 Bandwidth

   The  Bandwidth request-header field describes the estimated bandwidth
   available to the client, expressed as a positive integer and measured
   in bits per second. The bandwidth available to the client may change
   during an RTSP session, e.g., due to modem retraining.


   Bandwidth _  "Bandwidth" ":" 1*DIGIT


   Example:

     Bandwidth: 4000



12.8 Blocksize

   The  Blocksize request-header field is sent from the client to the
   media server asking the server for a particular media packet size.
   This packet size does not include lower-layer headers such as IP,
   UDP, or RTP. The server is free to use a blocksize which is lower
   than the one requested. The server MAY truncate this packet size to



H. Schulzrinne et. al.                                       [Page 44]


Internet Draft                    RTSP                 February 22, 2002



       Header              type  support  methods
       ____________________________________________________________
       Accept               R     opt.    entity
       Accept-Encoding      R     opt.    entity
       Accept-Language      R     opt.    all
       Accept-Ranges        R     opt.    all
       Allow                e     opt.    all
       Authorization        R     opt.    all
       Bandwidth            R     opt.    all
       Blocksize            R     opt.    all but OPTIONS, TEARDOWN
       Cache-Control        g     opt.    SETUP
       Conference           R     opt.    SETUP
       Connection           g     req.    all
       Content-Base         e     opt.    entity
       Content-Encoding     e     req.    SET_PARAMETER
       Content-Encoding     e     req.    DESCRIBE, ANNOUNCE
       Content-Language     e     req.    DESCRIBE, ANNOUNCE
       Content-Length       e     req.    SET_PARAMETER, ANNOUNCE
       Content-Length       e     req.    entity
       Content-Location     e     opt.    entity
       Content-Type         e     req.    SET_PARAMETER, ANNOUNCE
       CSeq                 g     req.    all
       Date                 g     opt.    all
       Expires              e     opt.    DESCRIBE, ANNOUNCE
       From                 R     opt.    all
       If-Match             R     opt.    SETUP
       If-Modified-Since    R     opt.    DESCRIBE, SETUP
       Last-Modified        e     opt.    entity
       Location             r     opt.    201, 30x
       Proxy-Authenticate   r     req.    407
       Proxy-Require        R     req.    all
       Public               r     opt.    all
       Range                R     opt.    PLAY, PAUSE, RECORD
       Range                r     opt.    PLAY, PAUSE, RECORD
       Referer              R     opt.    all
       Require              R     req.    all
       Retry-After          r     opt.    all
       RTP-Info             r     req.    PLAY
       Scale                g     opt.    PLAY, RECORD
       Session              g     req.    all but OPTIONS
       Server               r     opt.    all
       Speed                g     opt.    PLAY
       Transport            g     req.    SETUP
       Unsupported          r     req.    all
       User-Agent           R     opt.    all
       Vary                 r     opt.    all
       Via                  g     opt.    all
       WWW-Authenticate     r     opt.    all


H. Schulzrinne et. al.                                       [Page 45]


Internet Draft                    RTSP                 February 22, 2002


   Table 3: Overview of RTSP header fields

   the closest multiple of the minimum, media-specific block size, or
   override it with the media-specific size if necessary. The block size
   MUST be a positive decimal number, measured in octets. The server
   only returns an error (416) if the value is syntactically invalid.


   Blocksize _  "Blocksize" ":" 1*DIGIT


12.9 Cache-Control

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

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

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


   Cache-Control                          =               "Cache-Control" ":" 1#cache-directive
   cache-directive                        =               cache-request-directive
   |                          cache-response-directive
   cache-request-directive                =               "no-cache"
   |                                 "max-stale"
   |                                 "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 ) ]




H. Schulzrinne et. al.                                       [Page 46]


Internet Draft                    RTSP                 February 22, 2002


        no-cache: Indicates that the media stream MUST NOT be cached
             anywhere. This allows an origin server to prevent caching
             even by caches that have been configured to return stale
             responses to client requests.

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

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

        no-transform: An intermediate cache (proxy) may find it useful
             to convert the media type of a certain stream. A proxy
             might, for example, convert between video formats to save
             cache space or to reduce the amount of traffic on a slow
             link. Serious operational problems may occur, however, when
             these transformations have been applied to streams intended
             for certain kinds of applications. For example,
             applications for medical imaging, scientific data analysis
             and those using end-to-end authentication all depend on
             receiving a stream that is bit-for-bit identical to the
             original entity-body. Therefore, if a response includes the
             no-transform directive, an intermediate cache or proxy MUST
             NOT change the encoding of the stream. Unlike HTTP, RTSP
             does not provide for partial transformation at this point,
             e.g., allowing translation into a different language.

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

        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.



H. Schulzrinne et. al.                                       [Page 47]


Internet Draft                    RTSP                 February 22, 2002


        min-fresh: Indicates that the client is willing to accept a
             media stream whose freshness lifetime is no less than its
             current age plus the specified time in seconds. That is,
             the client wants a response that will still be fresh for at
             least the specified number of seconds.

        must-revalidate: When the  must-revalidate directive is present
             in a SETUP response received by a cache, that cache MUST
             NOT use the entry after it becomes stale to respond to a
             subsequent request without first revalidating it with the
             origin server.  That is, the cache must do an end-to-end
             revalidation every time, if, based solely on the origin
             server's  Expires, the cached response is stale.)

12.10 Conference

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


   Conference _  "Conference" ":" conference-id


   Example:

     Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr



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

12.11 Connection

   See [H14.10]

12.12 Content-Base

   See [H14.11]

12.13 Content-Encoding

   See [H14.12]

12.14 Content-Language

   See [H14.13]



H. Schulzrinne et. al.                                       [Page 48]


Internet Draft                    RTSP                 February 22, 2002


12.15 Content-Length

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

12.16 Content-Location

   See [H14.15]

12.17 Content-Type

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

12.18 CSeq

   The  CSeq general-header field specifies the sequence number for an
   RTSP request-response pair. This field MUST be present in all
   requests and responses. For every RTSP request containing the given
   sequence number, the corresponding response will have the same
   number. Any retransmitted request must contain the same sequence
   number as the original (i.e. the sequence number is not incremented
   for retransmissions of the same request).


   CSeq _  "Cseq" ":" 1*DIGIT


12.19 Date

   See [H14.19].

12.20 Expires

   The  Expires entity-header field gives a date and time after which
   the description or media-stream should be considered stale. The
   interpretation depends on the method:

        DESCRIBE response: The  Expires header indicates a date and time
             after which the description should be considered stale.

   A stale cache entry may not normally be returned by a cache (either a
   proxy cache or an user agent cache) unless it is first validated with



H. Schulzrinne et. al.                                       [Page 49]


Internet Draft                    RTSP                 February 22, 2002


   the origin server (or with an intermediate cache that has a fresh
   copy of the entity). See section 13 for further discussion of the
   expiration model.

   The presence of an  Expires field does not imply that the original
   resource will change or cease to exist at, before, or after that
   time.

   The format is an absolute date and time as defined by  HTTP-date in
   [H3.3]; it MUST be in  RFC1123-date format:


   Expires _  "Expires" ":" HTTP-date


   An example of its use is


     Expires: Thu, 01 Dec 1994 16:00:00 GMT



   RTSP/1.0 clients and caches MUST treat other invalid date formats,
   especially including the value "0", as having occurred in the past
   (i.e., already expired).

   To mark a response as "already expired," an origin server should use
   an  Expires date that is equal to the  Date header value. To mark a
   response as "never expires," an origin server should use an  Expires
   date approximately one year from the time the response is sent.
   RTSP/1.0 servers should not send  Expires dates more than one year in
   the future.

   The presence of an  Expires header field with a date value of some
   time in the future on a media stream that otherwise would by default
   be non-cacheable indicates that the media stream is cacheable, unless
   indicated otherwise by a  Cache-Control header field (Section 12.9).

12.21 From

   See [H14.22].

12.22 Host

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

12.23 If-Match



H. Schulzrinne et. al.                                       [Page 50]


Internet Draft                    RTSP                 February 22, 2002


   See [H14.25].

   The  If-Match request-header field is especially useful for ensuring
   the integrity of the presentation description, in both the case where
   it is fetched via means external to RTSP (such as HTTP), or in the
   case where the server implementation is guaranteeing the integrity of
   the description between the time of the DESCRIBE message and the
   SETUP message.

   The identifier is an opaque identifier, and thus is not specific to
   any particular session description language.

12.24 If-Modified-Since

   The  If-Modified-Since request-header field is used with the DESCRIBE
   and SETUP methods to make them conditional. If the requested variant
   has not been modified since the time specified in this field, a
   description will not be returned from the server (DESCRIBE) or a
   stream will not be set up (SETUP). Instead, a 304 (Not Modified)
   response will be returned without any message-body.


   If-Modified-Since _  "If-Modified-Since" ":" HTTP-date


   An example of the field is:


     If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT



12.25 Last-Modified

   The  Last-Modified entity-header field indicates the date and time at
   which the origin server believes the presentation description or
   media stream was last modified. See [H14.29]. For the methods
   DESCRIBE or ANNOUNCE, the header field indicates the last
   modification date and time of the description, for SETUP that of the
   media stream.

12.26 Location

   See [H14.30].

12.27 Proxy-Authenticate

   See [H14.33].



H. Schulzrinne et. al.                                       [Page 51]


Internet Draft                    RTSP                 February 22, 2002


12.28 Proxy-Require

   The  Proxy-Require request-header field is used to indicate proxy-
   sensitive features that MUST be supported by the proxy. Any Proxy-
   Require header features that are not supported by the proxy MUST be
   negatively acknowledged by the proxy to the client if not supported.
   Servers should treat this field identically to the Require field.

   See Section 12.33 for more details on the mechanics of this message
   and a usage example.

12.29 Public

   See [H14.35].

12.30  Range

   The  Range request and response header field specifies a range of
   time. The range can be specified in a number of units. This
   specification defines the  smpte (Section 3.5), npt (Section 3.6),
   and  clock (Section 3.7) range units. Within RTSP, byte ranges
   [H14.36.1] are not meaningful and MUST NOT be used. The header may
   also contain a  time parameter in UTC, specifying the time at which
   the operation is to be made effective. Servers supporting the Range
   header MUST understand the NPT range format and SHOULD understand the
   SMPTE range format. The  Range response header indicates what range
   of time is actually being played or recorded. If the  Range header is
   given in a time format that is not understood, the recipient should
   return 501 (Not Implemented).

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


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


   Example:

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



H. Schulzrinne et. al.                                       [Page 52]


Internet Draft                    RTSP                 February 22, 2002


        The notation is similar to that used for the HTTP/1.1 [2]
        byte-range header. It allows clients to select an excerpt
        from the media object, and to play from a given point to
        the end as well as from the current location to a given
        point. The start of playback can be scheduled for any time
        in the future, although a server may refuse to keep server
        resources for extended idle periods.

12.31 Referer

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

12.32 Retry-After

   See [H14.38].

12.33 Require

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


        This is to make sure that the client-server interaction
        will proceed without delay when all options are understood
        by both sides, and only slow down if options are not
        understood (as in the case above).  For a well-matched
        client-server pair, the interaction proceeds quickly,
        saving a round-trip often required by negotiation
        mechanisms. In addition, it also removes state ambiguity
        when the client requires features that the server does not
        understand.


   Require _  "Require" ":" 1#option-tag


   Example:

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

   S->C:   RTSP/1.0 551 Option not supported
           CSeq: 302



H. Schulzrinne et. al.                                       [Page 53]


Internet Draft                    RTSP                 February 22, 2002


           Unsupported: funky-feature

   C->S:   SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 303

   S->C:   RTSP/1.0 200 OK
           CSeq: 303



   In this example, "funky-feature" is the feature tag which indicates
   to the client that the fictional  Funky-Parameter field is required.
   The relationship between "funky-feature" and  Funky-Parameter is not
   communicated via the RTSP exchange, since that relationship is an
   immutable property of "funky-feature" and thus should not be
   transmitted with every exchange.

   Proxies and other intermediary devices SHOULD ignore features that
   are not understood in this field. If a particular extension requires
   that intermediate devices support it, the extension should be tagged
   in the Proxy-Require field instead (see Section 12.28).

12.34 RTP-Info

   The  RTP-Info response-header field is used to set RTP-specific
   parameters in the PLAY response.

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

        seq: Indicates the sequence number of the first packet of the
             stream. This allows clients to gracefully deal with packets
             when seeking. The client uses this value to differentiate
             packets that originated before the seek from packets that
             originated after the seek.

        rtptime: Indicates the RTP timestamp corresponding to the time
             value in the  Range response header. (Note: For aggregate
             control, a particular stream may not actually generate a
             packet for the  Range time value returned or implied. Thus,
             there is no guarantee that the packet with the sequence
             number indicated by  seq actually has the timestamp
             indicated by rtptime.) The client uses this value to
             calculate the mapping of RTP time to NPT.


             A mapping from RTP timestamps to NTP timestamps (wall
             clock) is available via RTCP. However, this



H. Schulzrinne et. al.                                       [Page 54]


Internet Draft                    RTSP                 February 22, 2002


             information is not sufficient to generate a mapping
             from RTP timestamps to NPT. Furthermore, in order to
             ensure that this information is available at the
             necessary time (immediately at startup or after a
             seek), and that it is delivered reliably, this mapping
             is placed in the RTSP control channel.

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

   Syntax:

   RTP-Info       ______________________________  "RTP-Info" ":" 1#rtsp-info-spec
   rtsp-info-spec ______________________________  stream-url 1*parameter
   stream-url     ______________________________  quoted-url | unquoted-url
   unquoted-url   ______________________________  "url" "=" safe-url
   |               ";" "mode" = <"> 1#Method <">
   quoted-url     ______________________________  "url" "=" <"> needquote-url <">
   safe-url       ______________________________  url
   needquote-url  ______________________________  url
   url            ______________________________  ( absoluteURI | relativeURI )
   parameter      ______________________________  ";" "seq" "=" 1*DIGIT
   |                 ";" "rtptime" "=" 1*DIGIT


   Additional constraint: safe-url MUST NOT contain the semicolon (";")
   or comma (",") characters. The quoted-url form SHOULD only be used
   when a URL does not meet the safe-url constraint, in order to ensure
   compatibility with implementations conformant to RFC 2326 [25].

   absoluteURI and relativeURI are defined in RFC 2396 [26].

   Example:

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



12.35 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



H. Schulzrinne et. al.                                       [Page 55]


Internet Draft                    RTSP                 February 22, 2002


   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



12.36 Speed

   The  Speed request-header field requests the server to deliver data
   to the client at a particular speed, contingent on the server's
   ability and desire to serve the media stream at the given speed.
   Implementation by the server is OPTIONAL. The default is the bit rate
   of the stream.

   The parameter value is expressed as a decimal ratio, e.g., a value of
   2.0 indicates that data is to be delivered twice as fast as normal. A
   speed of zero is invalid. If the request contains a  Range parameter,
   the new speed value will take effect at that time.


   Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]


   Example:




H. Schulzrinne et. al.                                       [Page 56]


Internet Draft                    RTSP                 February 22, 2002


     Speed: 2.5



   Use of this field changes the bandwidth used for data delivery. It is
   meant for use in specific circumstances where preview of the
   presentation at a higher or lower rate is necessary. Implementors
   should keep in mind that bandwidth for the session may be negotiated
   beforehand (by means other than RTSP), and therefore re-negotiation
   may be necessary. When data is delivered over UDP, it is highly
   recommended that means such as RTCP be used to track packet loss
   rates.

12.37 Server

   See [H14.39]

12.38 Session

   The  Session request-header and response-header field identifies an
   RTSP session started by the media server in a SETUP response and
   concluded by  TEARDOWN on the presentation URL. The session
   identifier is chosen by the media server (see Section 3.4) and MUST
   be returned in the SETUP response. Once a client receives a  Session
   identifier, it MUST return it for any request related to that
   session.


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


   The  timeout parameter is only allowed in a response header.  The
   server uses it to indicate to the client how long the server is
   prepared to wait between RTSP commands before closing the session due
   to lack of activity (see Section A). The timeout is measured in
   seconds, with a default of 60 seconds (1 minute).

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

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



H. Schulzrinne et. al.                                       [Page 57]


Internet Draft                    RTSP                 February 22, 2002


        client.

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

12.39 Timestamp

   The  Timestamp general-header field describes when the client sent
   the request to the server. The value of the timestamp is of
   significance only to the client and may use any timescale. The server
   MUST echo the exact same value and MAY, if it has accurate
   information about this, add a floating point number indicating the
   number of seconds that has elapsed since it has received the request.
   The timestamp is used by the client to compute the round-trip time to
   the server so that it can adjust the timeout value for
   retransmissions.


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


12.40 Transport

   The  Transport request-header field indicates which transport
   protocol is to be used and configures its parameters such as
   destination address, compression, multicast time-to-live and
   destination port for a single stream. It sets those values not
   already determined by a presentation description.

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

   The  Transport header field MAY also be used to change certain
   transport parameters. A server MAY refuse to change parameters of an
   existing stream.

   The server MAY return a  Transport response-header field in the
   response to indicate the values actually chosen.

   A  Transport request header field may contain a list of transport
   options acceptable to the client, in the form of multiple transport-
   spec entries. In that case, the server MUST return a single option (
   transport-spec) which was actually chosen.

   A  transport-spec transport option may only contain one of any given
   parameter within it. Parameters may be given in any order.
   Additionally, it may only contain the  unicast or multicast transport



H. Schulzrinne et. al.                                       [Page 58]


Internet Draft                    RTSP                 February 22, 2002


   parameter.


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

   The syntax for the transport specifier is

   transport
   /
   profile
   /
   lower-transport


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

   Below are the configuration parameters associated with transport:

   General parameters:

        unicast |  multicast: This parameter is a mutually exclusive
             indication of whether unicast or multicast delivery will be
             attempted. One of the two values MUST be specified. Clients
             that are capable of handling both unicast and multicast
             transmission MUST indicate such capability by including two
             full transport-specs with separate parameters for each.

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

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




H. Schulzrinne et. al.                                       [Page 59]


Internet Draft                    RTSP                 February 22, 2002


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

        layers: The number of multicast layers to be used for this media
             stream. The layers are sent to consecutive addresses
             starting at the  destination address.

        mode: The  mode parameter indicates the methods to be supported
             for this session. Valid values are PLAY and RECORD. If not
             provided, the default is PLAY.

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

        interleaved: The  interleaved parameter implies mixing the media
             stream with the control stream in whatever protocol is
             being used by the control stream, using the mechanism
             defined in Section 10.12. The argument provides the channel
             number to be used in the $ statement. This parameter may be
             specified as a range, e.g., interleaved=4-5 in cases where
             the transport choice for the media stream requires it.


             This allows RTP/RTCP to be handled similarly to the
             way that it is done with UDP, i.e., one channel for
             RTP and the other for RTCP.

   Multicast-specific:

        ttl: multicast time-to-live.

   RTP-specific:

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

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



H. Schulzrinne et. al.                                       [Page 60]


Internet Draft                    RTSP                 February 22, 2002


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

        ssrc: The  ssrc parameter indicates the RTP SSRC [27] 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, and is expressed as an
             eight digit hexidecimal value.


   Transport                               ______________________________________________  "Transport" ":" 1#transport-spec
   transport-spec                                                 =                        transport-id *parameter
   transport-id                                                   =                        transport-protocol "/" profile ["/" lower-transport]
   ; no LWS is allowed inside transport-id
   transport-protocol                                             =                        "RTP" | token
   profile                                                        =                        "AVP" | token
   lower-transport                                                =                        "TCP" | "UDP" | token
   parameter                                                      =                        ";" ( "unicast" | "multicast" )
   |                                                ";" "source" [ "=" address ]
   |                                              ";" "destination" [ "=" address ]
   |                                        ";" "interleaved" "=" channel [ "-" channel ]
   |                                                        ";" "append"
   |                                                      ";" "ttl" "=" ttl
   |                                                  ";" "layers" "=" 1*DIGIT
   |                                              ";" "port" "=" port [ "-" port ]
   |                                           ";" "client_port" "=" port [ "-" port ]
   |                                           ";" "server_port" "=" port [ "-" port ]
   |                                                  ";" "source" "=" address
   |                                                     ";" "ssrc" "=" ssrc
   |                                                  ";" "mode" "=" mode-spec
   ttl                                                            =                        1*3(DIGIT)
   port                                                           =                        1*5(DIGIT)
   ssrc                                                           =                        8*8(HEX)
   channel                                                        =                        1*3(DIGIT)
   address                                                        =                        host
   mode-spec                                                      =                        <"> 1#mode <"> | mode
   mode                                                           =                        "PLAY" | "RECORD" | token


   Below is a usage example, showing a client advertising the capability
   to handle multicast or unicast, preferring multicast. Since this is a
   unicast-only stream, the server responds with the proper transport
   parameters for unicast.





H. Schulzrinne et. al.                                       [Page 61]


Internet Draft                    RTSP                 February 22, 2002


     C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
           CSeq: 302
           Transport: RTP/AVP;multicast;mode="PLAY",
               RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"

     S->C: RTSP/1.0 200 OK
           CSeq: 302
           Date: 23 Jan 1997 15:35:06 GMT
           Session: 47112344
           Transport: RTP/AVP;unicast;client_port=3456-3457;
               server_port=6256-6257;mode="PLAY"



12.41 Unsupported

   The  Unsupported response-header field lists the features not
   supported by the server. In the case where the feature was specified
   via the  Proxy-Require field (Section 12.33), if there is a proxy on
   the path between the client and the server, the proxy MUST insert a
   response message with a status code of 551 (Option Not Supported).

   See Section 12.33 for a usage example.


   Unsupported _  "Unsupported" ":" 1#option-tag


12.42 User-Agent

   See [H14.42]

12.43 Vary

   See [H14.43]

12.44 Via

   See [H14.44].

12.45 WWW-Authenticate

   See [H14.46].

13 Caching

   In HTTP, response-request pairs are cached. RTSP differs
   significantly in that respect. Responses are not cacheable, with the



H. Schulzrinne et. al.                                       [Page 62]


Internet Draft                    RTSP                 February 22, 2002


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

   On receiving a  SETUP or  PLAY request, a proxy ascertains whether it
   has an up-to-date copy of the continuous media content and its
   description. It can determine whether the copy is up-to-date by
   issuing a  SETUP or  DESCRIBE request, respectively, and comparing
   the  Last-Modified header with that of the cached copy. If the copy
   is not up-to-date, it modifies the SETUP transport parameters as
   appropriate and forwards the request to the origin server. Subsequent
   control commands such as PLAY or  PAUSE then pass the proxy
   unmodified. The proxy delivers the continuous media data to the
   client, while possibly making a local copy for later reuse. The exact
   behavior allowed to the cache is given by the cache-response
   directives described in Section 12.9. A cache MUST answer any
   DESCRIBE requests if it is currently serving the stream to the
   requestor, as it is possible that low-level details of the stream
   description may have changed on the origin-server.

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

   To the client, an RTSP proxy cache appears like a regular media
   server, to the media origin server like a client. Just as an HTTP
   cache has to store the content type, content language, and so on for
   the objects it caches, a media cache has to store the presentation
   description.  Typically, a cache eliminates all transport-references
   (that is, multicast information) from the presentation description,
   since these are independent of the data delivery from the cache to
   the client.  Information on the encodings remains the same. If the
   cache is able to translate the cached media data, it would create a
   new presentation description with all the encoding possibilities it
   can offer.

14 Examples

   The following examples refer to stream description formats that are
   not standards, such as RTSL. The following examples are not to be
   used as a reference for those formats.

14.1 Media on Demand (Unicast)



H. Schulzrinne et. al.                                       [Page 63]


Internet Draft                    RTSP                 February 22, 2002


   Client C requests a movie from media servers A ( audio.example.com )
   and V ( video.example.com ). The media description is stored on a web
   server W. The media description contains descriptions of the
   presentation and all its streams, including the codecs that are
   available, dynamic RTP payload types, the protocol stack, and content
   information such as language or copyright restrictions. It may also
   give an indication about the timeline of the movie.

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


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

   W->C: HTTP/1.0 200 OK
         Content-Type: application/sdp

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

   C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 1
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057

   A->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 12345678
         Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
                    server_port=5000-5001

   C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 1
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059

   V->C: RTSP/1.0 200 OK
         CSeq: 1
         Session: 23456789
         Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
                    server_port=5002-5003

   C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0



H. Schulzrinne et. al.                                       [Page 64]


Internet Draft                    RTSP                 February 22, 2002


         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-

   V->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 23456789
         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://video.example.com/twister/video;
           seq=12312232;rtptime=78712811

   C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-

   A->C: RTSP/1.0 200 OK
         CSeq: 2
         Session: 12345678
         Range: smpte=0:10:00-0:20:00
         RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
           seq=876655;rtptime=1032181

   C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
         CSeq: 3
         Session: 12345678

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

   C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
         CSeq: 3
         Session: 23456789

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



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

14.2 Streaming of a Container file

   For purposes of this example, a container file is a storage entity in



H. Schulzrinne et. al.                                       [Page 65]


Internet Draft                    RTSP                 February 22, 2002


   which multiple continuous media types pertaining to the same end-user
   presentation are present. In effect, the container file represents an
   RTSP presentation, with each of its components being RTSP streams.
   Container files are a widely used means to store such presentations.
   While the components are transported as independent streams, it is
   desirable to maintain a common context for those streams at the
   server end.


        This enables the server to keep a single storage handle
        open easily. It also allows treating all the streams
        equally in case of any prioritization of streams by the
        server.

   It is also possible that the presentation author may wish to prevent
   selective retrieval of the streams by the client in order to preserve
   the artistic effect of the combined media presentation. Similarly, in
   such a tightly bound presentation, it is desirable to be able to
   control all the streams via a single control message using an
   aggregate URL.

   The following is an example of using a single RTSP session to control
   multiple streams. It also illustrates the use of aggregate URLs.

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


   C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
         CSeq: 1

   M->C: RTSP/1.0 200 OK
         CSeq: 1
         Content-Type: application/sdp
         Content-Length: 164

         v=0
         o=- 2890844256 2890842807 IN IP4 172.16.2.93
         s=RTSP Session
         i=An Example of RTSP Session Usage
         a=control:rtsp://foo/twister
         t=0 0
         m=audio 0 RTP/AVP 0
         a=control:rtsp://foo/twister/audio
         m=video 0 RTP/AVP 26
         a=control:rtsp://foo/twister/video




H. Schulzrinne et. al.                                       [Page 66]


Internet Draft                    RTSP                 February 22, 2002


   C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;unicast;client_port=8000-8001

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;unicast;client_port=8000-8001;
                    server_port=9000-9001
         Session: 12345678

   C->M: SETUP rtsp://foo/twister/video RTSP/1.0
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003
         Session: 12345678

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Transport: RTP/AVP;unicast;client_port=8002-8003;
                    server_port=9004-9005
         Session: 12345678

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

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

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

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

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

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

   C->M: SETUP rtsp://foo/twister RTSP/1.0



H. Schulzrinne et. al.                                       [Page 67]


Internet Draft                    RTSP                 February 22, 2002


         CSeq: 7
         Transport: RTP/AVP;unicast;client_port=10000

   M->C: RTSP/1.0 459 Aggregate operation not allowed
         CSeq: 7




   In the first instance of failure, the client tries to pause one
   stream (in this case video) of the presentation. This is disallowed
   for that presentation by the server. In the second instance, the
   aggregate URL may not be used for  SETUP and one control message is
   required per stream to set up transport parameters.

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

14.3 Single Stream Container Files

   Some RTSP servers may treat all files as though they are "container
   files", yet other servers may not support such a concept. Because of
   this, clients SHOULD use the rules set forth in the session
   description for request URLs, rather than assuming that a consistent
   URL may always be used throughout. Here's an example of how a multi-
   stream server might expect a single-stream file to be served:


       C->S  DESCRIBE rtsp://foo.com/test.wav RTSP/1.0
             Accept: application/x-rtsp-mh, application/sdp
             CSeq: 1

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

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

       C->S  SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
             Transport: RTP/AVP/UDP;unicast;



H. Schulzrinne et. al.                                       [Page 68]


Internet Draft                    RTSP                 February 22, 2002


                        client_port=6970-6971;mode="PLAY"
             CSeq: 2

       S->C  RTSP/1.0 200 OK
             Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
                        server_port=6970-6971;mode="PLAY"
             CSeq: 2
             Session: 2034820394

       C->S  PLAY rtsp://foo.com/test.wav RTSP/1.0
             CSeq: 3
             Session: 2034820394

       S->C  RTSP/1.0 200 OK
             CSeq: 3
             Session: 2034820394
             RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
               seq=981888;rtptime=3781123



   Note the different URL in the  SETUP command, and then the switch
   back to the aggregate URL in the  PLAY command. This makes complete
   sense when there are multiple streams with aggregate control, but is
   less than intuitive in the special case where the number of streams
   is one.

   In this special case, it is recommended that servers be forgiving of
   implementations that send:


       C->S  PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
             CSeq: 3



   In the worst case, servers should send back:


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



   One would also hope that server implementations are also forgiving of
   the following:





H. Schulzrinne et. al.                                       [Page 69]


Internet Draft                    RTSP                 February 22, 2002


       C->S  SETUP rtsp://foo.com/test.wav RTSP/1.0
             Transport: rtp/avp/udp;client_port=6970-6971;mode="PLAY"
             CSeq: 2



   Since there is only a single stream in this file, it's not ambiguous
   what this means.

14.4 Live Media Presentation Using Multicast

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


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

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

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

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

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

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

   C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 2
         Transport: RTP/AVP;multicast

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;multicast;destination=224.2.0.1;



H. Schulzrinne et. al.                                       [Page 70]


Internet Draft                    RTSP                 February 22, 2002


                    port=3456-3457;ttl=16
         Session: 0456804596

   C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
         CSeq: 3
         Session: 0456804596

   M->C: RTSP/1.0 200 OK
         CSeq: 3
         Session: 0456804596



14.5 Playing media into an existing session

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


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

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

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

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

   M->C: RTSP/1.0 200 OK
         CSeq: 2
         Transport: RTP/AVP;multicast;destination=225.219.201.15;
                    port=7000-7001;ttl=127
         Session: 91389234234



H. Schulzrinne et. al.                                       [Page 71]


Internet Draft                    RTSP                 February 22, 2002


         Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr

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

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



14.6 Recording

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



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

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

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

   C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
         CSeq: 91
         Transport: RTP/AVP;multicast;destination=224.0.1.11;



H. Schulzrinne et. al.                                       [Page 72]


Internet Draft                    RTSP                 February 22, 2002


                    port=21010-21011;mode=record;ttl=127

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

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

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

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

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



15 Syntax

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

15.1 Base Syntax


   OCTET                                                ____________________________  <any 8-bit sequence of data>
   CHAR                                                 ____________________________  <any US-ASCII character (octets 0 - 127)>
   UPALPHA                                              ____________________________  <any US-ASCII uppercase letter "A".."Z">
   LOALPHA                                              ____________________________  <any US-ASCII lowercase letter "a".."z">
   ALPHA                                                ____________________________  UPALPHA | LOALPHA
   DIGIT                                                ____________________________  <any US-ASCII digit "0".."9">
   CTL                                                  ____________________________  <any US-ASCII control character
   (octets 0 - 31) and DEL (127)>
   CR                                                   ____________________________  <US-ASCII CR, carriage return (13)>
   LF                                                   ____________________________  <US-ASCII LF, linefeed (10)>
   SP                                                   ____________________________  <US-ASCII SP, space (32)>



H. Schulzrinne et. al.                                       [Page 73]


Internet Draft                    RTSP                 February 22, 2002


   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                                            ____________________________  "(" | ")" | "<" | ">" | "@"
   |                                                         "," | ";" | ":" | "
   \&\h'|\n(40u'\h'|\n(41u'\h'|\n(42u'
   " | <">
   |                                                     "/" | "[" | "]" | "?" | "="
   |                                                         "{" | "}" | SP | HT
   token                                                ____________________________  1*<any CHAR except CTLs or tspecials>
   quoted-string                                        ____________________________  ( <"> *(qdtext) <"> )
   qdtext                                               ____________________________  <any TEXT except <">>
   quoted-pair                                          ____________________________  "
   \&\h'|\n(40u'\h'|\n(41u'\h'|\n(42u'
   " CHAR
   message-header                                       ____________________________  field-name ":" [ field-value ] CRLF
   field-name                                           ____________________________  token
   field-value                                          ____________________________  *( field-content | LWS )
   field-content                                        ____________________________  <the OCTETs making up the field-value and
   consisting
   of either *TEXT or combinations of token, tspecials,
   and quoted-string>
   safe                                                 ____________________________  "$" | "-" | "_" | "." | "+"
   extra                                                ____________________________  "!" | "*" | "'" | "(" | ")" | ","
   hex                                                  ____________________________  DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
   "a" | "b" | "c" | "d" | "e" | "f"
   escape                                               ____________________________  "%" hex hex
   reserved                                             ____________________________  ";" | "/" | "?" | ":" | "@" | "&" | "="
   unreserved                                           ____________________________  alpha | digit | safe | extra
   xchar                                                ____________________________  unreserved | reserved | escape


16 Security Considerations

   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security considerations outlined in [H15]
   apply.  Specifically, please note the following:

        Authentication Mechanisms: RTSP and HTTP share common
             authentication schemes, and thus should follow the same
             prescriptions with regards to authentication. See [H15.1]
             for client authentication issues, and [H15.2] for issues
             regarding support for multiple authentication mechanisms.

        Abuse of Server Log Information: RTSP and HTTP servers will
             presumably have similar logging mechanisms, and thus should



H. Schulzrinne et. al.                                       [Page 74]


Internet Draft                    RTSP                 February 22, 2002


             be equally guarded in protecting the contents of those
             logs, thus protecting the privacy of the users of the
             servers. See [H15.3] for HTTP server recommendations
             regarding server logs.

        Transfer of Sensitive Information: There is no reason to believe
             that information transferred via RTSP may be any less
             sensitive than that normally transmitted via HTTP.
             Therefore, all of the precautions regarding the protection
             of data privacy and user privacy apply to implementors of
             RTSP clients, servers, and proxies. See [H15.4] for further
             details.

        Attacks Based On File and Path Names: Though RTSP URLs are
             opaque handles that do not necessarily have file system
             semantics, it is anticipated that many implementations will
             translate portions of the request URLs directly to file
             system calls. In such cases, file systems SHOULD follow the
             precautions outlined in [H15.5], such as checking for ".."
             in path components.

        Personal Information: RTSP clients are often privy to the same
             information that HTTP clients are (user name, location,
             etc.)  and thus should be equally. See [H15.6] for further
             recommendations.

        Privacy Issues Connected to Accept Headers: Since may of the
             same "Accept" headers exist in RTSP as in HTTP, the same
             caveats outlined in [H15.7] with regards to their use
             should be followed.

        DNS Spoofing: Presumably, given the longer connection times
             typically associated to RTSP sessions relative to HTTP
             sessions, RTSP client DNS optimizations should be less
             prevalent. Nonetheless, the recommendations provided in
             [H15.8] are still relevant to any implementation which
             attempts to rely on a DNS-to-IP mapping to hold beyond a
             single use of the mapping.

        Location Headers and Spoofing: If a single server supports
             multiple organizations that do not trust one another, then
             it must check the values of  Location and  Content-Location
             header fields in responses that are generated under control
             of said organizations to make sure that they do not attempt
             to invalidate resources over which they have no authority.
             ([H15.9])

   In addition to the recommendations in the current HTTP specification



H. Schulzrinne et. al.                                       [Page 75]


Internet Draft                    RTSP                 February 22, 2002


   (RFC 2068 [2], as of this writing), future HTTP specifications may
   provide additional guidance on security issues.

   The following are added considerations for RTSP implementations.

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

             The attacker may initiate traffic flows to one or more IP
             addresses by specifying them as the destination in  SETUP
             requests. While the attacker's IP address may be known in
             this case, this is not always useful in prevention of more
             attacks or ascertaining the attackers identity. Thus, an
             RTSP server SHOULD only allow client-specified destinations
             for RTSP-initiated traffic flows if the server has verified
             the client's identity, either against a database of known
             users using RTSP authentication mechanisms (preferably
             digest authentication or stronger), or other secure means.

        Session hijacking: Since there is no relation between a
             transport layer connection and an RTSP session, it is
             possible for a malicious client to issue requests with
             random session identifiers which would affect unsuspecting
             clients. The server SHOULD use a large, random and non-
             sequential session identifier to minimize the possibility
             of this kind of attack.

        Authentication: Servers SHOULD implement both basic and digest
             [8] authentication. In environments requiring tighter
             security for the control messages, transport layer
             mechanisms such as TLS (RFC 2246 [7]) SHOULD be used.

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

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




H. Schulzrinne et. al.                                       [Page 76]


Internet Draft                    RTSP                 February 22, 2002


A RTSP Protocol State Machines

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

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


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



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


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

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

A.1 Client State Machine

   The client can assume the following states:

        Init :  SETUP has been sent, waiting for reply.

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

        Playing :  PLAY reply received

        Recording :  RECORD reply received

   In general, the client changes state on receipt of replies to
   requests.  Note that some requests are effective at a future time or
   position (such as a  PAUSE), and state also changes accordingly. If



H. Schulzrinne et. al.                                       [Page 77]


Internet Draft                    RTSP                 February 22, 2002


   no explicit SETUP is required for the object (for example, it is
   available via a multicast group), state begins at Ready are only two
   states, Ready and Playing The client also changes state from
   Playing/Recording to Ready when the end of the requested range is
   reached.

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


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




H. Schulzrinne et. al.                                       [Page 78]


Internet Draft                    RTSP                 February 22, 2002


A.2 Server State Machine

   The server can assume the following states:

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

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

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

        Recording : The server is recording media data.

   In general, the server changes state on receiving requests. If the
   server is in state Playing or Recording and in unicast mode, it MAY
   revert to Init and tear down the RTSP session if it has not received
   "wellness" information, such as RTCP reports or RTSP commands, from
   the client for a defined interval, with a default of one minute. The
   server can declare another timeout value in the Session response
   header (Section 12.38). If the server is in state Ready , it MAY
   revert to Init if it does not receive an RTSP request for an interval
   of more than one minute.  Note that some requests (such as PAUSE) may
   be effective at a future time or position, and server state changes
   at the appropriate time. The server reverts from state Playing or
   Recording to state Ready at the end of the range requested by the
   client.

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

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

   The "next state" column indicates the state assumed after sending a
   success response (2xx). If a request results in a status code of 3xx,
   the state becomes Init change.


   state      message received  next state
   _______________________________________
   Init
   SETUP
   Ready
   TEARDOWN
   Init
   Ready
   PLAY


H. Schulzrinne et. al.                                       [Page 79]


Internet Draft                    RTSP                 February 22, 2002


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


B Interaction with RTP

   RTSP allows media clients to control selected, non-contiguous
   sections of media presentations, rendering those streams with an RTP
   media layer[27]. The media layer rendering the RTP stream should not
   be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
   timestamps MUST be continuous and monotonic across jumps of NPT.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero.  First we play NPT 10 through 15, then skip ahead and play NPT
   18 through 20. The first segment is presented as RTP packets with
   sequence numbers 0 through 49 and timestamp 0 through 39,200. The
   second segment consists of RTP packets with sequence number 50
   through 69, with timestamps 40,000 through 55,200.


        We cannot assume that the RTSP client can communicate with
        the RTP media agent, as the two may be independent
        processes. If the RTP timestamp shows the same gap as the
        NPT, the media agent will assume that there is a pause in



H. Schulzrinne et. al.                                       [Page 80]


Internet Draft                    RTSP                 February 22, 2002


        the presentation. If the jump in NPT is large enough, the
        RTP timestamp may roll over and the media agent may believe
        later packets to be duplicates of packets just played out.

   For certain datatypes, tight integration between the RTSP layer and
   the RTP layer will be necessary. This by no means precludes the above
   restriction. Combined RTSP/RTP media clients should use the RTP-Info
   field to determine whether incoming RTP packets were sent before or
   after a seek.

   For continuous audio, the server SHOULD set the RTP marker bit at the
   beginning of serving a new  PLAY request. This allows the client to
   perform playout delay adaptation.

   For scaling (see Section 12.35), RTP timestamps should correspond to
   the playback timing. For example, when playing video recorded at 30
   frames/second at a scale of two and speed (Section 12.36) of one, the
   server would drop every second frame to maintain and deliver video
   packets with the normal timestamp spacing of 3,000 per frame, but NPT
   would increase by 1/15 second for each video frame.

   The client can maintain a correct display of NPT by noting the RTP
   timestamp value of the first packet arriving after repositioning. The
   sequence parameter of the  RTP-Info (Section 12.34) header provides
   the first sequence number of the next segment.

C Use of SDP for RTSP Session Descriptions

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

        aggregate control: A presentation composed of streams from one
             or more servers that are available for aggregate control.
             Such a description is typically retrieved by HTTP or other
             non-RTSP means.  However, they may be received with
             ANNOUNCE methods.

        non-aggregate control: A presentation composed of multiple
             streams from a single server that are not available for
             aggregate control.  Such a description is typically
             returned in reply to a DESCRIBE request on a URL, or
             received in an ANNOUNCE method.

   This appendix describes how an SDP file, retrieved, for example,
   through HTTP, determines the operation of an RTSP session. It also
   describes how a client should interpret SDP content returned in reply
   to a DESCRIBE request. SDP provides no mechanism by which a client



H. Schulzrinne et. al.                                       [Page 81]


Internet Draft                    RTSP                 February 22, 2002


   can distinguish, without human guidance, between several media
   streams to be rendered simultaneously and a set of alternatives
   (e.g., two audio streams spoken in different languages).

C.1 Definitions

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

C.1.1 Control URL

   The "a=control:" attribute is used to convey the control URL. This
   attribute is used both for the session and media descriptions. If
   used for individual media, it indicates the URL to be used for
   controlling that particular media stream. If found at the session
   level, the attribute indicates the URL for aggregate control.

   Example:

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



   This attribute may contain either relative and absolute URLs,
   following the rules and conventions set out in RFC 1808 [28].
   Implementations should look for a base URL in the following order:

        1.   the RTSP  Content-Base field;

        2.   the RTSP  Content-Location field;

        3.   the RTSP request URL.

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

C.1.2 Media Streams

   The "m=" field is used to enumerate the streams. It is expected that
   all the specified streams will be rendered with appropriate
   synchronization. If the session is unicast, the port number serves as
   a recommendation from the server to the client; the client still has
   to include it in its SETUP request and may ignore this
   recommendation.  If the server has no preference, it SHOULD set the
   port number value to zero.




H. Schulzrinne et. al.                                       [Page 82]


Internet Draft                    RTSP                 February 22, 2002


   Example:

     m=audio 0 RTP/AVP 31



C.1.3 Payload Type(s)

   The payload type(s) are specified in the "m=" field. In case the
   payload type is a static payload type from RFC 1890 [1], no other
   information is required. In case it is a dynamic payload type, the
   media attribute "rtpmap" is used to specify what the media is.  The
   "encoding name" within the "rtpmap" attribute may be one of those
   specified in RFC 1890 (Sections 5 and 6), or an experimental encoding
   with a "X-" prefix as specified in SDP (RFC 2327 [6]). Codec-specific
   parameters are not specified in this field, but rather in the "fmtp"
   attribute described below.  Implementors seeking to register new
   encodings should follow the procedure in RFC 1890 [1]. If the media
   type is not suited to the RTP AV profile, then it is recommended that
   a new profile be created and the appropriate profile name be used in
   lieu of "RTP/AVP" in the "m=" field.

C.1.4 Format-Specific Parameters

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

C.1.5 Range of Presentation

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

   Examples:

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



C.1.6 Time of Availability




H. Schulzrinne et. al.                                       [Page 83]


Internet Draft                    RTSP                 February 22, 2002


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

C.1.7 Connection Information

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

C.1.8 Entity Tag

   The optional "a=etag" attribute identifies a version of the session
   description. It is opaque to the client. SETUP requests may include
   this identifier in the  If-Match field (see section 12.23) to only
   allow session establishment if this attribute value still corresponds
   to that of the current description.  The attribute value is opaque
   and may contain any character allowed within SDP attribute values.

   Example:

     a=etag:158bb3e7c7fd62ce67f12b533f06b83a




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

C.2 Aggregate Control Not Available

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



H. Schulzrinne et. al.                                       [Page 84]


Internet Draft                    RTSP                 February 22, 2002


   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I came from a web page
   c=IN IP4 0.0.0.0
   t=0 0
   m=video 8002 RTP/AVP 31
   a=control:rtsp://audio.com/movie.aud
   m=audio 8004 RTP/AVP 3
   a=control:rtsp://video.com/movie.vid



   Note that the position of the control URL in the description implies
   that the client establishes separate RTSP control sessions to the
   servers audio.com and video.com

   It is recommended that an SDP file contains the complete media
   initialization information even if it is delivered to the media
   client through non-RTSP means. This is necessary as there is no
   mechanism to indicate that the client should request more detailed
   media stream information via DESCRIBE.

C.3 Aggregate Control Available

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

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

   Example:

   v=0
   o=- 2890844256 2890842807 IN IP4 204.34.34.32
   s=I contain
   i=<more info>
   c=IN IP4 0.0.0.0
   t=0 0
   a=control:rtsp://example.com/movie/



H. Schulzrinne et. al.                                       [Page 85]


Internet Draft                    RTSP                 February 22, 2002


   m=video 8002 RTP/AVP 31
   a=control:trackID=1
   m=audio 8004 RTP/AVP 3
   a=control:trackID=2



   In this example, the client is required to establish a single RTSP
   session to the server, and uses the URLs
   rtsp://example.com/movie/trackID=1 and
   rtsp://example.com/movie/trackID=2 to set up the video and audio
   streams, respectively. The URL rtsp://example.com/movie/ controls the
   whole movie.

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

D Minimal RTSP implementation

D.1 Client

   A client implementation MUST be able to do the following :

        o Generate the following requests: SETUP, TEARDOWN, and one of
          PLAY (i.e., a minimal playback client) or RECORD (i.e., a
          minimal recording client). If RECORD is implemented, ANNOUNCE
          MUST be implemented as well.

        o Include the following headers in requests:  CSeq, Connection,
          Session,  Transport. If ANNOUNCE is implemented, the
          capability to include headers Content-Language,  Content-
          Encoding, Content-Length, and  Content-Type should be as well.

        o Parse and understand the following headers in responses:
          CSeq,  Connection,  Session, Transport,  Content-Language,
          Content-Encoding,  Content-Length, Content-Type. If RECORD is
          implemented, the Location header must be understood as well.
          RTP-compliant implementations should also implement RTP-Info.

        o Understand the class of each error code received and notify
          the end-user, if one is present, of error codes in classes 4xx
          and 5xx. The notification requirement may be relaxed if the
          end-user explicitly does not want it for one or all status
          codes.

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



H. Schulzrinne et. al.                                       [Page 86]


Internet Draft                    RTSP                 February 22, 2002


          should implement the ANNOUNCE method, merely that it MUST
          respond positively or negatively to any request received from
          the server.

   Though not required, the following are RECOMMENDED.

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

        o Inclusion of the  User-Agent header.

        o Understand SDP session descriptions as defined in Appendix C

        o Accept media initialization formats (such as SDP) from
          standard input, command line, or other means appropriate to
          the operating environment to act as a "helper application" for
          other applications (such as web browsers).


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

D.1.1 Basic Playback

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

        o generate the PAUSE request;

        o implement the REDIRECT method, and the  Location header.

D.1.2 Authentication-enabled

   In order to access media presentations from RTSP servers that require
   authentication, the client MUST additionally be able to do the
   following:

        o recognize the 401 (Unauthorized) status code;

        o parse and include the  WWW-Authenticate header;

        o implement Basic Authentication and Digest Authentication.

D.2 Server

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



H. Schulzrinne et. al.                                       [Page 87]


Internet Draft                    RTSP                 February 22, 2002


        o Implement the following methods: SETUP, TEARDOWN, OPTIONS and
          either PLAY (for a minimal playback server) or RECORD (for a
          minimal recording server).

          If RECORD is implemented, ANNOUNCE SHOULD be implemented as
          well.

        o Include the following headers in responses:  Connection,
          Content-Length,  Content-Type,  Content-Language, Content-
          Encoding,  Transport,  Public. The capability to include the
          Location header should be implemented if the RECORD method is.
          RTP-compliant implementations should also implement the  RTP-
          Info field.

        o Parse and respond appropriately to the following headers in
          requests:  Connection,  Session,  Transport, Require.

   Though not required, the following are highly recommended at the time
   of publication for practical interoperability with initial
   implementations and/or to be a "good citizen".

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

        o Inclusion of the  Server header.

        o Implement the DESCRIBE method.

        o Generate SDP session descriptions as defined in Appendix C


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

D.2.1 Basic Playback

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

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

        o Implement the PAUSE method.

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



H. Schulzrinne et. al.                                       [Page 88]


Internet Draft                    RTSP                 February 22, 2002


   servers:

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

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

        o Include mappings from data-specific timestamps to NPT. When
          RTP is used, the  rtptime portion of the  RTP-Info field may
          be used to map RTP timestamps to NPT.


        Client implementations may use the presence of length
        information to determine if the clip is seekable, and
        visably disable seeking features for clips for which the
        length information is unavailable. A common use of the
        presentation length is to implement a "slider bar" which
        serves as both a progress indicator and a timeline
        positioning tool.

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

D.2.2 Authentication-enabled

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

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

        o Parse and include the  WWW-Authenticate header

        o Implement Basic Authentication and Digest Authentication

E Changes

   Since RFC 2326, the following issues were addressed:

        o http://rtsp.org/bug448521 - URLs in Rtp-Info need to be quoted

        o http://rtsp.org/bug448525 - Syntax for SSRC should be
          clarified

        o http://rtsp.org/bug461083 - Body w/o Content-Length
          clarification




H. Schulzrinne et. al.                                       [Page 89]


Internet Draft                    RTSP                 February 22, 2002


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

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

        o http://rtsp.org/bug477416 - BNF error section 3.6 NPT

        o http://rtsp.org/bug477421 - When to send response

        o http://rtsp.org/bug507347 - Removal of destination redirection

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

   A word-by-word diff from RFC 2326 can be found at
   http://rtsp.org/2002/drafts

F Author Addresses

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   electronic mail:  schulzrinne@cs.columbia.edu

   Anup Rao
   Cisco
   USA
   electronic mail:  anrao@cisco.com

   Robert Lanphier
   RealNetworks
   P.O. Box 91123
   Seattle, WA 98111-9223
   USA
   electronic mail:  robla@real.com

G Acknowledgements

   This draft is based on the functionality of the original RTSP draft
   submitted in October 1996. It also borrows format and descriptions
   from HTTP/1.1.

   This document has benefited greatly from the comments of all those
   participating in the MMUSIC-WG. In addition to those already



H. Schulzrinne et. al.                                       [Page 90]


Internet Draft                    RTSP                 February 22, 2002


   mentioned, the following individuals have contributed to this
   specification:

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

H Bibliography

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

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

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

   [4] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," Request for Comments 2119, Internet Engineering Task Force,
   Mar. 1997.

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

   [6] M. Handley and V. Jacobson, "SDP: session description protocol,"
   Request for Comments 2327, Internet Engineering Task Force, Apr.
   1998.

   [7] T. Dierks and C. Allen, "The TLS protocol version 1.0," Request
   for Comments 2246, Internet Engineering Task Force, Jan. 1999.

   [8] J. Franks, P. Hallam-Baker, J. Hostetler, P. Leach, A. Luotonen,
   E. Sink, and L. Stewart, "An extension to HTTP : Digest access



H. Schulzrinne et. al.                                       [Page 91]


Internet Draft                    RTSP                 February 22, 2002


   authentication," Request for Comments 2069, Internet Engineering Task
   Force, Jan. 1997.

   [9] J. Postel, "User datagram protocol," Request for Comments 768,
   Internet Engineering Task Force, Aug. 1980.

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

   [11] J. Postel, "Transmission control protocol," Request for Comments
   793, Internet Engineering Task Force, Sept. 1981.

   [12] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
   session initiation protocol," Request for Comments 2543, Internet
   Engineering Task Force, Mar. 1999.

   [13] 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.

   [14] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
   Request for Comments 1961, Internet Engineering Task Force, June
   1996.

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

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

   [17] D. Crocker, Ed., and P. Overell, "Augmented BNF for syntax
   specifications:  ABNF," Request for Comments 2234, Internet
   Engineering Task Force, Nov.  1997.

   [18] R. Braden and Ed, "Requirements for internet hosts - application
   and support," Request for Comments 1123, Internet Engineering Task
   Force, Oct.  1989.

   [19] R. Elz, "A compact representation of IPv6 addresses," Request
   for Comments 1924, Internet Engineering Task Force, Apr. 1996.

   [20] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource



H. Schulzrinne et. al.                                       [Page 92]


Internet Draft                    RTSP                 February 22, 2002


   locators (URL)," Request for Comments 1738, Internet Engineering Task
   Force, Dec.  1994.

   [21] H. Schulzrinne, "A comprehensive multimedia control architecture
   for the Internet," in Proc. International Workshop on Network and
   Operating System Support for Digital Audio and Video (NOSSDAV) , (St.
   Louis, Missouri), May 1997.

   [22] F. Yergeau, "UTF-8, a transformation format of ISO 10646,"
   Request for Comments 2279, Internet Engineering Task Force, Jan.
   1998.

   [23] R. Braden, "T/TCP -- TCP extensions for transactions functional
   specification," Request for Comments 1644, Internet Engineering Task
   Force, July 1994.

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

   [25] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
   protocol (RTSP)," Request for Comments 2326, Internet Engineering
   Task Force, Apr.  1998.

   [26] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
   identifiers (URI): generic syntax," Request for Comments 2396,
   Internet Engineering Task Force, Aug. 1998.

   [27] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
   a transport protocol for real-time applications," Request for
   Comments 1889, Internet Engineering Task Force, Jan. 1996.

   [28] R. Fielding, "Relative uniform resource locators," Request for
   Comments 1808, Internet Engineering Task Force, June 1995.


   Full Copyright Statement

   Copyright (C) The Internet Society (2002). All Rights Reserved.

   This document and translations of it may be copied and furnished to
   others, and derivative works that comment on or otherwise explain it
   or assist in its implmentation may be prepared, copied, published and
   distributed, in whole or in part, without restriction of any kind,
   provided that the above copyright notice and this paragraph are
   included on all such copies and derivative works. However, this
   document itself may not be modified in any way, such as by removing
   the copyright notice or references to the Internet Society or other
   Internet organizations, except as needed for the purpose of



H. Schulzrinne et. al.                                       [Page 93]


Internet Draft                    RTSP                 February 22, 2002


   developing Internet standards in which case the procedures for
   copyrights defined in the Internet Standards process must be
   followed, or as required to translate it into languages other than
   English.

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

   This document and the information contained herein is provided on an
   "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
   TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
   BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
   HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
   MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.





































H. Schulzrinne et. al.                                       [Page 94]


                           Table of Contents



   1          Introduction ........................................    3
   1.1        Purpose .............................................    3
   1.2        Requirements ........................................    4
   1.3        Terminology .........................................    4
   1.4        Protocol Properties .................................    7
   1.5        Extending RTSP ......................................    8
   1.6        Overall Operation ...................................    9
   1.7        RTSP States .........................................   10
   1.8        Relationship with Other Protocols ...................   11
   2          Notational Conventions ..............................   12
   3          Protocol Parameters .................................   12
   3.1        H3.1 ................................................   12
   3.2        RTSP URL ............................................   12
   3.3        Conference Identifiers ..............................   14
   3.4        Session Identifiers .................................   14
   3.5        SMPTE Relative Timestamps ...........................   14
   3.6        Normal Play Time ....................................   15
   3.7        Absolute Time .......................................   16
   3.8        Option Tags .........................................   17
   3.8.1      Registering New Option Tags with IANA ...............   17
   4          RTSP Message ........................................   17
   4.1        Message Types .......................................   18
   4.2        Message Headers .....................................   18
   4.3        Message Body ........................................   18
   4.4        Message Length ......................................   18
   5          General Header Fields ...............................   19
   6          Request .............................................   19
   6.1        Request Line ........................................   19
   6.2        Request Header Fields ...............................   20
   7          Response ............................................   21
   7.1        Status-Line .........................................   21
   7.1.1      Status Code and Reason Phrase .......................   21
   7.1.2      Response Header Fields ..............................   24
   8          Entity ..............................................   24
   8.1        Entity Header Fields ................................   25
   8.2        Entity Body .........................................   25
   9          Connections .........................................   25
   9.1        Pipelining ..........................................   27
   9.2        Reliability and Acknowledgements ....................   27
   10         Method Definitions ..................................   28
   10.1       OPTIONS .............................................   28



H. Schulzrinne et. al.                                        [Page 1]


Internet Draft                    RTSP                 February 22, 2002


   10.2       DESCRIBE ............................................   29
   10.3       ANNOUNCE ............................................   30
   10.4       SETUP ...............................................   31
   10.5       PLAY ................................................   32
   10.6       PAUSE ...............................................   34
   10.7       TEARDOWN ............................................   36
   10.8       GET_PARAMETER .......................................   36
   10.9       SET_PARAMETER .......................................   37
   10.10      REDIRECT ............................................   38
   10.11      RECORD ..............................................   38
   10.12      Embedded (Interleaved) Binary Data ..................   39
   11         Status Code Definitions .............................   40
   11.1       Success 2xx .........................................   40
   11.1.1     250 Low on Storage Space ............................   40
   11.2       Redirection 3xx .....................................   40
   11.3       Client Error 4xx ....................................   41
   11.4       400 Bad Request .....................................   41
   11.4.1     405 Method Not Allowed ..............................   41
   11.4.2     451 Parameter Not Understood ........................   41
   11.4.3     452 Conference Not Found ............................   41
   11.4.4     453 Not Enough Bandwidth ............................   41
   11.4.5     454 Session Not Found ...............................   41
   11.4.6     455 Method Not Valid in This State ..................   41
   11.4.7     456 Header Field Not Valid for Resource .............   42
   11.4.8     457 Invalid Range ...................................   42
   11.4.9     458 Parameter Is Read-Only ..........................   42
   11.4.10    459 Aggregate Operation Not Allowed .................   42
   11.4.11    460 Only Aggregate Operation Allowed ................   42
   11.4.12    461 Unsupported Transport ...........................   42
   11.4.13    462 Destination Unreachable .........................   42
   11.5       Server Error 5xx ....................................   42
   11.5.1     551 Option not supported ............................   42
   12         Header Field Definitions ............................   43
   12.1       Accept ..............................................   43
   12.2       Accept-Encoding .....................................   43
   12.3       Accept-Language .....................................   43
   12.4       Accept-Ranges .......................................   44
   12.5       Allow ...............................................   44
   12.6       Authorization .......................................   44
   12.7       Bandwidth ...........................................   44
   12.8       Blocksize ...........................................   44
   12.9       Cache-Control .......................................   46
   12.10      Conference ..........................................   48
   12.11      Connection ..........................................   48
   12.12      Content-Base ........................................   48
   12.13      Content-Encoding ....................................   48
   12.14      Content-Language ....................................   48
   12.15      Content-Length ......................................   49



H. Schulzrinne et. al.                                        [Page 2]


Internet Draft                    RTSP                 February 22, 2002


   12.16      Content-Location ....................................   49
   12.17      Content-Type ........................................   49
   12.18      CSeq ................................................   49
   12.19      Date ................................................   49
   12.20      Expires .............................................   49
   12.21      From ................................................   50
   12.22      Host ................................................   50
   12.23      If-Match ............................................   50
   12.24      If-Modified-Since ...................................   51
   12.25      Last-Modified .......................................   51
   12.26      Location ............................................   51
   12.27      Proxy-Authenticate ..................................   51
   12.28      Proxy-Require .......................................   52
   12.29      Public ..............................................   52
   12.30       Range ..............................................   52
   12.31      Referer .............................................   53
   12.32      Retry-After .........................................   53
   12.33      Require .............................................   53
   12.34      RTP-Info ............................................   54
   12.35      Scale ...............................................   55
   12.36      Speed ...............................................   56
   12.37      Server ..............................................   57
   12.38      Session .............................................   57
   12.39      Timestamp ...........................................   58
   12.40      Transport ...........................................   58
   12.41      Unsupported .........................................   62
   12.42      User-Agent ..........................................   62
   12.43      Vary ................................................   62
   12.44      Via .................................................   62
   12.45      WWW-Authenticate ....................................   62
   13         Caching .............................................   62
   14         Examples ............................................   63
   14.1       Media on Demand (Unicast) ...........................   63
   14.2       Streaming of a Container file .......................   65
   14.3       Single Stream Container Files .......................   68
   14.4       Live Media Presentation Using Multicast .............   70
   14.5       Playing media into an existing session ..............   71
   14.6       Recording ...........................................   72
   15         Syntax ..............................................   73
   15.1       Base Syntax .........................................   73
   16         Security Considerations .............................   74
   A          RTSP Protocol State Machines ........................   77
   A.1        Client State Machine ................................   77
   A.2        Server State Machine ................................   79
   B          Interaction with RTP ................................   80
   C          Use of SDP for RTSP Session Descriptions ............   81
   C.1        Definitions .........................................   82
   C.1.1      Control URL .........................................   82



H. Schulzrinne et. al.                                        [Page 3]


Internet Draft                    RTSP                 February 22, 2002


   C.1.2      Media Streams .......................................   82
   C.1.3      Payload Type(s) .....................................   83
   C.1.4      Format-Specific Parameters ..........................   83
   C.1.5      Range of Presentation ...............................   83
   C.1.6      Time of Availability ................................   83
   C.1.7      Connection Information ..............................   84
   C.1.8      Entity Tag ..........................................   84
   C.2        Aggregate Control Not Available .....................   84
   C.3        Aggregate Control Available .........................   85
   D          Minimal RTSP implementation .........................   86
   D.1        Client ..............................................   86
   D.1.1      Basic Playback ......................................   87
   D.1.2      Authentication-enabled ..............................   87
   D.2        Server ..............................................   87
   D.2.1      Basic Playback ......................................   88
   D.2.2      Authentication-enabled ..............................   89
   E          Changes .............................................   89
   F          Author Addresses ....................................   90
   G          Acknowledgements ....................................   90
   H          Bibliography ........................................   91































H. Schulzrinne et. al.                                        [Page 4]


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