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

                  Real Time Streaming Protocol (RTSP)

STATUS OF THIS MEMO

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

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

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

   Distribution of this document is unlimited.

  Abstract:

   The Real Time Streaming Protocol, or RTSP, is an application-level
   protocol for control over the delivery of data with real-time
   properties. RTSP provides an extensible framework to enable
   controlled, on-demand delivery of real-time data, such as audio and
   video. Sources of data can include both live data feeds and stored
   clips. This protocol is intended to control multiple data delivery
   sessions, provide a means for choosing delivery channels such as UDP,
   multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC
   1889).

















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Contents

     * 1 Introduction
          + 1.1 Purpose
          + 1.2 Requirements
          + 1.3 Terminology
          + 1.4 Protocol Properties
          + 1.5 Extending RTSP
          + 1.6 Overall Operation
          + 1.7 RTSP States
          + 1.8 Relationship with Other Protocols
     * 2 Notational Conventions
     * 3 Protocol Parameters
          + 3.1 RTSP Version
          + 3.2 RTSP URL
          + 3.3 Conference Identifiers
          + 3.4 Session Identifiers
          + 3.5 SMPTE Relative Timestamps
          + 3.6 Normal Play Time
          + 3.7 Absolute Time
     * 4 RTSP Message
          + 4.1 Message Types
          + 4.2 Message Headers
          + 4.3 Message Body
          + 4.4 Message Length
     * 5 General Header Fields
     * 6 Request
          + 6.1 Request Line
          + 6.2 Request Header Fields
     * 7 Response
          + 7.1 Status-Line
               o 7.1.1 Status Code and Reason Phrase
               o 7.1.2 Response Header Fields
     * 8 Entity
          + 8.1 Entity Header Fields
          + 8.2 Entity Body
     * 9 Connections
          + 9.1 Pipelining
          + 9.2 Reliability and Acknowledgements
     * 10 Method Definitions
          + 10.1 OPTIONS
          + 10.2 DESCRIBE
          + 10.3 ANNOUNCE
          + 10.4 SETUP
          + 10.5 PLAY
          + 10.6 PAUSE
          + 10.7 TEARDOWN
          + 10.8 GET_PARAMETER
          + 10.9 SET_PARAMETER
          + 10.10 REDIRECT



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          + 10.11 RECORD
          + 10.12 Embedded (Interleaved) Binary Data
     * 11 Status Code Definitions
          + 11.1 Redirection 3xx
          + 11.2 Client Error 4xx
               o 11.2.1 405 Method Not Allowed
               o 11.2.2 451 Parameter Not Understood
               o 11.2.3 452 Conference Not Found
               o 11.2.4 453 Not Enough Bandwidth
               o 11.2.5 45x Session Not Found
               o 11.2.6 45x Method Not Valid in This State
               o 11.2.7 45x Header Field Not Valid for Resource
               o 11.2.8 45x Invalid Range
               o 11.2.9 45x Parameter Is Read-Only
               o 11.2.10 45x Aggregate operation not allowed
               o 11.2.11 45x Only aggregate operation allowed
     * 12 Header Field Definitions
          + 12.1 Accept
          + 12.2 Accept-Encoding
          + 12.3 Accept-Language
          + 12.4 Allow
          + 12.5 Authorization
          + 12.6 Bandwidth
          + 12.7 Blocksize
          + 12.8 C-PEP
          + 12.9 C-PEP-Info
          + 12.10 Cache-Control
          + 12.11 Conference
          + 12.12 Connection
          + 12.13 Content-Encoding
          + 12.14 Content-Language
          + 12.15 Content-Length
          + 12.16 Content-Type
          + 12.17 Date
          + 12.18 Expires
          + 12.19 From
          + 12.20 Host
          + 12.21 If-Modified-Since
          + 12.22 Last-Modified
          + 12.23 Location
          + 12.24 PEP
          + 12.25 PEP-Info
          + 12.26 Proxy-Authenticate
          + 12.27 Public
          + 12.28 Range
          + 12.29 Referer
          + 12.30 Retry-After
          + 12.31 Scale
          + 12.32 Speed
          + 12.33 Server



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          + 12.34 Session
          + 12.35 Transport
          + 12.36 Transport-Info
          + 12.37 User-Agent
          + 12.38 Vary
          + 12.39 Via
          + 12.40 WWW-Authenticate
     * 13 Caching
     * 14 Examples
          + 14.1 Media on Demand (Unicast)
          + 14.2 Streaming of a Container file
          + 14.3 Live Media Presentation Using Multicast
          + 14.4 Playing media into an existing session
          + 14.5 Recording
     * 15 Syntax
          + 15.1 Base Syntax
     * 16 Security Considerations
     * A RTSP Protocol State Machines
          + A.1 Client State Machine
          + A.2 Server State Machine
     * B Open Issues
     * C Changes
     * D Author Addresses
     * E Acknowledgements
     * References

1 Introduction

1.1 Purpose

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








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

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

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

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






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

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

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

1.2 Requirements

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

1.3 Terminology

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

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

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

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











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

   Participant:
          Participants are members of conferences. A participant may be a
          machine, e.g., a media record or playback server.

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

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

   (Media) stream:
          A single media instance, e.g., an audio stream or a video
          stream as well as a single whiteboard or shared application
          group. When using RTP, a stream consists of all RTP and RTCP
          packets created by a source within an RTP session. This is
          equivalent to the definition of a DSM-CC stream([18]).

   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.

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



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

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

1.4 Protocol Properties

   RTSP has the following properties:

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

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

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

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




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

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

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

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

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

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

   HTTP-friendly:
          Where sensible, RTSP re-uses HTTP concepts, so that the
          existing infrastructure can be re-used. This infrastructure
          includes PICS (Platform for Internet Content Selection [20])
          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.



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

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

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

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








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

1.6 Overall Operation

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

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

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

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




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

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

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

1.7 RTSP States

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

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



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1.8 Relationship with Other Protocols

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

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

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

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

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

2 Notational Conventions

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

   All the mechanisms specified in this document are described in both
   prose and an augmented Backus-Naur form (BNF) similar to that used in
   RFC 2068 [H2.1]. It is described in detail in [14].





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   In this draft, we use indented and smaller-type paragraphs to provide
   background and motivation. Some of these paragraphs are marked with
   HS, AR and RL, designating opinions and comments by the individual
   authors which may not be shared by the co-authors and require
   resolution.

3 Protocol Parameters

3.1 RTSP Version

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

3.2 RTSP URL

     The ``rtsp'', ``rtspu'' and ``rtsps'' 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:" | "rtsps:" )
                "//" host [ ":" port ] [abs_path]
  host     = <A legal Internet host domain name of IP address
             (in dotted decimal form), as defined by Section 2.1
             of RFC 1123>
  port     = *DIGIT

   abs_path is defined in [H3.2.1].

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

   The scheme rtsp requires that commands are issued via a reliable
   protocol (within the Internet, TCP), while the scheme rtspu identifies
   an unreliable protocol (within the Internet, UDP). The scheme rtsps
   indicates that a TCP connection secured by TLS [7] must be used.

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

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





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   A presentation or a stream is identified by an textual media
   identifier, using the character set and escape conventions [H3.2] of
   URLs [16]. URLs may refer to a stream or an aggregate of streams ie. a
   presentation. Accordingly, requests described in Section 10 can apply
   to either the whole presentation or an individual stream within the
   presentation. Note that some request methods can only be applied to
   streams, not presentations and vice versa.

   For example, the RTSP URL
  rtsp://media.example.com:554/twister/audiotrack

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

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

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

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

   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.









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

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

3.4 Session Identifiers

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

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

3.5 SMPTE Relative Timestamps

     A SMPTE relative time-stamp expresses time relative to the start of
   the clip. Relative timestamps are expressed as SMPTE time codes for
   frame-level access accuracy. The time code has the format
   hours:minutes:seconds:frames.subframes, with the origin at the start
   of the clip. For NTSC, the frame rate is 29.97 frames per second. This
   is handled by 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" "=" smpte-time "-" [ smpte-time ]
  smpte-time = 2DIGIT ":" 2DIGIT ":" 2DIGIT [ ":" 2DIGIT ] [ "." 2DIGIT]

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





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3.6 Normal Play Time

   Normal play time (NPT) indicates the stream absolute position relative
   to the beginning of the presentation, measured in seconds and
   microseconds. The beginning of a presentation corresponds to 0 seconds
   and 0 microseconds. Negative values are not defined. The microsecond
   field is always less than 1,000,000. NPT is defined as in DSM-CC [18]:
   ``Intuitively, NPT is the clock the viewer associates with a program.
   It is often digitally displayed on a VCR. NPT advances normally when
   in normal play mode (scale = 1), advances at a faster rate when in
   fast scan forward (high positive scale ratio), decrements when in scan
   reverse (high negative scale ratio) and is fixed in pause mode. NPT is
   (logically) equivalent to SMPTE time codes.'' [18]

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

   Examples:
  npt=123:45-125

3.7 Absolute Time

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

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

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

  19961108T143720.25Z

   Example

4 RTSP Message

     RTSP is a text-based protocol and uses the ISO 10646 character set
   in UTF-8 encoding (RFC 2044). Lines are terminated by CRLF, but
   receivers should be prepared to also interpret CR and LF by themselves
   as line terminators.






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

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

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

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

4.1 Message Types

   See [H4.1]

4.2 Message Headers

   See [H4.2]

4.3 Message Body

     See [H4.3]

4.4 Message Length

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











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   1.
          Any response message which MUST NOT include a message-body
          (such as the 1xx, 204, and 304 responses) is always terminated
          by the first empty line after the header fields, regardless of
          the entity-header fields present in the message. (Note: An
          empty line consists of only CRLF.)
   2.
          If a Content-Length header field (section 12.15) is present,
          its value in bytes represents the length of the message-body.
          If this header field is not present, a value of zero is
          assumed.
   3.
          By the server closing the connection. (Closing the connection
          cannot be used to indicate the end of a request body, since
          that would leave no possibility for the server to send back a
          response.)

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

     Given the moderate length of presentation descriptions returned,
     the server should always be able to determine its length, even if
     it is generated dynamically, making the chunked transfer encoding
     unnecessary. Even though Content-Length must be present if there is
     any entity body, the rules ensure reasonable behavior even if the
     length is not given explicitly.

5 General Header Fields

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

      general-header     =     Cache-Control     ; Section 12.10
                         |     Connection        ; Section 12.12
                         |     Date              ; Section 12.17
                         |     Via               ; Section 12.39

6 Request

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





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

6.1 Request Line

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

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

  extension-method = token

  Request-URI = "*" | absolute_URI

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

  seq-no = 1*DIGIT

6.2 Request Header Fields

  request-header  =          Accept                   ; Section 12.1
                  |          Accept-Encoding          ; Section 12.2
                  |          Accept-Language          ; Section 12.3
                  |          Authorization            ; Section 12.5
                  |          From                     ; Section 12.19
                  |          If-Modified-Since        ; Section 12.21
                  |          Range                    ; Section 12.28
                  |          Referer                  ; Section 12.29
                  |          User-Agent               ; Section 12.37






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

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

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

  OPTIONS * RTSP/1.0

7 Response

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

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

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

7.1 Status-Line

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









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

  7.1.1 Status Code and Reason Phrase

   The Status-Code element is a 3-digit integer result code of the
   attempt to understand and satisfy the request. These codes are fully
   defined in section11. The Reason-Phrase is intended to give a short
   textual description of the Status-Code. The Status-Code is intended
   for use by automata and the Reason-Phrase is intended for the human
   user. The client is not required to examine or display the
   Reason-Phrase.

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

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

   The individual values of the numeric status codes defined for
   RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
   presented below. The reason phrases listed here are only recommended -
   they may be replaced by local equivalents without affecting the
   protocol. Note that RTSP adopts most HTTP/1.1 status codes and adds
   RTSP-specific status codes in the starting at 450 to avoid conflicts
   with newly defined HTTP status codes.
















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   Status-Code    = "100"   ; Continue
                  | "200"   ; OK
                  | "201"   ; Created
                  | "300"   ; Multiple Choices
                  | "301"   ; Moved Permanently
                  | "302"   ; Moved Temporarily
                  | "303"   ; See Other
                  | "304"   ; Not Modified
                  | "305"   ; Use Proxy
                  | "400"   ; Bad Request
                  | "401"   ; Unauthorized
                  | "402"   ; Payment Required
                  | "403"   ; Forbidden
                  | "404"   ; Not Found
                  | "405"   ; Method Not Allowed
                  | "406"   ; Not Acceptable
                  | "407"   ; Proxy Authentication Required
                  | "408"   ; Request Time-out
                  | "409"   ; Conflict
                  | "410"   ; Gone
                  | "411"   ; Length Required
                  | "412"   ; Precondition Failed
                  | "413"   ; Request Entity Too Large
                  | "414"   ; Request-URI Too Large
                  | "415"   ; Unsupported Media Type
                  | "451"   ; Parameter Not Understood
                  | "452"   ; Conference Not Found
                  | "453"   ; Not Enough Bandwidth
                  | "45x"   ; Session Not Found
                  | "45x"   ; Method Not Valid in This State
                  | "45x"   ; Header Field Not Valid for Resource
                  | "45x"   ; Invalid Range
                  | "45x"   ; Parameter Is Read-Only
                  | "45x"   ; Aggregate operation not allowed
                  | "45x"   ; Only aggregate operation allowed
                  | "500"   ; Internal Server Error
                  | "501"   ; Not Implemented
                  | "502"   ; Bad Gateway
                  | "503"   ; Service Unavailable
                  | "504"   ; Gateway Time-out
                  | "505"   ; RTSP Version not supported
                  | extension-code







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   extension-code = 3DIGIT

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

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
































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   Code          Reason
   100           Continue                         all
   200           OK                               all
   201           Created                          RECORD
   300           Multiple Choices                 all
   301           Moved Permanently                all
   302           Moved Temporarily                all
   303           See Other                        all
   305           Use Proxy                        all
   400           Bad Request                      all
   401           Unauthorized                     all
   402           Payment Required                 all
   403           Forbidden                        all
   404           Not Found                        all
   405           Method Not Allowed               all
   406           Not Acceptable                   all
   407           Proxy Authentication Required    all
   408           Request Timeout                  all
   409           Conflict                         RECORD
   410           Gone                             all
   411           Length Required                  SETUP
   412           Precondition Failed              DESCRIBE, SETUP
   413           Request Entity Too Large         SETUP
   414           Request-URI Too Long             all
   415           Unsupported Media Type           SETUP
   45x           Session not found                all
   45x           Invalid parameter                SETUP
   45x           Not Enough Bandwidth             SETUP
   45x           Illegal Conference Identifier    SETUP
   45x           Illegal Session Identifier       PLAY, RECORD, TEARDOWN
   45x           Parameter Is Read-Only           SET_PARAMETER
   45x           Header Field Not Valid           all
   45x           Method Not Valid In This State   all
   45x           Aggregate operation not allowed  all
   45x           Only aggregate operation allowed  all
   500           Internal Server Error            all
   501           Not Implemented                  all
   502           Bad Gateway                      all
   503           Service Unavailable              all
   504           Gateway Timeout                  all
   505           RTSP Version Not Supported       all
!
   Table 1: Status codes and their usage with RTSP methods






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  7.1.2 Response Header Fields

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

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

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

8 Entity

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

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

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.











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     entity-header       =    Allow               ; Section 12.4
                         |    Content-Encoding    ; Section 12.13
                         |    Content-Language    ; Section 12.14
                         |    Content-Length      ; Section 12.15
                         |    Content-Type        ; Section 12.16
                         |    Expires             ; Section 12.18
                         |    Last-Modified       ; Section 12.22
                         |    extension-header
     extension-header    =    message-header

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

8.2 Entity Body

   See [H7.2]

9 Connections

     RTSP requests can be transmitted in several different ways:

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

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

   Unlike HTTP, RTSP allows the media server to send requests to the
   media client. However, this is only supported for persistent
   connections, as the media server otherwise has no reliable way of
   reaching the client. Also, this is the only way that requests from
   media server to client are likely to traverse firewalls.

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.




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9.2 Reliability and Acknowledgements

   Requests are acknowledged by the receiver unless they are sent to a
   multicast group. If there is no acknowledgement, the sender may resend
   the same message after a timeout of one round-trip time (RTT). The
   round-trip time is estimated as in TCP (RFC TBD), with an initial
   round-trip value of 500 ms. An implementation MAY cache the last RTT
   measurement as the initial value for future connections. If a reliable
   transport protocol is used to carry RTSP, the timeout value MAY be set
   to an arbitrarily large value.

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

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

     This avoids ambiguities when computing round-trip time estimates.

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

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

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




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

10 Method Definitions

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

      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               P,S        required
      PAUSE             C->S               P,S        recommended
      PLAY              C->S               P,S        required
      RECORD            C->S               P,S        optional
      REDIRECT          S->C               P,S        optional
      SETUP             C->S               S          required
      SET_PARAMETER     C->S, S->C         P,S        optional
      TEARDOWN          C->S               P,S        required
!
   Table 2: Overview of RTSP methods, their direction, and what objects (P:
   presentation, S: stream) they operate on

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

10.1 OPTIONS

     The behavior is equivalent to that described in [H9.2]. An OPTIONS
   request may be issued at any time, e.g., if the client is about to try
   a non-standard request. It does not influence server state.






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   Example :
  C->S:  OPTIONS * RTSP/1.0 1
         PEP: {{map "http://www.iana.org/rtsp/implicit-play"}}
              {{map "http://www.iana.org/rtsp/record-feature"}}
         C-PEP: {{map "http://www.iana.org/rtsp/udp-control"}}
                {{map "http://www.iana.org/rtsp/gzipped-messages"}}

  S->C:  RTSP/1.0 200 2 OK
         PEP-Info: {{map "http://www.iana.org/rtsp/implicit-play"}
                    {for "/" *}}
                   {{map "http://www.iana.org/rtsp/record-feature"}
                    {for "/" *}}
         C-PEP-Info: {{map "http://www.iana.org/rtsp/udp-control"}
                      {for "/" *}}
                     {{map "http://www.iana.org/rtsp/gzipped-messages"}
                      {for "/" *}}
         Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE

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

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.

   Example:

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

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











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        v=0
        o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
        s=SDP Seminar
        i=A Seminar on the session description protocol
        u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
        e=mjh@isi.edu (Mark Handley)
        c=IN IP4 224.2.17.12/127
        t=2873397496 2873404696
        a=recvonly
        m=audio 3456 RTP/AVP 0
        m=video 2232 RTP/AVP 31
        m=whiteboard 32416 UDP WB
        a=orient:portrait

ANNOUNCE

     The ANNOUNCE method serves two purposes:

   When sent from client to server, ANNOUNCE posts the description of a
   presentation or media object identified by the request URL to a
   server.

   When sent from server to client, ANNOUNCE updates the session
   description in real-time.

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

   Example:

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













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        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/1.0 200 312 OK

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 it, it must respond with error
   ``45x Method not valid in this state'' ). For the benefit of any
   intervening firewalls, a client must indicate the transport parameters
   even if it has no influence over these parameters, for example, where
   the server advertises a fixed multicast address.

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

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

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

  S->C: RTSP/1.0 200 302 OK
        Date: 23 Jan 1997 15:35:06 GMT
        Transport: RTP/AVP;port=4588









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PLAY

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

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

     This allows precise editing.

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

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

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

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

   See the description of the PAUSE request for further examples.

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

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






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

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

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

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

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

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

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

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

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

PAUSE

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






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

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

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

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

   Example:

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

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

TEARDOWN

     Stop 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.




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   Example:

  C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 892
        Session: 1234

  S->C: RTSP/1.0 200 892 OK

GET_PARAMETER

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

   Example:

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

        packets_received
        jitter

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

        packets_received: 10
        jitter: 0.3838

SET_PARAMETER

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

   A request SHOULD only contain a single parameter to allow the client
   to determine why a particular request failed. A server MUST allow a
   parameter to be set repeatedly to the same value, but it MAY disallow
   changing parameter values.

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



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

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

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

   Example:

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

        barparam: barstuff

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

        barparam

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.

   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/1.0 732
        Location: rtsp://bigserver.com:8001
        Range: clock=19960213T143205Z-










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RECORD

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

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

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

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

  C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0 954
        Session: 1234
        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 12.35.

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





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

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

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

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

11 Status Code Definitions

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

11.1 Redirection 3xx

   See [H10.3].

   Within RTSP, redirection may be used for load balancing or redirecting
   stream requests to a server topologically closer to the client.
   Mechanisms to determine topological proximity are beyond the scope of
   this specification.

11.2 Client Error 4xx

  11.2.1 405 Method Not Allowed

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

  11.2.2 451 Parameter Not Understood

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



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  11.2.3 452 Conference Not Found

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

  11.2.4 453 Not Enough Bandwidth

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

  11.2.5 45x Session Not Found

   The RTSP session identifier is invalid or has timed out.

  11.2.6 45x Method Not Valid in This State

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

  11.2.7 45x Header Field Not Valid for Resource

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

  11.2.8 45x Invalid Range

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

  11.2.9 45x Parameter Is Read-Only

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

  11.2.10 45x Aggregate operation not allowed

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

  11.2.11 45x Only aggregate operation allowed

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




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12 Header Field Definitions

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

   Tables 3 summarizes the header fields used by RTSP. Type ``g''
   designates general request headers, to be found in both requests and
   responses, type ``R'' designates request headers, type ``r'' response
   headers, type ``e'' 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 'r' will be send in every request of
   this type; merely, that client (for response headers) and server (for
   request headers) MUST implement them. The last column lists the method
   for which this header field is meaningful; the designation ``entity''
   refers to all methods that return a message body. Within this
   specification, DESCRIBE and GET_PARAMETER fall into this class.

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




























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

12.1 Accept

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





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

   See [H14.1] for syntax.

   Example of use:
  Accept: application/rtsl, application/sdp;level=2

12.2 Accept-Encoding

     See [H14.3]

12.3 Accept-Language

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

12.4 Allow

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

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

12.5 Authorization

     See [H14.8]

12.6 Bandwidth

     The Bandwidth request header field describes the estimated bandwidth
   available to the client, expressed as a positive integer and measured
   in bits per second.

   The bandwidth available to the client may change during an RTSP
   session, e.g., due to modem retraining.

  Bandwidth  = "Bandwidth" ":" 1*DIGIT

   Example:
  Bandwidth: 4000



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12.7 Blocksize

     This request header field is sent from the client to the media
   server asking the server for a particular media packet size. This
   packet size does not include lower-layer headers such as IP, UDP, or
   RTP. The server is free to use a blocksize which is lower than the one
   requested. The server MAY truncate this packet size to the closest
   multiple of the minimum media-specific block size or override it with
   the media specific size if necessary. The block size is a strictly
   positive decimal number and measured in octets. The server only
   returns an error (416) if the value is syntactically invalid.

12.8 C-PEP

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

12.9 C-PEP-Info

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

12.10 Cache-Control

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

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

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






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  Cache-Control   = "Cache-Control" ":" 1#cache-directive

  cache-directive = cache-request-directive
                  | cache-response-directive

  cache-request-directive =
        "no-cache"
      | "max-stale"
      | "min-fresh"
      | "only-if-cached"
      | cache-extension

  cache-response-directive =
        "public"
      | "private"
      | "no-cache"
      | "no-transform"
      | "must-revalidate"
      | "proxy-revalidate"
      | "max-age" "=" delta-seconds
      | cache-extension

      cache-extension = token [ "=" ( token | quoted-string ) ]

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

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

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












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

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

   max-stale:
          Indicates that the client is willing to accept a media stream
          that has exceeded its expiration time. If max-stale is assigned
          a value, then the client is willing to accept a response that
          has exceeded its expiration time by no more than the specified
          number of seconds. If no value is assigned to max-stale, then
          the client is willing to accept a stale response of any age.

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





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

12.11 Conference

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

  Conference = "Conference" ":" conference-id

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

12.12 Connection

     See [H14.10].

12.13 Content-Encoding

     See [H14.12]

12.14 Content-Language

     See [H14.13]

12.15 Content-Length

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

12.16 Content-Type

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





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12.17 Date

     See [H14.19].

12.18 Expires

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

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

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

  Expires = "Expires" ":" HTTP-date

   An example of its use is

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

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

   To mark a response as "already expired," an origin server should use
   an Expires date that is equal to the Date header value.

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

   The presence of an Expires header field with a date value of some time
   in the future on a media stream that otherwise would by default be
   non-cacheable indicates that the media stream is cachable, unless
   indicated otherwise by a Cache-Control header field (Section 12.10).








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12.19 From

     See [H14.22].

12.20 Host

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

12.21 If-Modified-Since

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

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

   An example of the field is:

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

12.22 Last-Modified

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

12.23 Location

     See [H14.30].

12.24 PEP

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







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12.25 PEP-Info

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

12.26 Proxy-Authenticate

     See [H14.33].

12.27 Public

   See [H14.35].

12.28 Range

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

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

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

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

     The notation is similar to that used for the HTTP/1.1 header. It
     allows to select a clip from the media object, to play from a given
     point to the end and from the current location to a given point.
     The start of playback can be scheduled for at any time in the
     future, although a server may refuse to keep server resources for
     extended idle periods.

12.29 Referer

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







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12.30 Retry-After

     See [H14.38].

12.31 Scale

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

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

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

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

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

   Example of playing in reverse at 3.5 times normal rate:

  Scale: -3.5

12.32 Speed

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







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

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

   Example:
  Speed: 2.5

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

12.33 Server

     See [H14.39]

12.34 Session

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

  Session  = "Session" ":" session-id

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

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





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12.35 Transport

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

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

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

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

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

   The syntax for the transport specifier is
   transport/profile/lower-transport. Defaults for "lower-transport" are
   specific to the profile. For RTP/AVP, the default is UDP.

   Below are the configuration parameters associated with transport:

   General parameters:

   destination:
          The address to which a stream will be sent. The client may
          specify the multicast address with the destination parameter. 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 to avoid becoming the
          unwitting perpetrator of a remote-controlled denial-of-service
          attack. This is particularly important if RTSP commands are
          issued via UDP, but TCP cannot be relied upon as reliable means
          of client identification by itself. A server SHOULD not allow a
          client to direct media streams to an address that differs from
          the address commands are coming from.







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   mode:
          The mode parameter indicates the methods to be supported for
          this session. Valid values are PLAY and RECORD. If not
          provided, the default is PLAY. For RECORD, the append flag
          indicates that the media data should be appended to the
          existing resource rather than overwriting it. If appending is
          requested and the server does not support this, it MUST refuse
          the request rather than overwrite the resouce identified by the
          URI. The append parameter is ignored if the mode parameter does
          not contain RECORD.

   interleaved:
          The interleaved parameter implies mixing the media stream with
          the control stream, in whatever protocol is being used by the
          control stream. Currently, the next-layer protocols RTP is
          defined. The `channel' parameter defines the channel number to
          be used in the $ statement (see section 10.12).

   Multicast specific:

   ttl:
          multicast time-to-live

   RTP Specific:

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

   port:
          RTP/RTCP destination ports on client. The client receives RTCP
          reports on the value of port plus one, as is standard RTP
          convention.

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

   ssrc:
          Indicates the RTP SSRC [19, Sec. 3] value that should be
          (request) or will be (response) used by the media server. This
          parameter is only valid for unicast transmission. It identifies
          the synchronization source to be associated with the media
          stream.





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

   Example:
  Transport: RTP/AVP;compressed;ttl=127;port=3456;
    mode="PLAY,RECORD;append"

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

12.36 Transport-Info

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

   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.







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

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

12.37 User-Agent

     See [H14.42]

12.38 Vary

     See [H14.43]

12.39 Via

     See [H14.44].

12.40 WWW-Authenticate

     See [H14.46].

13 Caching

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









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

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

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

14 Examples

     The following examples reference stream description formats that are
   not finalized, such as RTSL and SDP. Please do not use these examples
   as a reference for those formats.

14.1 Media on Demand (Unicast)

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



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   In our example, the client is only interested in the last part of the
   movie. The server requires authentication for this movie.

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

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

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

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

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

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

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

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

V->C: RTSP/1.0 200 2 OK

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

A->C: RTSP/1.0 200 2 OK

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





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A->C: RTSP/1.0 200 3 OK

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

V->C: RTSP/1.0 200 3 OK

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

14.2 Streaming of a Container file

   For purposes of this example, a container file is a storage entity in
   which multiple continuous media types pertaining to the same end-user
   presentation are present. In effect, the container file represents a
   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 essentially transported as independant
   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 retreival of the streams by 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 a RTSP URL to the
   container file.

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

M->C: RTSP/1.0 200 1 OK
      Content-Type: application/sdp
      Content-Length: 64



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      s= sample rtsp presentation
      r = rtsp://foo/twister   /* aggregate URL*/
      m= audio 0 RTP/AVP 0
      r = rtsp://foo/twister/audio
      m=video 0 RTP/AVP 26
      r = rtsp://foo/twister/video

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

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

C->M: SETUP rtsp://foo/twister/video RTSP/1.0 3
      Transport: RTP/AVP;port=8002
      Session: 1234

M->C: RTSP/1.0 200 3 OK
      Session: 1234

C->M: PLAY  rtsp://foo/twister  RTSP/1.0 4
      Range: npt=0-
      Session: 1234

M->C: RTSP/1.0 200 4 OK
      Session: 1234

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

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

C->M: PAUSE rtsp://foo/twister RTSP/1.0 6
      Session: 1234

M->C: RTSP/1.0 200 6 OK
      Session: 1234

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









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

   In the first instance of failure, the client tries to pause one
   stream(in this case video) of the presentation which 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 setup transport parameters.

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

14.3 Live Media Presentation Using Multicast

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

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

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

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

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

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

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

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









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

M->C: RTSP/1.0 200 3 OK

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

14.4 Playing media into an existing session

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

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

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

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

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

14.5 Recording

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

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

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







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M->C: RTSP/1.0 200 90 OK

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

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

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

15 Syntax

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

15.1 Base Syntax

OCTET     = <any 8-bit sequence of data>
CHAR      = <any US-ASCII character (octets 0 - 127)>
UPALPHA   = <any US-ASCII uppercase letter "A".."Z">
LOALPHA   = <any US-ASCII lowercase letter "a".."z">
ALPHA     = UPALPHA | LOALPHA
DIGIT     = <any US-ASCII digit "0".."9">
CTL       = <any US-ASCII control character
             (octets 0 - 31) and DEL (127)>
CR        = <US-ASCII CR, carriage return (13)>
LF        = <US-ASCII LF, linefeed (10)>
SP        = <US-ASCII SP, space (32)>
HT        = <US-ASCII HT, horizontal-tab (9)>
<">       = <US-ASCII double-quote mark (34)>
CRLF      = CR LF
LWS       = [CRLF] 1*( SP | HT )
TEXT      = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@"
          | "," | ";" | ":" | "\" | <">
          | "/" | "[" | "]" | "?" | "="
          | "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">>
quoted-pair = "\" CHAR





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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>

16 Security Considerations

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

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

   Since there is no relation between a transport layer connection and an
   RTSP session, it is possible for a malicious client to issue requests
   with random session identifiers which would affect unsuspecting
   clients. This does not require spoofing of network packet addresses.
   The server SHOULD use a large random session identifier to make this
   attack more difficult.

   Both problems can be be prevented by appropriate authentication.

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

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

A RTSP Protocol State Machines

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

   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
   comprised 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:





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

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

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

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

A.1 Client State Machine

   The client can assume the following states:

   Init:
          SETUP has been sent, waiting for reply.

   Ready:
          SETUP reply received OR after playing, PAUSE reply received.

   Playing:
          PLAY reply received

   Recording:
          RECORD reply received

   In general, the client changes state on receipt of replies to
   requests. Note that some requests are effective at a future time or
   position(such as a PAUSE), and state also changes accordingly. If no
   explicit SETUP is required for the object (for example, it is
   available via a multicast group), state begins at READY. In this case,
   there are only two states, READY and PLAYING.

   The 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.




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

A.2 Server State Machine

   The server can assume the following states:

   Init:
          The initial state, no valid SETUP received.

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

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

   Recording:
          The server is recording media data.

   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, from the client for a
   defined interval, with a default of one minute. If the server is in
   state Ready, it MAY revert to Init if it does not receive an RTSP
   request for an interval of more than one minute. Note that some
   requests(such as PAUSE) may be effective at a future time or position,
   and server state transitions at the appropriate time. The server
   reverts from state Playing or Recording to state Ready at the end of
   the range requested by the client.





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   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, state starts at
   READY, there are only two states READY and PLAYING.

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

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























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B Open Issues

   1.
          Define text/rtsp-parameter MIME type.
   2.
          Reverse: Scale: -1, with reversed start times, or both?
   3.
          HS believes that RTSP should only control individual media
          objects rather than aggregates. This avoids disconnects between
          presentation descriptions and streams and avoids having to deal
          separately with single-host and multi-host case. Cost: several
          PLAY/PAUSE/RECORD in one packet, one for each stream.
   4.
          Allow changing of transport for a stream that's playing? May
          not be a great idea since the same can be accomplished by tear
          down and re-setup. Exception: near-video-on-demand, where the
          server changes the address in a PLAY response. Servers may not
          be able to reliably send TEARDOWN to clients and the client
          wouldn't know why this happened in any event.
   5.
          How does the server get back to the client unless a persistent
          connection is used? Probably cannot, in general.
   6.
          Server issues TEARDOWN and other 'event' notifications to
          client? This raises the problem discussed in the previous open
          issue, but is useful for the client if the data stream contains
          no end indication.






















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C Changes

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

     * Allowing the Transport header to direct media streams to unicast
       and multicast addresses, with an appropriate warning about
       denial-of-service attacks.
     * Add mode parameter to Transport header to allow RECORD or PLAY.
     * The Embedded binary data section was modified to clearly indicate
       the stream the data corresponds to, and a reference to the
       Transport header was added.
     * The Transport header format has been changed to use a more general
       means to specify data channel and application level protocol. It
       also conveys the port to be used at the server for RTCP messages,
       and the start sequence number that will be used in the RTP
       packets.
     * The use of the Session: header has been enhanced. Requests for
       multiple URLs may be sent in a single session.
     * There is a distinction between aggregate(presentation) URLs and
       stream URLs. Error codes have been added to reflect the fact that
       some methods may be allowed only on a particular type of URL.
     * Example showing the use of aggregate/presentation control using a
       single RTSP session has been added.
     * Support for the PEP(Protocol Extension Protocol) headers has been
       added.
     * Server-Client DESCRIBE messages have been renamed to ANNOUNCE for
       better clarity and differentiation.

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

D Author Addresses

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









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   Anup Rao
   Netscape Communications Corp.
   501 E. Middlefield Road
   Mountain View, CA 94043
   USA
   electronic mail: anup@netscape.com

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

E Acknowledgements

   This draft is based on the functionality of the original RTSP draft
   submitted in October 96. It also borrows format and descriptions from
   HTTP/1.1.

   This document has benefited greatly from the comments of all those
   participating in the MMUSIC-WG. In addition to those already
   mentioned, the following individuals have contributed to this
   specification:

            Rahul Agarwal             Eduardo F. Llach
            Bruce Butterfield         Rob McCool
             Steve Casner             David Oran
            Martin Dunsmuir           Sujal Patel
            Eric Fleischman
            Mark Handley              Igor Plotnikov
            Peter Haight              Pinaki Shah
            Brad Hefta-Gaub           Jeff Smith
            John K. Ho                Alexander Sokolsky
            Ruth Lang                 Dale Stammen
            Stephanie Leif            John Francis Stracke













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References

   1
          H. Schulzrinne, ``RTP profile for audio and video conferences
          with minimal control,'' RFC 1890, Internet Engineering Task
          Force, Jan. 1996.
   2
          D. Kristol and L. Montulli, ``HTTP state management
          mechanism,'' RFC 2109, Internet Engineering Task Force, Feb.
          1997.
   3
          F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
          ``Internationalization of the hypertext markup language,'' RFC
          2070, Internet Engineering Task Force, Jan. 1997.
   4
          S. Bradner, ``Key words for use in RFCs to indicate requirement
          levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997.
   5
          R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T.
          Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC
          2068, Internet Engineering Task Force, Jan. 1997.
   6
          M. Handley, ``SDP: Session description protocol,'' Internet
          Draft, Internet Engineering Task Force, Nov. 1996.
          Work in progress.
   7
          A. Freier, P. Karlton, and P. Kocher, ``The TLS protocol,''
          Internet Draft, Internet Engineering Task Force, Dec. 1996.
          Work in progress.
   8
          J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and
          E. L. Stewart, ``An extension to HTTP: digest access
          authentication,'' RFC 2069, Internet Engineering Task Force,
          Jan. 1997.
   9
          J. Postel, ``User datagram protocol,'' STD 6, RFC 768, Internet
          Engineering Task Force, Aug. 1980.
   10
          R. Hinden and C. Partridge, ``Version 2 of the reliable data
          protocol (RDP),'' RFC 1151, Internet Engineering Task Force,
          Apr. 1990.
   11
          J. Postel, ``Transmission control protocol,'' STD 7, RFC 793,
          Internet Engineering Task Force, Sept. 1981.
   12
          M. Handley, H. Schulzrinne, and E. Schooler, ``SIP: Session
          initiation protocol,'' Internet Draft, Internet Engineering
          Task Force, Dec. 1996.
          Work in progress.
   13
          P. McMahon, ``GSS-API authentication method for SOCKS version
          5,'' RFC 1961, Internet Engineering Task Force, June 1996.
   14
          D. Crocker, ``Augmented BNF for syntax specifications: ABNF,''
          Internet Draft, Internet Engineering Task Force, Oct. 1996.
          Work in progress.
   15
          R. Elz, ``A compact representation of IPv6 addresses,'' RFC
          1924, Internet Engineering Task Force, Apr. 1996.
   16
          T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform
          resource locators (URL),'' RFC 1738, Internet Engineering Task
          Force, Dec. 1994.
   17
          International Telecommunication Union, ``Visual telephone
          systems and equipment for local area networks which provide a
          non-guaranteed quality of service,'' Recommendation H.323,
          Telecommunication Standardization Sector of ITU, Geneva,
          Switzerland, May 1996.
   18
          ISO/IEC, ``Information technology - generic coding of moving
          pictures and associated audio informaiton - part 6: extension
          for digital storage media and control,'' Draft International
          Standard ISO 13818-6, International Organization for
          Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
          Nov. 1995.
   19
          H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
          ``RTP: a transport protocol for real-time applications,'' RFC
          1889, Internet Engineering Task Force, Jan. 1996.
   20
          J. Miller, P. Resnick, and D. Singer, ``Rating Services and
          Rating Systems(and Their Machine Readable Descriptions), ''
          REC-PICS-services-961031, Worldwide Web Consortium, Oct. 1996.
   21
          D. Connolly, R. Khare, H.F. Nielsen, ``PEP - an Extension
          Mechanism for HTTP", Internet draft, work-in-progress. W3C
          Draft WD-http-pep-970714
          http://www.w3.org/TR/WD-http-pep-970714, July, 1996.



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