Real Time Streaming Protocol (RTSP) (original) (raw)

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Network Working Group H. Schulzrinne Request for Comments: 2326 Columbia U. Category: Standards Track A. Rao Netscape R. Lanphier RealNetworks April 1998

              Real Time Streaming Protocol (RTSP)

Status of this Memo

This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.

Copyright Notice

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

Abstract

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

Table of Contents

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    + 3.2 RTSP URL .............................................. [14](#page-14)
    + 3.3 Conference Identifiers ................................ [16](#page-16)
    + 3.4 Session Identifiers ................................... [16](#page-16)
    + 3.5 SMPTE Relative Timestamps ............................. [16](#page-16)
    + 3.6 Normal Play Time ...................................... [17](#page-17)
    + 3.7 Absolute Time ......................................... [18](#page-18)
    + 3.8 Option Tags ........................................... [18](#page-18)
         o 3.8.1 Registering New Option Tags with IANA .......... [18](#page-18)

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         o 11.3.4 453 Not Enough Bandwidth ...................... [42](#page-42)
         o 11.3.5 454 Session Not Found ......................... [42](#page-42)
         o 11.3.6 455 Method Not Valid in This State ............ [42](#page-42)
         o 11.3.7 456 Header Field Not Valid for Resource ....... [42](#page-42)
         o 11.3.8 457 Invalid Range ............................. [43](#page-43)
         o 11.3.9 458 Parameter Is Read-Only .................... [43](#page-43)
         o 11.3.10 459 Aggregate Operation Not Allowed .......... [43](#page-43)
         o 11.3.11 460 Only Aggregate Operation Allowed ......... [43](#page-43)
         o 11.3.12 461 Unsupported Transport .................... [43](#page-43)
         o 11.3.13 462 Destination Unreachable .................. [43](#page-43)
         o 11.3.14 551 Option not supported ..................... [43](#page-43)

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    + 12.37 Session ............................................. [57](#page-57)
    + 12.38 Timestamp ........................................... [58](#page-58)
    + 12.39 Transport ........................................... [58](#page-58)
    + 12.40 Unsupported ......................................... [62](#page-62)
    + 12.41 User-Agent .......................................... [62](#page-62)
    + 12.42 Vary ................................................ [62](#page-62)
    + 12.43 Via ................................................. [62](#page-62)
    + 12.44 WWW-Authenticate .................................... [62](#page-62)

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1 Introduction

1.1 Purpose

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

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

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

The streams controlled by RTSP may use RTP [1], but the operation of RTSP does not depend on the transport mechanism used to carry continuous media. The protocol is intentionally similar in syntax and operation to HTTP/1.1 [2] so that extension mechanisms to HTTP can in most cases also be added to RTSP. However, RTSP differs in a number of important aspects from HTTP:

 * RTSP introduces a number of new methods and has a different
   protocol identifier.
 * An RTSP server needs to maintain state by default in almost all
   cases, as opposed to the stateless nature of HTTP.
 * Both an RTSP server and client can issue requests.
 * Data is carried out-of-band by a different protocol. (There is an
   exception to this.)
 * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
   consistent with current HTML internationalization efforts [3].
 * The Request-URI always contains the absolute URI. Because of
   backward compatibility with a historical blunder, HTTP/1.1 [2]
   carries only the absolute path in the request and puts the host
   name in a separate header field.

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

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The protocol supports the following operations:

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

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

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

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

1.2 Requirements

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

1.3 Terminology

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

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

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

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

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

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

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

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

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

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

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

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Media server indirection: Redirection of a media client to a different media server.

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

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

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

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

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

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

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

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

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      stream with TEARDOWN.

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

1.4 Protocol Properties

RTSP has the following properties:

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

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

Secure: RTSP re-uses web security mechanisms. All HTTP authentication mechanisms such as basic (RFC 2068 [2, Section 11.1]) and digest authentication (RFC 2069 [8]) are directly applicable. One may also reuse transport or network layer security mechanisms.

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

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

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

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

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

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

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

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

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

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

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

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

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 control mechanism.

1.5 Extending RTSP

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

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

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

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

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

 * Existing methods can be extended with new parameters, as long as
   these parameters can be safely ignored by the recipient. (This is
   equivalent to adding new parameters to an HTML tag.) If the
   client needs negative acknowledgement when a method extension is
   not supported, a tag corresponding to the extension may be added
   in the Require: field (see [Section 12.32](#section-12.32)).
 * New methods can be added. If the recipient of the message does
   not understand the request, it responds with error code 501 (Not
   implemented) and the sender should not attempt to use this method
   again. A client may also use the OPTIONS method to inquire about
   methods supported by the server. The server SHOULD list the
   methods it supports using the Public response header.
 * A new version of the protocol can be defined, allowing almost all
   aspects (except the position of the protocol version number) to
   change.

1.6 Overall Operation

Each presentation and media stream may be identified by an RTSP URL. The overall presentation and the properties of the media the presentation is made up of are defined by a presentation description file, the format of which is outside the scope of this specification. The presentation description file may be obtained by the client using

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HTTP or other means such as email and may not necessarily be stored on the media server.

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

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

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

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

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

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

1.7 RTSP States

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

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server. Also, during its lifetime, a single media stream may be controlled by RTSP requests issued sequentially on different TCP connections. Therefore, the server needs to maintain "session state" to be able to correlate RTSP requests with a stream. The state transitions are described in Section A.

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

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

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

PAUSE: Temporarily halts a stream without freeing server resources.

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

      RTSP methods that contribute to state use the Session header
      field ([Section 12.37](#section-12.37)) to identify the RTSP session whose state
      is being manipulated. The server generates session identifiers
      in response to SETUP requests ([Section 10.4](#section-10.4)).

1.8 Relationship with Other Protocols

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

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

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request has been acknowledged.

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

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

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

2 Notational Conventions

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

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

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

3 Protocol Parameters

3.1 RTSP Version

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

3.2 RTSP URL

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

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

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             of [RFC 1123](./rfc1123) \cite{[rfc1123](./rfc1123)}>

port = *DIGIT

abs_path is defined in [H3.2.1].

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

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

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

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

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

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

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

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

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

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

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

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

3.3 Conference Identifiers

Conference identifiers are opaque to RTSP and are encoded using standard URI encoding methods (i.e., LWS is escaped with %). They can contain any octet value. The conference identifier MUST be globally unique. For H.323, the conferenceID value is to be used.

conference-id = 1*xchar

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

3.4 Session Identifiers

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

 session-id   =   1*( ALPHA | DIGIT | safe )

3.5 SMPTE Relative Timestamps

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

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smpte-range = smpte-type "=" smpte-time "-" [ smpte-time ] smpte-type = "smpte" | "smpte-30-drop" | "smpte-25" ; other timecodes may be added smpte-time = 12DIGIT ":" 12DIGIT ":" 12DIGIT [ ":" 12DIGIT ] [ "." 1*2DIGIT ]

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

3.6 Normal Play Time

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

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

NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the viewer associates with a program. It is often digitally displayed on a VCR. NPT advances normally when in normal play mode (scale = 1), advances at a faster rate when in fast scan forward (high positive scale ratio), decrements when in scan reverse (high negative scale ratio) and is fixed in pause mode. NPT is (logically) equivalent to SMPTE time codes." [5]

npt-range = ( npt-time "-" [ npt-time ] ) | ( "-" npt-time ) npt-time = "now" | npt-sec | npt-hhmmss npt-sec = 1DIGIT [ "." DIGIT ] npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." DIGIT ] npt-hh = 1DIGIT ; any positive number npt-mm = 12DIGIT ; 0-59 npt-ss = 12DIGIT ; 0-59

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

 The syntax conforms to ISO 8601. The npt-sec notation is optimized
 for automatic generation, the ntp-hhmmss notation for consumption
 by human readers. The "now" constant allows clients to request to

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 receive the live feed rather than the stored or time-delayed
 version. This is needed since neither absolute time nor zero time
 are appropriate for this case.

3.7 Absolute Time

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

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

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

 19961108T143720.25Z

3.8 Option Tags

Option tags are unique identifiers used to designate new options in RTSP. These tags are used in Require (Section 12.32) and Proxy- Require (Section 12.27) header fields.

Syntax:

 option-tag   =   1*xchar

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

3.8.1 Registering New Option Tags with IANA

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

 * Name and description of option. The name may be of any length,
   but SHOULD be no more than twenty characters long. The name MUST
   not contain any spaces, control characters or periods.
 * Indication of who has change control over the option (for
   example, IETF, ISO, ITU-T, other international standardization
   bodies, a consortium or a particular company or group of
   companies);

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 * A reference to a further description, if available, for example
   (in order of preference) an RFC, a published paper, a patent
   filing, a technical report, documented source code or a computer
   manual;
 * For proprietary options, contact information (postal and email
   address);

4 RTSP Message

RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2279 [21]). Lines are terminated by CRLF, but receivers should be prepared to also interpret CR and LF by themselves as line terminators.

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

 The 10646 character set avoids tricky character set switching, but
 is invisible to the application as long as US-ASCII is being used.
 This is also the encoding used for RTCP. ISO 8859-1 translates
 directly into Unicode with a high-order octet of zero. ISO 8859-1
 characters with the most-significant bit set are represented as
 1100001x 10xxxxxx. (See [RFC 2279](./rfc2279) [21])

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

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

4.1 Message Types

See [H4.1]

4.2 Message Headers

See [H4.2]

4.3 Message Body

See [H4.3]

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4.4 Message Length

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

  1. Any response message which MUST NOT include a message body
(such as the 1xx, 204, and 304 responses) is always terminated
by the first empty line after the header fields, regardless of
the entity-header fields present in the message. (Note: An
empty line consists of only CRLF.)
  2. If a Content-Length header field ([section 12.14](#section-12.14)) is present,
its value in bytes represents the length of the message-body.
If this header field is not present, a value of zero is
assumed.
  3. By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a
response.)

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

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

5 General Header Fields

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

  general-header     =     Cache-Control     ; [Section 12.8](#section-12.8)
                     |     Connection        ; [Section 12.10](#section-12.10)
                     |     Date              ; [Section 12.18](#section-12.18)
                     |     Via               ; [Section 12.43](#section-12.43)

6 Request

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

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

6.1 Request Line

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

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

extension-method = token

Request-URI = "*" | absolute_URI

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

6.2 Request Header Fields

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

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

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

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

 OPTIONS * RTSP/1.0

7 Response

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

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

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

7.1 Status-Line

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

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

7.1.1 Status Code and Reason Phrase

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

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

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

The individual values of the numeric status codes defined for RTSP/1.0, and an example set of corresponding Reason-Phrase's, are presented below. The reason phrases listed here are only recommended

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Status-Code = "100" ; Continue | "200" ; OK | "201" ; Created | "250" ; Low on Storage Space | "300" ; Multiple Choices | "301" ; Moved Permanently | "302" ; Moved Temporarily | "303" ; See Other | "304" ; Not Modified | "305" ; Use Proxy | "400" ; Bad Request | "401" ; Unauthorized | "402" ; Payment Required | "403" ; Forbidden | "404" ; Not Found | "405" ; Method Not Allowed | "406" ; Not Acceptable | "407" ; Proxy Authentication Required | "408" ; Request Time-out | "410" ; Gone | "411" ; Length Required | "412" ; Precondition Failed | "413" ; Request Entity Too Large | "414" ; Request-URI Too Large | "415" ; Unsupported Media Type | "451" ; Parameter Not Understood | "452" ; Conference Not Found | "453" ; Not Enough Bandwidth | "454" ; Session Not Found | "455" ; Method Not Valid in This State | "456" ; Header Field Not Valid for Resource | "457" ; Invalid Range | "458" ; Parameter Is Read-Only | "459" ; Aggregate operation not allowed | "460" ; Only aggregate operation allowed | "461" ; Unsupported transport | "462" ; Destination unreachable | "500" ; Internal Server Error | "501" ; Not Implemented | "502" ; Bad Gateway | "503" ; Service Unavailable | "504" ; Gateway Time-out | "505" ; RTSP Version not supported | "551" ; Option not supported | extension-code

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extension-code = 3DIGIT

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

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

Code reason

100 Continue all

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

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

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400 Bad Request all 401 Unauthorized all 402 Payment Required all 403 Forbidden all 404 Not Found all 405 Method Not Allowed all 406 Not Acceptable all 407 Proxy Authentication Required all 408 Request Timeout all 410 Gone all 411 Length Required all 412 Precondition Failed DESCRIBE, SETUP 413 Request Entity Too Large all 414 Request-URI Too Long all 415 Unsupported Media Type all 451 Invalid parameter SETUP 452 Illegal Conference Identifier SETUP 453 Not Enough Bandwidth SETUP 454 Session Not Found all 455 Method Not Valid In This State all 456 Header Field Not Valid all 457 Invalid Range PLAY 458 Parameter Is Read-Only SET_PARAMETER 459 Aggregate Operation Not Allowed all 460 Only Aggregate Operation Allowed all 461 Unsupported Transport all 462 Destination Unreachable all

500 Internal Server Error all 501 Not Implemented all 502 Bad Gateway all 503 Service Unavailable all 504 Gateway Timeout all 505 RTSP Version Not Supported all 551 Option not support all

  Table 1: Status codes and their usage with RTSP methods

7.1.2 Response Header Fields

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

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

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

8 Entity

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

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

8.1 Entity Header Fields

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

 entity-header       =    Allow               ; [Section 12.4](#section-12.4)
                     |    Content-Base        ; [Section 12.11](#section-12.11)
                     |    Content-Encoding    ; [Section 12.12](#section-12.12)
                     |    Content-Language    ; [Section 12.13](#section-12.13)
                     |    Content-Length      ; [Section 12.14](#section-12.14)
                     |    Content-Location    ; [Section 12.15](#section-12.15)
                     |    Content-Type        ; [Section 12.16](#section-12.16)
                     |    Expires             ; [Section 12.19](#section-12.19)
                     |    Last-Modified       ; [Section 12.24](#section-12.24)
                     |    extension-header
 extension-header    =    message-header

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

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8.2 Entity Body

See [H7.2]

9 Connections

RTSP requests can be transmitted in several different ways:

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

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

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

9.1 Pipelining

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

9.2 Reliability and Acknowledgements

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

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

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

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 transport stack will not deliver the application-layer
 retransmission before the first attempt has reached the receiver.
 If the packet loss is caused by congestion, multiple
 retransmissions at different layers will exacerbate the congestion.

 If RTSP is used over a small-RTT LAN, standard procedures for
 optimizing initial TCP round trip estimates, such as those used in
 T/TCP ([RFC 1644](./rfc1644)) [22], can be beneficial.

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

Each request carries a sequence number in the CSeq header (Section 12.17), which is incremented by one for each distinct request transmitted. If a request is repeated because of lack of acknowledgement, the request MUST carry the original sequence number (i.e., the sequence number is not incremented).

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

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

10 Method Definitions

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

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

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

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

10.1 OPTIONS

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

Example:

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

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

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

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10.2 DESCRIBE

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

Example:

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

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

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

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

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

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

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

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

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

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

10.3 ANNOUNCE

The ANNOUNCE method serves two purposes:

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

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

Example:

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

       v=0
       o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4

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

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

10.4 SETUP

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

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

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

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

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

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

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

10.5 PLAY

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

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

 This allows precise editing.

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

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

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

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

See the description of the PAUSE request for further examples.

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

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

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

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

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

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

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

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

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

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

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

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10.6 PAUSE

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

Example:

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

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

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

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

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

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play request and discards the second play request.

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

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

10.7 TEARDOWN

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

Example: C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 892 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 892

10.8 GET_PARAMETER

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

Example:

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

       packets_received
       jitter

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

       packets_received: 10
       jitter: 0.3838

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

10.9 SET_PARAMETER

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

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

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

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

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

Example:

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

       barparam: barstuff

 S->C: RTSP/1.0 451 Invalid Parameter

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       CSeq: 421
       Content-length: 10
       Content-type: text/parameters

       barparam

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

10.10 REDIRECT

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

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

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

10.11 RECORD

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

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

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

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

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

10.12 Embedded (Interleaved) Binary Data

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

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

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

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

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

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

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

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       Session: 12345678

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

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

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

11 Status Code Definitions

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

11.1 Success 2xx

11.1.1 250 Low on Storage Space

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

11.2 Redirection 3xx

See [H10.3].

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

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11.3 Client Error 4xx

11.3.1 405 Method Not Allowed

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

11.3.2 451 Parameter Not Understood

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

11.3.3 452 Conference Not Found

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

11.3.4 453 Not Enough Bandwidth

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

11.3.5 454 Session Not Found

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

11.3.6 455 Method Not Valid in This State

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

11.3.7 456 Header Field Not Valid for Resource

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

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11.3.8 457 Invalid Range

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

11.3.9 458 Parameter Is Read-Only

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

11.3.10 459 Aggregate Operation Not Allowed

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

11.3.11 460 Only Aggregate Operation Allowed

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

11.3.12 461 Unsupported Transport

The Transport field did not contain a supported transport specification.

11.3.13 462 Destination Unreachable

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

11.3.14 551 Option not supported

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

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12 Header Field Definitions

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

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

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

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Overview of RTSP header fields

12.1 Accept

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

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

See [H14.1] for syntax.

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

12.2 Accept-Encoding

 See [H14.3]

12.3 Accept-Language

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

12.4 Allow

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

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

12.5 Authorization

 See [H14.8]

12.6 Bandwidth

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

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Bandwidth = "Bandwidth" ":" 1*DIGIT

Example: Bandwidth: 4000

12.7 Blocksize

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

12.8 Cache-Control

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

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

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

Cache-Control = "Cache-Control" ":" 1#cache-directive cache-directive = cache-request-directive | cache-response-directive cache-request-directive = "no-cache" | "max-stale" | "min-fresh" | "only-if-cached" | cache-extension cache-response-directive = "public" | "private" | "no-cache" | "no-transform" | "must-revalidate"

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                        |   "proxy-revalidate"
                        |   "max-age" "=" delta-seconds
                        |   cache-extension

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

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

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

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

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

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

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

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

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

12.9 Conference

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

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

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

12.10 Connection

See [H14.10]

12.11 Content-Base

See [H14.11]

12.12 Content-Encoding

See [H14.12]

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12.13 Content-Language

See [H14.13]

12.14 Content-Length

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

12.15 Content-Location

See [H14.15]

12.16 Content-Type

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

12.17 CSeq

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

12.18 Date

See [H14.19].

12.19 Expires

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

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

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

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

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

Expires = "Expires" ":" HTTP-date

An example of its use is

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

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

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

The presence of an Expires header field with a date value of some time in the future on a media stream that otherwise would by default be non-cacheable indicates that the media stream is cacheable, unless indicated otherwise by a Cache-Control header field (Section 12.8).

12.20 From

See [H14.22].

12.21 Host

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

12.22 If-Match

See [H14.25].

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

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

12.23 If-Modified-Since

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

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

An example of the field is:

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

12.24 Last-Modified

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

12.25 Location

See [H14.30].

12.26 Proxy-Authenticate

See [H14.33].

12.27 Proxy-Require

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

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should treat this field identically to the Require field.

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

12.28 Public

See [H14.35].

12.29 Range

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

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

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

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

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

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12.30 Referer

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

12.31 Retry-After

See [H14.38].

12.32 Require

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

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

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

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

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

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

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

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

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Proxies and other intermediary devices SHOULD ignore features that are not understood in this field. If a particular extension requires that intermediate devices support it, the extension should be tagged in the Proxy-Require field instead (see Section 12.27).

12.33 RTP-Info

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

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

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

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

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

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

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Syntax:

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

Example:

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

12.34 Scale

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

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

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

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

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

Example of playing in reverse at 3.5 times normal rate:

 Scale: -3.5

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12.35 Speed

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

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

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

Example: Speed: 2.5

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

12.36 Server

See [H14.39]

12.37 Session

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

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

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

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

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

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

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

12.38 Timestamp

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

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

12.39 Transport

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

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

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

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

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

The syntax for the transport specifier is

   transport/profile/lower-transport.

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

Below are the configuration parameters associated with transport:

General parameters:

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

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

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

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

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

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

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

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

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

Multicast specific:

ttl: multicast time-to-live

RTP Specific:

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

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

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

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

Transport = "Transport" ":" 1#transport-spec transport-spec = transport-protocol/profile[/lower-transport] parameter transport-protocol = "RTP" profile = "AVP" lower-transport = "TCP" | "UDP" parameter = ( "unicast" | "multicast" ) | ";" "destination" [ "=" address ] | ";" "interleaved" "=" channel [ "-" channel ] | ";" "append" | ";" "ttl" "=" ttl | ";" "layers" "=" 1DIGIT | ";" "port" "=" port [ "-" port ] | ";" "client_port" "=" port [ "-" port ] | ";" "server_port" "=" port [ "-" port ] | ";" "ssrc" "=" ssrc | ";" "mode" = <"> 1#mode <"> ttl = 13(DIGIT) port = 15(DIGIT) ssrc = 88(HEX) channel = 13(DIGIT) address = host mode = <"> *Method <"> | Method

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

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

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12.40 Unsupported

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

See Section 12.32 for a usage example.

12.41 User-Agent

See [H14.42]

12.42 Vary

See [H14.43]

12.43 Via

See [H14.44].

12.44 WWW-Authentica

See [H14.46].

13 Caching

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

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

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described in Section 12.8. A cache MUST answer any DESCRIBE requests if it is currently serving the stream to the requestor, as it is possible that low-level details of the stream description may have changed on the origin-server.

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

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

14 Examples

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

14.1 Media on Demand (Unicast)

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

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

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

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

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

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

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

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

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

 C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
       CSeq: 2
       Session: 23456789
       Range: smpte=0:10:00-

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

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

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

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       Range: smpte=0:10:00-0:20:00
       RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
         seq=876655;rtptime=1032181

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

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

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

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

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

14.2 Streaming of a Container file

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

14.3 Single Stream Container Files

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

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

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      CSeq: 1

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

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

C->S  SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
      Transport: RTP/AVP/UDP;unicast;
                 client_port=6970-6971;mode=play
      CSeq: 2

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

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

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

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

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

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

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In the worst case, servers should send back:

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

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

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

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

14.4 Live Media Presentation Using Multicast

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

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

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

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

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

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

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

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

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       Transport: RTP/AVP;multicast

 M->C: RTSP/1.0 200 OK
       CSeq: 2
       Transport: RTP/AVP;multicast;destination=224.2.0.1;
                  port=3456-3457;ttl=16
       Session: 0456804596

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

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

14.5 Playing media into an existing session

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

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

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

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

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

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

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

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

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

14.6 Recording

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

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

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

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

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

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

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

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

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

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

15 Syntax

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

15.1 Base Syntax

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

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

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

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

token = 1* quoted-string = ( <"> *(qdtext) <"> ) qdtext = <any TEXT except <">> quoted-pair = "" CHAR

message-header = field-name ":" [ field-value ] CRLF field-name = token field-value = *( field-content | LWS ) field-content = <the OCTETs making up the field-value and consisting of either *TEXT or combinations of token, tspecials, and quoted-string>

safe = "$" | "-" | "_" | "." | "+" extra = "!" | "*" | "$'$" | "(" | ")" | ","

hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" | "a" | "b" | "c" | "d" | "e" | "f" escape = "%" hex hex reserved = ";" | "/" | "?" | ":" | "@" | "&" | "="

unreserved = alpha | digit | safe | extra xchar = unreserved | reserved | escape

16 Security Considerations

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

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

Abuse of Server Log Information: RTSP and HTTP servers will presumably have similar logging mechanisms, and thus should be equally guarded in protecting the contents of those logs, thus protecting the privacy of the

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      users of the servers. See [H15.3] for HTTP server
      recommendations regarding server logs.

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

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

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

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

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

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

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

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The following are added considerations for RTSP implementations.

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

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

Authentication: Servers SHOULD implement both basic and digest [8] authentication. In environments requiring tighter security for the control messages, the RTSP control stream may be encrypted.

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

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

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Appendix A: RTSP Protocol State Machines

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

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

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

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

 This example does not imply a standard way to represent streams in
 URLs or a relation to the filesystem. See [Section 3.2](#section-3.2).

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

A.1 Client State Machine

The client can assume the following states:

Init: SETUP has been sent, waiting for reply.

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

Playing: PLAY reply received

Recording: RECORD reply received

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

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

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

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

A.2 Server State Machine

The server can assume the following states:

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

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

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

Recording: The server is recording media data.

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

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

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

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

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

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Appendix B: Interaction with RTP

RTSP allows media clients to control selected, non-contiguous sections of media presentations, rendering those streams with an RTP media layer[24]. The media layer rendering the RTP stream should not be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be continuous and monotonic across jumps of NPT.

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

 We cannot assume that the RTSP client can communicate with the RTP
 media agent, as the two may be independent processes. If the RTP
 timestamp shows the same gap as the NPT, the media agent will
 assume that there is a pause in the presentation. If the jump in
 NPT is large enough, the RTP timestamp may roll over and the media
 agent may believe later packets to be duplicates of packets just
 played out.

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

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

For scaling (see Section 12.34), RTP timestamps should correspond to the playback timing. For example, when playing video recorded at 30 frames/second at a scale of two and speed (Section 12.35) of one, the server would drop every second frame to maintain and deliver video packets with the normal timestamp spacing of 3,000 per frame, but NPT would increase by 1/15 second for each video frame.

The client can maintain a correct display of NPT by noting the RTP timestamp value of the first packet arriving after repositioning. The sequence parameter of the RTP-Info (Section 12.33) header provides the first sequence number of the next segment.

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Appendix C: Use of SDP for RTSP Session Descriptions

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

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

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

This appendix describes how an SDP file, retrieved, for example, through HTTP, determines the operation of an RTSP session. It also describes how a client should interpret SDP content returned in reply to a DESCRIBE request. SDP provides no mechanism by which a client can distinguish, without human guidance, between several media streams to be rendered simultaneously and a set of alternatives (e.g., two audio streams spoken in different languages).

C.1 Definitions

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

C.1.1 Control URL

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

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

This attribute may contain either relative and absolute URLs, following the rules and conventions set out in RFC 1808 [25]. Implementations should look for a base URL in the following order:

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

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

C.1.2 Media streams

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

Example: m=audio 0 RTP/AVP 31

C.1.3 Payload type(s)

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

C.1.4 Format-specific parameters

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

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C.1.5 Range of presentation

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

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

C.1.6 Time of availability

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

C.1.7 Connection Information

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

C.1.8 Entity Tag

The optional "a=etag" attribute identifies a version of the session description. It is opaque to the client. SETUP requests may include this identifier in the If-Match field (see section 12.22) to only allow session establishment if this attribute value still corresponds to that of the current description. The attribute value is opaque and may contain any character allowed within SDP attribute values.

Example: a=etag:158bb3e7c7fd62ce67f12b533f06b83a

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

C.2 Aggregate Control Not Available

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

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

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

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

C.3 Aggregate Control Available

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

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

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

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

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Appendix D: Minimal RTSP implementation

D.1 Client

A client implementation MUST be able to do the following :

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

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

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

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

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D.1.1 Basic Playback

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

D.1.2 Authentication-enabled

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

D.2 Server

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

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

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

 * Implement RTP/AVP/UDP as a valid transport.
 * Inclusion of the Server header.
 * Implement the DESCRIBE method.
 * Generate SDP session descriptions as defined in [Appendix C](#appendix-C)

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

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D.2.1 Basic Playback

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

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

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

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

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

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

D.2.2 Authentication-enabled

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

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

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Appendix E: Authors' Addresses

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

EMail: schulzrinne@cs.columbia.edu

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

EMail: anup@netscape.com

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

EMail: robla@real.com

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Appendix F: Acknowledgements

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

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

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

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References

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

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

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

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

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

6 Handley, M., and V. Jacobson, "SDP: Session Description Protocol", RFC 2327, April 1998.

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

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

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

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

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

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

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RFC 2326 Real Time Streaming Protocol April 1998

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

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

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

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

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

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

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

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

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

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

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

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

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Full Copyright Statement

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