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rtsp协议相关之-rfc2326

王朝other·作者佚名  2006-01-09
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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

* 1 Introduction ................................................. 5

+ 1.1 Purpose ............................................... 5

+ 1.2 Requirements .......................................... 6

+ 1.3 Terminology ........................................... 6

+ 1.4 Protocol Properties ................................... 9

+ 1.5 Extending RTSP ........................................ 11

+ 1.6 Overall Operation ..................................... 11

+ 1.7 RTSP States ........................................... 12

+ 1.8 Relationship with Other Protocols ..................... 13

* 2 Notational Conventions ....................................... 14

* 3 Protocol Parameters .......................................... 14

+ 3.1 RTSP Version .......................................... 14

Schulzrinne, et. al. Standards Track [Page 1]

RFC 2326 Real Time Streaming Protocol April 1998

+ 3.2 RTSP URL .............................................. 14

+ 3.3 Conference Identifiers ................................ 16

+ 3.4 Session Identifiers ................................... 16

+ 3.5 SMPTE Relative Timestamps ............................. 16

+ 3.6 Normal Play Time ...................................... 17

+ 3.7 Absolute Time ......................................... 18

+ 3.8 Option Tags ........................................... 18

o 3.8.1 Registering New Option Tags with IANA .......... 18

* 4 RTSP Message ................................................. 19

+ 4.1 Message Types ......................................... 19

+ 4.2 Message Headers ....................................... 19

+ 4.3 Message Body .......................................... 19

+ 4.4 Message Length ........................................ 20

* 5 General Header Fields ........................................ 20

* 6 Request ...................................................... 20

+ 6.1 Request Line .......................................... 21

+ 6.2 Request Header Fields ................................. 21

* 7 Response ..................................................... 22

+ 7.1 Status-Line ........................................... 22

o 7.1.1 Status Code and Reason Phrase .................. 22

o 7.1.2 Response Header Fields ......................... 26

* 8 Entity ....................................................... 27

+ 8.1 Entity Header Fields .................................. 27

+ 8.2 Entity Body ........................................... 28

* 9 Connections .................................................. 28

+ 9.1 Pipelining ............................................ 28

+ 9.2 Reliability and Acknowledgements ...................... 28

* 10 Method Definitions .......................................... 29

+ 10.1 OPTIONS .............................................. 30

+ 10.2 DESCRIBE ............................................. 31

+ 10.3 ANNOUNCE ............................................. 32

+ 10.4 SETUP ................................................ 33

+ 10.5 PLAY ................................................. 34

+ 10.6 PAUSE ................................................ 36

+ 10.7 TEARDOWN ............................................. 37

+ 10.8 GET_PARAMETER ........................................ 37

+ 10.9 SET_PARAMETER ........................................ 38

+ 10.10 REDIRECT ............................................ 39

+ 10.11 RECORD .............................................. 39

+ 10.12 Embedded (Interleaved) Binary Data .................. 40

* 11 Status Code Definitions ..................................... 41

+ 11.1 Success 2xx .......................................... 41

o 11.1.1 250 Low on Storage Space ...................... 41

+ 11.2 Redirection 3xx ...................................... 41

+ 11.3 Client Error 4xx ..................................... 42

o 11.3.1 405 Method Not Allowed ........................ 42

o 11.3.2 451 Parameter Not Understood .................. 42

o 11.3.3 452 Conference Not Found ...................... 42

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RFC 2326 Real Time Streaming Protocol April 1998

o 11.3.4 453 Not Enough Bandwidth ...................... 42

o 11.3.5 454 Session Not Found ......................... 42

o 11.3.6 455 Method Not Valid in This State ............ 42

o 11.3.7 456 Header Field Not Valid for Resource ....... 42

o 11.3.8 457 Invalid Range ............................. 43

o 11.3.9 458 Parameter Is Read-Only .................... 43

o 11.3.10 459 Aggregate Operation Not Allowed .......... 43

o 11.3.11 460 Only Aggregate Operation Allowed ......... 43

o 11.3.12 461 Unsupported Transport .................... 43

o 11.3.13 462 Destination Unreachable .................. 43

o 11.3.14 551 Option not supported ..................... 43

* 12 Header Field Definitions .................................... 44

+ 12.1 Accept ............................................... 46

+ 12.2 Accept-Encoding ...................................... 46

+ 12.3 Accept-Language ...................................... 46

+ 12.4 Allow ................................................ 46

+ 12.5 Authorization ........................................ 46

+ 12.6 Bandwidth ............................................ 46

+ 12.7 Blocksize ............................................ 47

+ 12.8 Cache-Control ........................................ 47

+ 12.9 Conference ........................................... 49

+ 12.10 Connection .......................................... 49

+ 12.11 Content-Base ........................................ 49

+ 12.12 Content-Encoding .................................... 49

+ 12.13 Content-Language .................................... 50

+ 12.14 Content-Length ...................................... 50

+ 12.15 Content-Location .................................... 50

+ 12.16 Content-Type ........................................ 50

+ 12.17 CSeq ................................................ 50

+ 12.18 Date ................................................ 50

+ 12.19 Expires ............................................. 50

+ 12.20 From ................................................ 51

+ 12.21 Host ................................................ 51

+ 12.22 If-Match ............................................ 51

+ 12.23 If-Modified-Since ................................... 52

+ 12.24 Last-Modified........................................ 52

+ 12.25 Location ............................................ 52

+ 12.26 Proxy-Authenticate .................................. 52

+ 12.27 Proxy-Require ....................................... 52

+ 12.28 Public .............................................. 53

+ 12.29 Range ............................................... 53

+ 12.30 Referer ............................................. 54

+ 12.31 Retry-After ......................................... 54

+ 12.32 Require ............................................. 54

+ 12.33 RTP-Info ............................................ 55

+ 12.34 Scale ............................................... 56

+ 12.35 Speed ............................................... 57

+ 12.36 Server .............................................. 57

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RFC 2326 Real Time Streaming Protocol April 1998

+ 12.37 Session ............................................. 57

+ 12.38 Timestamp ........................................... 58

+ 12.39 Transport ........................................... 58

+ 12.40 Unsupported ......................................... 62

+ 12.41 User-Agent .......................................... 62

+ 12.42 Vary ................................................ 62

+ 12.43 Via ................................................. 62

+ 12.44 WWW-Authenticate .................................... 62

* 13 Caching ..................................................... 62

* 14 Examples .................................................... 63

+ 14.1 Media on Demand (Unicast) ............................ 63

+ 14.2 Streaming of a Container file ........................ 65

+ 14.3 Single Stream Container Files ........................ 67

+ 14.4 Live Media Presentation Using Multicast .............. 69

+ 14.5 Playing media into an existing session ............... 70

+ 14.6 Recording ............................................ 71

* 15 Syntax ...................................................... 72

+ 15.1 Base Syntax .......................................... 72

* 16 Security Considerations ..................................... 73

* A RTSP Protocol State Machines ................................. 76

+ A.1 Client State Machine .................................. 76

+ A.2 Server State Machine .................................. 77

* B Interaction with RTP ......................................... 79

* C Use of SDP for RTSP Session Descriptions ..................... 80

+ C.1 Definitions ........................................... 80

o C.1.1 Control URL .................................... 80

o C.1.2 Media streams .................................. 81

o C.1.3 Payload type(s) ................................ 81

o C.1.4 Format-specific parameters ..................... 81

o C.1.5 Range of presentation .......................... 82

o C.1.6 Time of availability ........................... 82

o C.1.7 Connection Information ......................... 82

o C.1.8 Entity Tag ..................................... 82

+ C.2 Aggregate Control Not Available ....................... 83

+ C.3 Aggregate Control Available ........................... 83

* D Minimal RTSP implementation .................................. 85

+ D.1 Client ................................................ 85

o D.1.1 Basic Playback ................................. 86

o D.1.2 Authentication-enabled ......................... 86

+ D.2 Server ................................................ 86

o D.2.1 Basic Playback ................................. 87

o D.2.2 Authentication-enabled ......................... 87

* E Authors' Addresses ........................................... 88

* F Acknowledgements ............................................. 89

* References ..................................................... 90

* Full Copyright Statement ....................................... 92

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RFC 2326 Real Time Streaming Protocol April 1998

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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RFC 2326 Real Time Streaming Protocol April 1998

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).

* 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) to identify the RTSP session whose state

is being manipulated. The server generates session identifiers

in response to SETUP requests (Section 10.4).

1.8 Relationship with Other Protocols

RTSP has some overlap in functionality with HTTP. It also may

interact with HTTP in that the initial contact with streaming content

is often to be made through a web page. The current protocol

specification aims to allow different hand-off points between a web

server and the media server implementing RTSP. For example, the

presentation description can be retrieved using HTTP or RTSP, which

reduces roundtrips in web-browser-based scenarios, yet also allows

for standalone RTSP servers and clients which do not rely on HTTP at

all.

However, RTSP differs fundamentally from HTTP in that data delivery

takes place out-of-band in a different protocol. HTTP is an

asymmetric protocol where the client issues requests and the server

responds. In RTSP, both the media client and media server can issue

requests. RTSP requests are also not stateless; they may set

parameters and continue to control a media stream long after the

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RFC 2326 Real Time Streaming Protocol April 1998

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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RFC 2326 Real Time Streaming Protocol April 1998

of RFC 1123 \cite{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 = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ]

[ "." 1*2DIGIT ]

Examples:

smpte=10:12:33:20-

smpte=10:07:33-

smpte=10:07:00-10:07:33:05.01

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

3.6 Normal Play Time

Normal play time (NPT) indicates the stream absolute position

relative to the beginning of the presentation. The timestamp consists

of a decimal fraction. The part left of the decimal may be expressed

in either seconds or hours, minutes, and seconds. The part right of

the decimal point measures fractions of a second.

The beginning of a presentation corresponds to 0.0 seconds. Negative

values are not defined. The special constant now is defined as the

current instant of a live event. It may be used only for live events.

NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the

viewer associates with a program. It is often digitally displayed on

a VCR. NPT advances normally when in normal play mode (scale = 1),

advances at a faster rate when in fast scan forward (high positive

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 = 1*DIGIT [ "." *DIGIT ]

npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]

npt-hh = 1*DIGIT ; any positive number

npt-mm = 1*2DIGIT ; 0-59

npt-ss = 1*2DIGIT ; 0-59

Examples:

npt=123.45-125

npt=12:05:35.3-

npt=now-

The syntax conforms to ISO 8601. The npt-sec notation is optimized

for automatic generation, the ntp-hhmmss notation for consumption

by human readers. The "now" constant allows clients to request to

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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 [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) 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

| Connection ; Section 12.10

| Date ; Section 12.18

| Via ; 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

*( general-header ; Section 5

| request-header ; Section 6.2

| entity-header ) ; Section 8.1

CRLF

[ message-body ] ; Section 4.3

6.1 Request Line

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

Method = "DESCRIBE" ; Section 10.2

| "ANNOUNCE" ; Section 10.3

| "GET_PARAMETER" ; Section 10.8

| "OPTIONS" ; Section 10.1

| "PAUSE" ; Section 10.6

| "PLAY" ; Section 10.5

| "RECORD" ; Section 10.11

| "REDIRECT" ; Section 10.10

| "SETUP" ; Section 10.4

| "SET_PARAMETER" ; Section 10.9

| "TEARDOWN" ; Section 10.7

| extension-method

extension-method = token

Request-URI = "*" | absolute_URI

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

6.2 Request Header Fields

request-header = Accept ; Section 12.1

| Accept-Encoding ; Section 12.2

| Accept-Language ; Section 12.3

| Authorization ; Section 12.5

| 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

*( general-header ; Section 5

| response-header ; Section 7.1.2

| entity-header ) ; Section 8.1

CRLF

[ message-body ] ; Section 4.3

7.1 Status-Line

The first line of a Response message is the Status-Line, consisting

of the protocol version followed by a numeric status code, and the

textual phrase associated with the status code, with each element

separated by SP characters. No CR or LF is allowed except in the

final CRLF sequence.

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

7.1.1 Status Code and Reason Phrase

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

- they may be replaced by local equivalents without affecting the

protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and

adds RTSP-specific status codes starting at x50 to avoid conflicts

with newly defined HTTP status codes.

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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

| Content-Base ; Section 12.11

| Content-Encoding ; Section 12.12

| Content-Language ; Section 12.13

| Content-Length ; Section 12.14

| Content-Location ; Section 12.15

| Content-Type ; Section 12.16

| Expires ; Section 12.19

| Last-Modified ; 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) [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 1*parameter

stream-url = "url" "=" url

parameter = ";" "seq" "=" 1*DIGIT

| ";" "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" "=" 1*DIGIT

| ";" "port" "=" port [ "-" port ]

| ";" "client_port" "=" port [ "-" port ]

| ";" "server_port" "=" port [ "-" port ]

| ";" "ssrc" "=" ssrc

| ";" "mode" = <"> 1\#mode <">

ttl = 1*3(DIGIT)

port = 1*5(DIGIT)

ssrc = 8*8(HEX)

channel = 1*3(DIGIT)

address = host

mode = <"> *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 = <any OCTET except CTLs>

tspecials = "(" | ")" | "<" | ">" | "@"

| "," | ";" | ":" | "\" | <">

| "/" | "[" | "]" | "?" | "="

| "{" | "}" | SP | HT

token = 1*<any CHAR except CTLs or tspecials>

quoted-string = ( <"> *(qdtext) <"> )

qdtext = <any TEXT except <">>

quoted-pair = "\" CHAR

message-header = field-name ":" [ field-value ] CRLF

field-name = token

field-value = *( field-content | LWS )

field-content = <the OCTETs making up the field-value and

consisting of either *TEXT or

combinations of token, tspecials, and

quoted-string>

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.

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.

Schulzrinne, et. al. Standards Track [Page 81]

RFC 2326 Real Time Streaming Protocol April 1998

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

Schulzrinne, et. al. Standards Track [Page 82]

RFC 2326 Real Time Streaming Protocol April 1998

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.

Schulzrinne, et. al. Standards Track [Page 83]

RFC 2326 Real Time Streaming Protocol April 1998

Example:

v=0

o=- 2890844256 2890842807 IN IP4 204.34.34.32

s=I contain

i=<more info>

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.

Schulzrinne, et. al. Standards Track [Page 84]

RFC 2326 Real Time Streaming Protocol April 1998

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

* 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.

Schulzrinne, et. al. Standards Track [Page 85]

RFC 2326 Real Time Streaming Protocol April 1998

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

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.

Schulzrinne, et. al. Standards Track [Page 86]

RFC 2326 Real Time Streaming Protocol April 1998

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

Schulzrinne, et. al. Standards Track [Page 87]

RFC 2326 Real Time Streaming Protocol April 1998

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

Schulzrinne, et. al. Standards Track [Page 88]

RFC 2326 Real Time Streaming Protocol April 1998

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.

Schulzrinne, et. al. Standards Track [Page 89]

RFC 2326 Real Time Streaming Protocol April 1998

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.

Schulzrinne, et. al. Standards Track [Page 90]

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.

Schulzrinne, et. al. Standards Track [Page 91]

RFC 2326 Real Time Streaming Protocol April 1998

Full Copyright Statement

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

This document and translations of it may be copied and furnished to

others, and derivative works that comment on or otherwise explain it

or assist in its implementation may be prepared, copied, published

and distributed, in whole or in part, without restriction of any

kind, provided that the above copyright notice and this paragraph are

included on all such copies and derivative works. However, this

document itself may not be modified in any way, such as by removing

the copyright notice or references to the Internet Society or other

Internet organizations, except as needed for the purpose of

developing Internet standards in which case the procedures for

copyrights defined in the Internet Standards process must be

followed, or as required to translate it into languages other than

English.

The limited permissions granted above are perpetual and will not be

revoked by the Internet Society or its successors or assigns.

This document and the information contained herein is provided on an

"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING

TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING

BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION

HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF

MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.

Schulzrinne, et. al. Standards Track [Page 92]

 
 
 
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