draft-ietf-mmusic-rtsp-08.txt   rfc2326.txt 
Internet Engineering Task Force MMUSIC WG
Internet Draft H. Schulzrinne, A. Rao, R. Lanphier
draft-ietf-mmusic-rtsp-08.txt Columbia U./Netscape/RealNetworks
January 15, 1998 Expires: July 15, 1998
Real Time Streaming Protocol (RTSP) Network Working Group H. Schulzrinne
Request for Comments: 2326 Columbia U.
Category: Standards Track A. Rao
Netscape
R. Lanphier
RealNetworks
April 1998
STATUS OF THIS MEMO Real Time Streaming Protocol (RTSP)
This document is an Internet-Draft. Internet-Drafts are working Status of this Memo
documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months This document specifies an Internet standards track protocol for the
and may be updated, replaced, or obsoleted by other documents at any Internet community, and requests discussion and suggestions for
time. It is inappropriate to use Internet-Drafts as reference improvements. Please refer to the current edition of the "Internet
material or to cite them other than as ``work in progress''. Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
To learn the current status of any Internet-Draft, please check the Copyright Notice
``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
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Distribution of this document is unlimited. Copyright (C) The Internet Society (1998). All Rights Reserved.
Abstract: Abstract
The Real Time Streaming Protocol, or RTSP, is an application-level The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP, sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and provide a means for choosing delivery multicast UDP and TCP, and provide a means for choosing delivery
mechanisms based upon RTP (RFC 1889). mechanisms based upon RTP (RFC 1889).
This is a snapshot of the current draft which will become the next
version of the ``official'' Internet Draft.
Copyright Notice:
Copyright (C) The Internet Society (1997). All Rights Reserved.
H. Schulzrinne, A. Rao, R. Lanphier Page 1
Table of Contents Table of Contents
* Contents * 1 Introduction ................................................. 5
* 1 Introduction + 1.1 Purpose ............................................... 5
+ 1.1 Purpose + 1.2 Requirements .......................................... 6
+ 1.2 Requirements + 1.3 Terminology ........................................... 6
+ 1.3 Terminology + 1.4 Protocol Properties ................................... 9
+ 1.4 Protocol Properties + 1.5 Extending RTSP ........................................ 11
+ 1.5 Extending RTSP + 1.6 Overall Operation ..................................... 11
+ 1.6 Overall Operation + 1.7 RTSP States ........................................... 12
+ 1.7 RTSP States + 1.8 Relationship with Other Protocols ..................... 13
+ 1.8 Relationship with Other Protocols * 2 Notational Conventions ....................................... 14
* 2 Notational Conventions * 3 Protocol Parameters .......................................... 14
* 3 Protocol Parameters + 3.1 RTSP Version .......................................... 14
+ 3.1 RTSP Version + 3.2 RTSP URL .............................................. 14
+ 3.2 RTSP URL + 3.3 Conference Identifiers ................................ 16
+ 3.3 Conference Identifiers + 3.4 Session Identifiers ................................... 16
+ 3.4 Session Identifiers + 3.5 SMPTE Relative Timestamps ............................. 16
+ 3.5 SMPTE Relative Timestamps + 3.6 Normal Play Time ...................................... 17
+ 3.6 Normal Play Time + 3.7 Absolute Time ......................................... 18
+ 3.7 Absolute Time + 3.8 Option Tags ........................................... 18
+ 3.8 Option Tags o 3.8.1 Registering New Option Tags with IANA .......... 18
o 3.8.1 Registering New Option Tags with IANA * 4 RTSP Message ................................................. 19
* 4 RTSP Message + 4.1 Message Types ......................................... 19
+ 4.1 Message Types + 4.2 Message Headers ....................................... 19
+ 4.2 Message Headers + 4.3 Message Body .......................................... 19
+ 4.3 Message Body + 4.4 Message Length ........................................ 20
+ 4.4 Message Length * 5 General Header Fields ........................................ 20
* 5 General Header Fields * 6 Request ...................................................... 20
* 6 Request + 6.1 Request Line .......................................... 21
+ 6.1 Request Line + 6.2 Request Header Fields ................................. 21
+ 6.2 Request Header Fields * 7 Response ..................................................... 22
* 7 Response + 7.1 Status-Line ........................................... 22
+ 7.1 Status-Line o 7.1.1 Status Code and Reason Phrase .................. 22
o 7.1.1 Status Code and Reason Phrase o 7.1.2 Response Header Fields ......................... 26
o 7.1.2 Response Header Fields * 8 Entity ....................................................... 27
* 8 Entity + 8.1 Entity Header Fields .................................. 27
+ 8.1 Entity Header Fields + 8.2 Entity Body ........................................... 28
+ 8.2 Entity Body * 9 Connections .................................................. 28
* 9 Connections + 9.1 Pipelining ............................................ 28
+ 9.1 Pipelining + 9.2 Reliability and Acknowledgements ...................... 28
+ 9.2 Reliability and Acknowledgements * 10 Method Definitions .......................................... 29
* 10 Method Definitions + 10.1 OPTIONS .............................................. 30
+ 10.1 OPTIONS + 10.2 DESCRIBE ............................................. 31
+ 10.2 DESCRIBE + 10.3 ANNOUNCE ............................................. 32
+ 10.3 ANNOUNCE + 10.4 SETUP ................................................ 33
+ 10.5 PLAY ................................................. 34
H. Schulzrinne, A. Rao, R. Lanphier Page 2 + 10.6 PAUSE ................................................ 36
+ 10.4 SETUP + 10.7 TEARDOWN ............................................. 37
+ 10.5 PLAY + 10.8 GET_PARAMETER ........................................ 37
+ 10.6 PAUSE + 10.9 SET_PARAMETER ........................................ 38
+ 10.7 TEARDOWN + 10.10 REDIRECT ............................................ 39
+ 10.8 GET_PARAMETER + 10.11 RECORD .............................................. 39
+ 10.9 SET_PARAMETER + 10.12 Embedded (Interleaved) Binary Data .................. 40
+ 10.10 REDIRECT * 11 Status Code Definitions ..................................... 41
+ 10.11 RECORD + 11.1 Success 2xx .......................................... 41
+ 10.12 Embedded (Interleaved) Binary Data o 11.1.1 250 Low on Storage Space ...................... 41
* 11 Status Code Definitions + 11.2 Redirection 3xx ...................................... 41
+ 11.1 Success 2xx + 11.3 Client Error 4xx ..................................... 42
o 11.1.1 250 Low on Storage Space o 11.3.1 405 Method Not Allowed ........................ 42
+ 11.2 Redirection 3xx o 11.3.2 451 Parameter Not Understood .................. 42
+ 11.3 Client Error 4xx o 11.3.3 452 Conference Not Found ...................... 42
o 11.3.1 405 Method Not Allowed o 11.3.4 453 Not Enough Bandwidth ...................... 42
o 11.3.2 451 Parameter Not Understood o 11.3.5 454 Session Not Found ......................... 42
o 11.3.3 452 Conference Not Found o 11.3.6 455 Method Not Valid in This State ............ 42
o 11.3.4 453 Not Enough Bandwidth o 11.3.7 456 Header Field Not Valid for Resource ....... 42
o 11.3.5 454 Session Not Found o 11.3.8 457 Invalid Range ............................. 43
o 11.3.6 455 Method Not Valid in This State o 11.3.9 458 Parameter Is Read-Only .................... 43
o 11.3.7 456 Header Field Not Valid for Resource o 11.3.10 459 Aggregate Operation Not Allowed .......... 43
o 11.3.8 457 Invalid Range o 11.3.11 460 Only Aggregate Operation Allowed ......... 43
o 11.3.9 458 Parameter Is Read-Only o 11.3.12 461 Unsupported Transport .................... 43
o 11.3.10 459 Aggregate Operation Not Allowed o 11.3.13 462 Destination Unreachable .................. 43
o 11.3.11 460 Only Aggregate Operation Allowed o 11.3.14 551 Option not supported ..................... 43
o 11.3.12 461 Unsupported Transport * 12 Header Field Definitions .................................... 44
o 11.3.13 462 Destination Unreachable + 12.1 Accept ............................................... 46
o 11.3.14 551 Option not supported + 12.2 Accept-Encoding ...................................... 46
* 12 Header Field Definitions + 12.3 Accept-Language ...................................... 46
+ 12.1 Accept + 12.4 Allow ................................................ 46
+ 12.2 Accept-Encoding + 12.5 Authorization ........................................ 46
+ 12.3 Accept-Language + 12.6 Bandwidth ............................................ 46
+ 12.4 Allow + 12.7 Blocksize ............................................ 47
+ 12.5 Authorization + 12.8 Cache-Control ........................................ 47
+ 12.6 Bandwidth + 12.9 Conference ........................................... 49
+ 12.7 Blocksize + 12.10 Connection .......................................... 49
+ 12.8 Cache-Control + 12.11 Content-Base ........................................ 49
+ 12.9 Conference + 12.12 Content-Encoding .................................... 49
+ 12.10 Connection + 12.13 Content-Language .................................... 50
+ 12.11 Content-Base + 12.14 Content-Length ...................................... 50
+ 12.12 Content-Encoding + 12.15 Content-Location .................................... 50
+ 12.13 Content-Language + 12.16 Content-Type ........................................ 50
+ 12.14 Content-Length + 12.17 CSeq ................................................ 50
+ 12.15 Content-Location + 12.18 Date ................................................ 50
+ 12.16 Content-Type + 12.19 Expires ............................................. 50
+ 12.17 CSeq + 12.20 From ................................................ 51
+ 12.21 Host ................................................ 51
H. Schulzrinne, A. Rao, R. Lanphier Page 3 + 12.22 If-Match ............................................ 51
+ 12.18 Date + 12.23 If-Modified-Since ................................... 52
+ 12.19 Expires + 12.24 Last-Modified........................................ 52
+ 12.20 From + 12.25 Location ............................................ 52
+ 12.21 Host + 12.26 Proxy-Authenticate .................................. 52
+ 12.22 If-Match + 12.27 Proxy-Require ....................................... 52
+ 12.23 If-Modified-Since + 12.28 Public .............................................. 53
+ 12.24 Last-Modified + 12.29 Range ............................................... 53
+ 12.25 Location + 12.30 Referer ............................................. 54
+ 12.26 Proxy-Authenticate + 12.31 Retry-After ......................................... 54
+ 12.27 Proxy-Require + 12.32 Require ............................................. 54
+ 12.28 Public + 12.33 RTP-Info ............................................ 55
+ 12.29 Range + 12.34 Scale ............................................... 56
+ 12.30 Referer + 12.35 Speed ............................................... 57
+ 12.31 Retry-After + 12.36 Server .............................................. 57
+ 12.32 Require + 12.37 Session ............................................. 57
+ 12.33 RTP-Info + 12.38 Timestamp ........................................... 58
+ 12.34 Scale + 12.39 Transport ........................................... 58
+ 12.35 Speed + 12.40 Unsupported ......................................... 62
+ 12.36 Server + 12.41 User-Agent .......................................... 62
+ 12.37 Session + 12.42 Vary ................................................ 62
+ 12.38 Timestamp + 12.43 Via ................................................. 62
+ 12.39 Transport + 12.44 WWW-Authenticate .................................... 62
+ 12.40 Unsupported * 13 Caching ..................................................... 62
+ 12.41 User-Agent * 14 Examples .................................................... 63
+ 12.42 Vary + 14.1 Media on Demand (Unicast) ............................ 63
+ 12.43 Via + 14.2 Streaming of a Container file ........................ 65
+ 12.44 WWW-Authenticate + 14.3 Single Stream Container Files ........................ 67
* 13 Caching + 14.4 Live Media Presentation Using Multicast .............. 69
* 14 Examples + 14.5 Playing media into an existing session ............... 70
+ 14.1 Media on Demand (Unicast) + 14.6 Recording ............................................ 71
+ 14.2 Streaming of a Container file * 15 Syntax ...................................................... 72
+ 14.3 Single Stream Container Files + 15.1 Base Syntax .......................................... 72
+ 14.4 Live Media Presentation Using Multicast * 16 Security Considerations ..................................... 73
+ 14.5 Playing media into an existing session * A RTSP Protocol State Machines ................................. 76
+ 14.6 Recording + A.1 Client State Machine .................................. 76
* 15 Syntax + A.2 Server State Machine .................................. 77
+ 15.1 Base Syntax * B Interaction with RTP ......................................... 79
* 16 Security Considerations * C Use of SDP for RTSP Session Descriptions ..................... 80
* A RTSP Protocol State Machines + C.1 Definitions ........................................... 80
+ A.1 Client State Machine o C.1.1 Control URL .................................... 80
+ A.2 Server State Machine o C.1.2 Media streams .................................. 81
* B Interaction with RTP o C.1.3 Payload type(s) ................................ 81
* C Use of SDP for RTSP Session Descriptions o C.1.4 Format-specific parameters ..................... 81
+ C.1 Definitions o C.1.5 Range of presentation .......................... 82
o C.1.1 Control URL o C.1.6 Time of availability ........................... 82
o C.1.2 Media streams o C.1.7 Connection Information ......................... 82
o C.1.8 Entity Tag ..................................... 82
H. Schulzrinne, A. Rao, R. Lanphier Page 4 + C.2 Aggregate Control Not Available ....................... 83
o C.1.3 Payload type(s) + C.3 Aggregate Control Available ........................... 83
o C.1.4 Format-specific parameters * D Minimal RTSP implementation .................................. 85
o C.1.5 Range of presentation + D.1 Client ................................................ 85
o C.1.6 Time of availability o D.1.1 Basic Playback ................................. 86
o C.1.7 Connection Information o D.1.2 Authentication-enabled ......................... 86
o C.1.8 Entity Tag + D.2 Server ................................................ 86
+ C.2 Aggregate Control Not Available o D.2.1 Basic Playback ................................. 87
+ C.3 Aggregate Control Available o D.2.2 Authentication-enabled ......................... 87
* D Minimal RTSP implementation * E Authors' Addresses ........................................... 88
+ D.1 Client * F Acknowledgements ............................................. 89
o D.1.1 Basic Playback * References ..................................................... 90
o D.1.2 Authentication-enabled * Full Copyright Statement ....................................... 92
+ D.2 Server
o D.2.1 Basic Playback
o D.2.2 Authentication-enabled
* E Author Addresses
* F Acknowledgements
* References
1 Introduction 1 Introduction
1.1 Purpose 1.1 Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls The Real-Time Streaming Protocol (RTSP) establishes and controls
either a single or several time-synchronized streams of continuous either a single or several time-synchronized streams of continuous
media such as audio and video. It does not typically deliver the media such as audio and video. It does not typically deliver the
continuous streams itself, although interleaving of the continuous continuous streams itself, although interleaving of the continuous
media stream with the control stream is possible (see Section 10.12). media stream with the control stream is possible (see Section 10.12).
In other words, RTSP acts as a ``network remote control'' for In other words, RTSP acts as a "network remote control" for
multimedia servers. multimedia servers.
The set of streams to be controlled is defined by a presentation The set of streams to be controlled is defined by a presentation
description. This memorandum does not define a format for a description. This memorandum does not define a format for a
presentation description. presentation description.
There is no notion of an RTSP connection; instead, a server maintains 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 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 to a transport-level connection such as a TCP connection. During an
RTSP session, an RTSP client may open and close many reliable RTSP session, an RTSP client may open and close many reliable
transport connections to the server to issue RTSP requests. transport connections to the server to issue RTSP requests.
Alternatively, it may use a connectionless transport protocol such as Alternatively, it may use a connectionless transport protocol such as
UDP. UDP.
The streams controlled by RTSP may use RTP [1], but the operation of The streams controlled by RTSP may use RTP [1], but the operation of
RTSP does not depend on the transport mechanism used to carry RTSP does not depend on the transport mechanism used to carry
continuous media. continuous media. The protocol is intentionally similar in syntax
and operation to HTTP/1.1 [2] so that extension mechanisms to HTTP
H. Schulzrinne, A. Rao, R. Lanphier Page 5 can in most cases also be added to RTSP. However, RTSP differs in a
The protocol is intentionally similar in syntax and operation to number of important aspects from HTTP:
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 * RTSP introduces a number of new methods and has a different
protocol identifier. protocol identifier.
* An RTSP server needs to maintain state by default in almost all * An RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP. cases, as opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests. * Both an RTSP server and client can issue requests.
* Data is carried out-of-band by a different protocol. (There is an * Data is carried out-of-band by a different protocol. (There is an
exception to this.) exception to this.)
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts [3]. consistent with current HTML internationalization efforts [3].
* The Request-URI always contains the absolute URI. Because of * The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1 [2] backward compatibility with a historical blunder, HTTP/1.1 [2]
carries only the absolute path in the request and puts the host carries only the absolute path in the request and puts the host
name in a separate header field. name in a separate header field.
This makes ``virtual hosting'' easier, where a single host with one This makes "virtual hosting" easier, where a single host with one
IP address hosts several document trees. IP address hosts several document trees.
The protocol supports the following operations: The protocol supports the following operations:
Retrieval of media from media server: Retrieval of media from media server:
The client can request a presentation description via HTTP or The client can request a presentation description via HTTP or
some other method. If the presentation is being multicast, the some other method. If the presentation is being multicast, the
presentation description contains the multicast addresses and presentation description contains the multicast addresses and
ports to be used for the continuous media. If the presentation ports to be used for the continuous media. If the presentation
is to be sent only to the client via unicast, the client is to be sent only to the client via unicast, the client
provides the destination for security reasons. provides the destination for security reasons.
Invitation of a media server to a conference: Invitation of a media server to a conference:
A media server can be ``invited'' to join an existing A media server can be "invited" to join an existing
conference, either to play back media into the presentation or conference, either to play back media into the presentation or
to record all or a subset of the media in a presentation. This to record all or a subset of the media in a presentation. This
mode is useful for distributed teaching applications. Several mode is useful for distributed teaching applications. Several
parties in the conference may take turns ``pushing the remote parties in the conference may take turns "pushing the remote
control buttons''. control buttons."
Addition of media to an existing presentation: Addition of media to an existing presentation:
Particularly for live presentations, it is useful if the server Particularly for live presentations, it is useful if the
can tell the client about additional media becoming available. server can tell the client about additional media becoming
available.
RTSP requests may be handled by proxies, tunnels and caches as in RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1 [2]. HTTP/1.1 [2].
H. Schulzrinne, A. Rao, R. Lanphier Page 6
1.2 Requirements 1.2 Requirements
The key words ``MUST'', ``MUST NOT'', ``REQUIRED'', ``SHALL'', ``SHALL The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
NOT'', ``SHOULD'', ``SHOULD NOT'', ``RECOMMENDED'', ``MAY'', and "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
``OPTIONAL'' in this document are to be interpreted as described in document are to be interpreted as described in RFC 2119 [4].
RFC 2119 [4].
1.3 Terminology 1.3 Terminology
Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not
listed here are defined as in HTTP/1.1. listed here are defined as in HTTP/1.1.
Aggregate control: Aggregate control:
The control of the multiple streams using a single timeline by The control of the multiple streams using a single timeline by
the server. For audio/video feeds, this means that the client the server. For audio/video feeds, this means that the client
may issue a single play or pause message to control both the may issue a single play or pause message to control both the
audio and video feeds. audio and video feeds.
Conference: Conference:
a multiparty, multimedia presentation, where ``multi'' implies a multiparty, multimedia presentation, where "multi" implies
greater than or equal to one. greater than or equal to one.
Client: Client:
The client requests continuous media data from the media The client requests continuous media data from the media
server. server.
Connection: Connection:
A transport layer virtual circuit established between two A transport layer virtual circuit established between two
programs for the purpose of communication. programs for the purpose of communication.
Container file: Container file:
A file which may contain multiple media streams which often A file which may contain multiple media streams which often
comprise a presentation when played together. RTSP servers may comprise a presentation when played together. RTSP servers may
offer aggregate control on these files, though the concept of a offer aggregate control on these files, though the concept of
container file is not embedded in the protocol. a container file is not embedded in the protocol.
Continuous media: Continuous media:
Data where there is a timing relationship between source and Data where there is a timing relationship between source and
sink; that is, the sink must reproduce the timing relationship sink; that is, the sink must reproduce the timing relationship
that existed at the source. The most common examples of that existed at the source. The most common examples of
continuous media are audio and motion video. Continuous media continuous media are audio and motion video. Continuous media
can be real-time (interactive), where there is a ``tight'' can be real-time (interactive), where there is a "tight"
timing relationship between source and sink, or streaming timing relationship between source and sink, or streaming
(playback), where the relationship is less strict. (playback), where the relationship is less strict.
H. Schulzrinne, A. Rao, R. Lanphier Page 7
Entity: Entity:
The information transferred as the payload of a request or The information transferred as the payload of a request or
response. An entity consists of metainformation in the form of response. An entity consists of metainformation in the form of
entity-header fields and content in the form of an entity-body, entity-header fields and content in the form of an entity-
as described in Section 8. body, as described in Section 8.
Media initialization: Media initialization:
Datatype/codec specific initialization. This includes such Datatype/codec specific initialization. This includes such
things as clockrates, color tables, etc. Any things as clockrates, color tables, etc. Any transport-
transport-independent information which is required by a client independent information which is required by a client for
for playback of a media stream occurs in the media playback of a media stream occurs in the media initialization
initialization phase of stream setup. phase of stream setup.
Media parameter: Media parameter:
Parameter specific to a media type that may be changed before Parameter specific to a media type that may be changed before
or during stream playback. or during stream playback.
Media server: Media server:
The server providing playback or recording services for one or The server providing playback or recording services for one or
more media streams. Different media streams within a more media streams. Different media streams within a
presentation may originate from different media servers. A presentation may originate from different media servers. A
media server may reside on the same or a different host as the media server may reside on the same or a different host as the
skipping to change at page 10, line ? skipping to change at page 8, line 18
(Media) stream: (Media) stream:
A single media instance, e.g., an audio stream or a video A single media instance, e.g., an audio stream or a video
stream as well as a single whiteboard or shared application stream as well as a single whiteboard or shared application
group. When using RTP, a stream consists of all RTP and RTCP group. When using RTP, a stream consists of all RTP and RTCP
packets created by a source within an RTP session. This is packets created by a source within an RTP session. This is
equivalent to the definition of a DSM-CC stream([5]). equivalent to the definition of a DSM-CC stream([5]).
Message: Message:
The basic unit of RTSP communication, consisting of a The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in structured sequence of octets matching the syntax defined in
Section 15 and transmitted via a connection or a connectionless Section 15 and transmitted via a connection or a
protocol. connectionless protocol.
Participant: Participant:
Member of a conference. A participant may be a machine, e.g., a Member of a conference. A participant may be a machine, e.g.,
media record or playback server. a media record or playback server.
H. Schulzrinne, A. Rao, R. Lanphier Page 8
Presentation: Presentation:
A set of one or more streams presented to the client as a A set of one or more streams presented to the client as a
complete media feed, using a presentation description as complete media feed, using a presentation description as
defined below. In most cases in the RTSP context, this implies defined below. In most cases in the RTSP context, this implies
aggregate control of those streams, but does not have to. aggregate control of those streams, but does not have to.
Presentation description: Presentation description:
A presentation description contains information about one or A presentation description contains information about one or
more media streams within a presentation, such as the set of more media streams within a presentation, such as the set of
encodings, network addresses and information about the content. encodings, network addresses and information about the
Other IETF protocols such as SDP (RFC XXXX [6]) use the term content. Other IETF protocols such as SDP (RFC 2327 [6]) use
``session'' for a live presentation. The presentation the term "session" for a live presentation. The presentation
description may take several different formats, including but description may take several different formats, including but
not limited to the session description format SDP. not limited to the session description format SDP.
Response: Response:
An RTSP response. If an HTTP response is meant, that is An RTSP response. If an HTTP response is meant, that is
indicated explicitly. indicated explicitly.
Request: Request:
An RTSP request. If an HTTP request is meant, that is indicated An RTSP request. If an HTTP request is meant, that is
explicitly. indicated explicitly.
RTSP session: RTSP session:
A complete RTSP ``transaction'', e.g., the viewing of a movie. A complete RTSP "transaction", e.g., the viewing of a movie.
A session typically consists of a client setting up a transport A session typically consists of a client setting up a
mechanism for the continuous media stream (SETUP), starting the transport mechanism for the continuous media stream (SETUP),
stream with PLAY or RECORD, and closing the stream with starting the stream with PLAY or RECORD, and closing the
TEARDOWN. stream with TEARDOWN.
Transport initialization: Transport initialization:
The negotiation of transport information (e.g., port numbers, The negotiation of transport information (e.g., port numbers,
transport protocols) between the client and the server. transport protocols) between the client and the server.
1.4 Protocol Properties 1.4 Protocol Properties
RTSP has the following properties: RTSP has the following properties:
Extendable: Extendable:
New methods and parameters can be easily added to RTSP. New methods and parameters can be easily added to RTSP.
Easy to parse: Easy to parse:
RTSP can be parsed by standard HTTP or MIME parsers. RTSP can be parsed by standard HTTP or MIME parsers.
H. Schulzrinne, A. Rao, R. Lanphier Page 9
Secure: Secure:
RTSP re-uses web security mechanisms, either at the transport RTSP re-uses web security mechanisms. All HTTP authentication
level (TLS, RFC XXXX [7]) or within the protocol itself. All mechanisms such as basic (RFC 2068 [2, Section 11.1]) and
HTTP authentication mechanisms such as basic (RFC 2068 [2, digest authentication (RFC 2069 [8]) are directly applicable.
Section 11.1]) and digest authentication (RFC 2069 [8]) are One may also reuse transport or network layer security
directly applicable. mechanisms.
Transport-independent: Transport-independent:
RTSP may use either an unreliable datagram protocol (UDP) (RFC RTSP may use either an unreliable datagram protocol (UDP) (RFC
768 [9]), a reliable datagram protocol (RDP, RFC 1151, not 768 [9]), a reliable datagram protocol (RDP, RFC 1151, not
widely used [10]) or a reliable stream protocol such as TCP widely used [10]) or a reliable stream protocol such as TCP
(RFC 793 [11]) as it implements application-level reliability. (RFC 793 [11]) as it implements application-level reliability.
Multi-server capable: Multi-server capable:
Each media stream within a presentation can reside on a Each media stream within a presentation can reside on a
different server. The client automatically establishes several different server. The client automatically establishes several
concurrent control sessions with the different media servers. concurrent control sessions with the different media servers.
Media synchronization is performed at the transport level. Media synchronization is performed at the transport level.
Control of recording devices: Control of recording devices:
The protocol can control both recording and playback devices, The protocol can control both recording and playback devices,
as well as devices that can alternate between the two modes as well as devices that can alternate between the two modes
(``VCR''). ("VCR").
Separation of stream control and conference initiation: Separation of stream control and conference initiation:
Stream control is divorced from inviting a media server to a Stream control is divorced from inviting a media server to a
conference. The only requirement is that the conference conference. The only requirement is that the conference
initiation protocol either provides or can be used to create a initiation protocol either provides or can be used to create a
unique conference identifier. In particular, SIP [12] or H.323 unique conference identifier. In particular, SIP [12] or H.323
[13] may be used to invite a server to a conference. [13] may be used to invite a server to a conference.
Suitable for professional applications: Suitable for professional applications:
RTSP supports frame-level accuracy through SMPTE time stamps to RTSP supports frame-level accuracy through SMPTE time stamps
allow remote digital editing. to allow remote digital editing.
Presentation description neutral: Presentation description neutral:
The protocol does not impose a particular presentation The protocol does not impose a particular presentation
description or metafile format and can convey the type of description or metafile format and can convey the type of
format to be used. However, the presentation description must format to be used. However, the presentation description must
contain at least one RTSP URI. contain at least one RTSP URI.
Proxy and firewall friendly: Proxy and firewall friendly:
The protocol should be readily handled by both application and The protocol should be readily handled by both application and
transport-layer (SOCKS [14]) firewalls. A firewall may need to transport-layer (SOCKS [14]) firewalls. A firewall may need to
understand the SETUP method to open a ``hole'' for the UDP understand the SETUP method to open a "hole" for the UDP media
media stream. stream.
HTTP-friendly: HTTP-friendly:
Where sensible, RTSP reuses HTTP concepts, so that the existing Where sensible, RTSP reuses HTTP concepts, so that the
infrastructure can be reused. This infrastructure includes PICS existing infrastructure can be reused. This infrastructure
(Platform for Internet Content Selection [15,16]) for includes PICS (Platform for Internet Content Selection
associating labels with content. However, RTSP does not just [15,16]) for associating labels with content. However, RTSP
add methods to HTTP since the controlling continuous media does not just add methods to HTTP since the controlling
requires server state in most cases. continuous media requires server state in most cases.
Appropriate server control: Appropriate server control:
If a client can start a stream, it must be able to stop a 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 stream. Servers should not start streaming to clients in such
way that clients cannot stop the stream. a way that clients cannot stop the stream.
Transport negotiation: Transport negotiation:
The client can negotiate the transport method prior to actually The client can negotiate the transport method prior to
needing to process a continuous media stream. actually needing to process a continuous media stream.
Capability negotiation: Capability negotiation:
If basic features are disabled, there must be some clean If basic features are disabled, there must be some clean
mechanism for the client to determine which methods are not mechanism for the client to determine which methods are not
going to be implemented. This allows clients to present the going to be implemented. This allows clients to present the
appropriate user interface. For example, if seeking is not appropriate user interface. For example, if seeking is not
allowed, the user interface must be able to disallow moving a allowed, the user interface must be able to disallow moving a
sliding position indicator. sliding position indicator.
An earlier requirement in RTSP was multi-client capability. An earlier requirement in RTSP was multi-client capability.
However, it was determined that a better approach was to make sure However, it was determined that a better approach was to make sure
that the protocol is easily extensible to the multi-client that the protocol is easily extensible to the multi-client
scenario. Stream identifiers can be used by several control scenario. Stream identifiers can be used by several control
streams, so that ``passing the remote'' would be possible. The streams, so that "passing the remote" would be possible. The
protocol would not address how several clients negotiate access; protocol would not address how several clients negotiate access;
this is left to either a ``social protocol'' or some other floor this is left to either a "social protocol" or some other floor
control mechanism. control mechanism.
1.5 Extending RTSP 1.5 Extending RTSP
Since not all media servers have the same functionality, media servers Since not all media servers have the same functionality, media
by necessity will support different sets of requests. For example: servers by necessity will support different sets of requests. For
example:
* A server may only be capable of playback thus has no need to * A server may only be capable of playback thus has no need to
support the RECORD request. support the RECORD request.
* A server may not be capable of seeking (absolute positioning) if * A server may not be capable of seeking (absolute positioning) if
it is to support live events only. it is to support live events only.
* Some servers may not support setting stream parameters and thus * Some servers may not support setting stream parameters and thus
not support GET_PARAMETER and SET_PARAMETER. not support GET_PARAMETER and SET_PARAMETER.
A server SHOULD implement all header fields described in Section 12. A server SHOULD implement all header fields described in Section 12.
It is up to the creators of presentation descriptions not to ask the 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], 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 where the methods described in [H19.6] are not likely to be supported
across all servers. across all servers.
RTSP can be extended in three ways, listed here in order of the RTSP can be extended in three ways, listed here in order of the
magnitude of changes supported: magnitude of changes supported:
* Existing methods can be extended with new parameters, as long as * Existing methods can be extended with new parameters, as long as
these parameters can be safely ignored by the recipient. (This is these parameters can be safely ignored by the recipient. (This is
equivalent to adding new parameters to an HTML tag.) If the client equivalent to adding new parameters to an HTML tag.) If the
needs negative acknowledgement when a method extension is not client needs negative acknowledgement when a method extension is
supported, a tag corresponding to the extension may be added in not supported, a tag corresponding to the extension may be added
the Require: field (see Section 12.32). in the Require: field (see Section 12.32).
* New methods can be added. If the recipient of the message does not * New methods can be added. If the recipient of the message does
understand the request, it responds with error code 501 (Not not understand the request, it responds with error code 501 (Not
implemented) and the sender should not attempt to use this method implemented) and the sender should not attempt to use this method
again. A client may also use the OPTIONS method to inquire about again. A client may also use the OPTIONS method to inquire about
methods supported by the server. The server SHOULD list the methods supported by the server. The server SHOULD list the
methods it supports using the Public response header. methods it supports using the Public response header.
* A new version of the protocol can be defined, allowing almost all * A new version of the protocol can be defined, allowing almost all
aspects (except the position of the protocol version number) to aspects (except the position of the protocol version number) to
change. change.
1.6 Overall Operation 1.6 Overall Operation
Each presentation and media stream may be identified by an RTSP URL. Each presentation and media stream may be identified by an RTSP URL.
The overall presentation and the properties of the media the The overall presentation and the properties of the media the
presentation is made up of are defined by a presentation description presentation is made up of are defined by a presentation description
file, the format of which is outside the scope of this specification. file, the format of which is outside the scope of this specification.
The presentation description file may be obtained by the client using The presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored on HTTP or other means such as email and may not necessarily be stored
the media server. on the media server.
For the purposes of this specification, a presentation description is For the purposes of this specification, a presentation description is
assumed to describe one or more presentations, each of which maintains assumed to describe one or more presentations, each of which
a common time axis. For simplicity of exposition and without loss of maintains a common time axis. For simplicity of exposition and
generality, it is assumed that the presentation description contains without loss of generality, it is assumed that the presentation
exactly one such presentation. A presentation may contain several description contains exactly one such presentation. A presentation
media streams. may contain several media streams.
The presentation description file contains a description of the media The presentation description file contains a description of the media
streams making up the presentation, including their encodings, streams making up the presentation, including their encodings,
language, and other parameters that enable the client to choose the language, and other parameters that enable the client to choose the
most appropriate combination of media. In this presentation most appropriate combination of media. In this presentation
description, each media stream that is individually controllable by description, each media stream that is individually controllable by
RTSP is identified by an RTSP URL, which points to the media server RTSP is identified by an RTSP URL, which points to the media server
handling that particular media stream and names the stream stored on handling that particular media stream and names the stream stored on
that server. Several media streams can be located on different that server. Several media streams can be located on different
servers; for example, audio and video streams can be split across servers; for example, audio and video streams can be split across
servers for load sharing. The description also enumerates which servers for load sharing. The description also enumerates which
transport methods the server is capable of. transport methods the server is capable of.
Besides the media parameters, the network destination address and port Besides the media parameters, the network destination address and
need to be determined. Several modes of operation can be port need to be determined. Several modes of operation can be
distinguished: distinguished:
Unicast: Unicast:
The media is transmitted to the source of the RTSP request, The media is transmitted to the source of the RTSP request,
with the port number chosen by the client. Alternatively, the with the port number chosen by the client. Alternatively, the
media is transmitted on the same reliable stream as RTSP. media is transmitted on the same reliable stream as RTSP.
Multicast, server chooses address: Multicast, server chooses address:
The media server picks the multicast address and port. This is The media server picks the multicast address and port. This is
the typical case for a live or near-media-on-demand the typical case for a live or near-media-on-demand
transmission. transmission.
Multicast, client chooses address: Multicast, client chooses address:
If the server is to participate in an existing multicast If the server is to participate in an existing multicast
conference, the multicast address, port and encryption key are conference, the multicast address, port and encryption key are
given by the conference description, established by means given by the conference description, established by means
outside the scope of this specification. outside the scope of this specification.
1.7 RTSP States 1.7 RTSP States
RTSP controls a stream which may be sent via a separate protocol, RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel. For example, RTSP control may independent of the control channel. For example, RTSP control may
occur on a TCP connection while the data flows via UDP. Thus, data 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 delivery continues even if no RTSP requests are received by the media
server. Also, during its lifetime, a single media stream may be server. Also, during its lifetime, a single media stream may be
controlled by RTSP requests issued sequentially on different TCP controlled by RTSP requests issued sequentially on different TCP
connections. Therefore, the server needs to maintain ``session state'' connections. Therefore, the server needs to maintain "session state"
to be able to correlate RTSP requests with a stream. The state to be able to correlate RTSP requests with a stream. The state
transitions are described in Section A. transitions are described in Section A.
Many methods in RTSP do not contribute to state. However, the Many methods in RTSP do not contribute to state. However, the
following play a central role in defining the allocation and usage of following play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and
TEARDOWN. TEARDOWN.
SETUP: SETUP:
Causes the server to allocate resources for a stream and start Causes the server to allocate resources for a stream and start
skipping to change at page 14, line 24 skipping to change at page 13, line 29
PLAY and RECORD: PLAY and RECORD:
Starts data transmission on a stream allocated via SETUP. Starts data transmission on a stream allocated via SETUP.
PAUSE: PAUSE:
Temporarily halts a stream without freeing server resources. Temporarily halts a stream without freeing server resources.
TEARDOWN: TEARDOWN:
Frees resources associated with the stream. The RTSP session Frees resources associated with the stream. The RTSP session
ceases to exist on the server. 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 1.8 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may interact RTSP has some overlap in functionality with HTTP. It also may
with HTTP in that the initial contact with streaming content is often interact with HTTP in that the initial contact with streaming content
to be made through a web page. The current protocol specification aims is often to be made through a web page. The current protocol
to allow different hand-off points between a web server and the media specification aims to allow different hand-off points between a web
server implementing RTSP. For example, the presentation description server and the media server implementing RTSP. For example, the
can be retrieved using HTTP or RTSP, which reduces roundtrips in presentation description can be retrieved using HTTP or RTSP, which
web-browser-based scenarios, yet also allows for standalone RTSP reduces roundtrips in web-browser-based scenarios, yet also allows
servers and clients which do not rely on HTTP at all. for standalone RTSP servers and clients which do not rely on HTTP at
all.
However, RTSP differs fundamentally from HTTP in that data delivery However, RTSP differs fundamentally from HTTP in that data delivery
takes place out-of-band in a different protocol. HTTP is an asymmetric takes place out-of-band in a different protocol. HTTP is an
protocol where the client issues requests and the server responds. In asymmetric protocol where the client issues requests and the server
RTSP, both the media client and media server can issue requests. RTSP responds. In RTSP, both the media client and media server can issue
requests are also not stateless; they may set parameters and continue requests. RTSP requests are also not stateless; they may set
to control a media stream long after the request has been parameters and continue to control a media stream long after the
acknowledged. request has been acknowledged.
Re-using HTTP functionality has advantages in at least two areas, Re-using HTTP functionality has advantages in at least two areas,
namely security and proxies. The requirements are very similar, so namely security and proxies. The requirements are very similar, so
having the ability to adopt HTTP work on caches, proxies and having the ability to adopt HTTP work on caches, proxies and
authentication is valuable. authentication is valuable.
While most real-time media will use RTP as a transport protocol, RTSP While most real-time media will use RTP as a transport protocol, RTSP
is not tied to RTP. is not tied to RTP.
RTSP assumes the existence of a presentation description format that RTSP assumes the existence of a presentation description format that
can express both static and temporal properties of a presentation can express both static and temporal properties of a presentation
containing several media streams. containing several media streams.
2 Notational Conventions 2 Notational Conventions
Since many of the definitions and syntax are identical to HTTP/1.1, Since many of the definitions and syntax are identical to HTTP/1.1,
this specification only points to the section where they are defined 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 rather than copying it. For brevity, [HX.Y] is to be taken to refer
Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]). 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 All the mechanisms specified in this document are described in both
prose and an augmented Backus-Naur form (BNF) similar to that used in 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 [H2.1]. It is described in detail in RFC 2234 [17], with the
difference that this RTSP specification maintains the ``1#'' notation difference that this RTSP specification maintains the "1#" notation
for comma-separated lists. for comma-separated lists.
In this draft, we use indented and smaller-type paragraphs to provide In this memo, we use indented and smaller-type paragraphs to provide
background and motivation. This is intended to give readers who were background and motivation. This is intended to give readers who were
not involved with the formulation of the specification an not involved with the formulation of the specification an
understanding of why things are the way that they are in RTSP. understanding of why things are the way that they are in RTSP.
3 Protocol Parameters 3 Protocol Parameters
3.1 RTSP Version 3.1 RTSP Version
[H3.1] applies, with HTTP replaced by RTSP. [H3.1] applies, with HTTP replaced by RTSP.
3.2 RTSP URL 3.2 RTSP URL
The ``rtsp'', ``rtspu'' and ``rtsps'' schemes are used to refer to The "rtsp" and "rtspu" schemes are used to refer to network resources
network resources via the RTSP protocol. This section defines the via the RTSP protocol. This section defines the scheme-specific
scheme-specific syntax and semantics for RTSP URLs. syntax and semantics for RTSP URLs.
rtsp_URL = ( "rtsp:" | "rtspu:" | "rtsps:" ) rtsp_URL = ( "rtsp:" | "rtspu:" )
"//" host [ ":" port ] [ abs_path ] "//" host [ ":" port ] [ abs_path ]
host = <A legal Internet host domain name of IP address host = <A legal Internet host domain name of IP address
(in dotted decimal form), as defined by Section 2.1 (in dotted decimal form), as defined by Section 2.1
of RFC 1123 \cite{rfc1123}> of RFC 1123 \cite{rfc1123}>
port = *DIGIT port = *DIGIT
abs_path is defined in [H3.2.1]. abs_path is defined in [H3.2.1].
Note that fragment and query identifiers do not have a well-defined Note that fragment and query identifiers do not have a well-defined
meaning at this time, with the interpretation left to the RTSP meaning at this time, with the interpretation left to the RTSP
server. server.
The scheme rtsp requires that commands are issued via a reliable The scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtspu identifies protocol (within the Internet, TCP), while the scheme rtspu identifies
an unreliable protocol (within the Internet, UDP). The scheme rtsps an unreliable protocol (within the Internet, UDP).
indicates that a TCP connection secured by TLS (RFC XXXX) [7] must be
used.
If the port is empty or not given, port 554 is assumed. The semantics 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 are that the identified resource can be controlled by RTSP at the
server listening for TCP (scheme ``rtsp'') connections or UDP (scheme server listening for TCP (scheme "rtsp") connections or UDP (scheme
``rtspu'') packets on that port of host, and the Request-URI for the "rtspu") packets on that port of host, and the Request-URI for the
resource is rtsp_URL. resource is rtsp_URL.
The use of IP addresses in URLs SHOULD be avoided whenever possible The use of IP addresses in URLs SHOULD be avoided whenever possible
(see RFC 1924 [19]). (see RFC 1924 [19]).
A presentation or a stream is identified by a textual media A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions [H3.2] of 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 URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of
streams, i.e., a presentation. Accordingly, requests described in streams, i.e., a presentation. Accordingly, requests described in
Section 10 can apply to either the whole presentation or an individual Section 10 can apply to either the whole presentation or an individual
stream within the presentation. Note that some request methods can stream within the presentation. Note that some request methods can
only be applied to streams, not presentations and vice versa. only be applied to streams, not presentations and vice versa.
For example, the RTSP URL: For example, the RTSP URL:
rtsp://media.example.com:554/twister/audiotrack rtsp://media.example.com:554/twister/audiotrack
identifies the audio stream within the presentation ``twister'', which identifies the audio stream within the presentation "twister", which
can be controlled via RTSP requests issued over a TCP connection to can be controlled via RTSP requests issued over a TCP connection to
port 554 of host media.example.com. port 554 of host media.example.com.
Also, the RTSP URL: Also, the RTSP URL:
rtsp://media.example.com:554/twister rtsp://media.example.com:554/twister
identifies the presentation ``twister'', which may be composed of identifies the presentation "twister", which may be composed of
audio and video streams. audio and video streams.
This does not imply a standard way to reference streams in URLs. This does not imply a standard way to reference streams in URLs.
The presentation description defines the hierarchical relationships The presentation description defines the hierarchical relationships
in the presentation and the URLs for the individual streams. A in the presentation and the URLs for the individual streams. A
presentation description may name a stream ``a.mov'' and the whole presentation description may name a stream "a.mov" and the whole
presentation ``b.mov''. presentation "b.mov".
The path components of the RTSP URL are opaque to the client and do The path components of the RTSP URL are opaque to the client and do
not imply any particular file system structure for the server. not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be used This decoupling also allows presentation descriptions to be used
with non-RTSP media control protocols simply by replacing the with non-RTSP media control protocols simply by replacing the
scheme in the URL. scheme in the URL.
3.3 Conference Identifiers 3.3 Conference Identifiers
Conference identifiers are opaque to RTSP and are encoded using Conference identifiers are opaque to RTSP and are encoded using
standard URI encoding methods (i.e., LWS is escaped with %). They can standard URI encoding methods (i.e., LWS is escaped with %). They can
contain any octet value. The conference identifier MUST be globally contain any octet value. The conference identifier MUST be globally
unique. For H.323, the conferenceID value is to be used. unique. For H.323, the conferenceID value is to be used.
conference-id = 1*xchar conference-id = 1*xchar
Conference identifiers are used to allow RTSP sessions to obtain Conference identifiers are used to allow RTSP sessions to obtain
parameters from multimedia conferences the media server is parameters from multimedia conferences the media server is
participating in. These conferences are created by protocols participating in. These conferences are created by protocols
outside the scope of this specification, e.g., H.323 [13] or SIP outside the scope of this specification, e.g., H.323 [13] or SIP
[12]. Instead of the RTSP client explicitly providing transport [12]. Instead of the RTSP client explicitly providing transport
information, for example, it asks the media server to use the information, for example, it asks the media server to use the
values in the conference description instead. values in the conference description instead.
3.4 Session Identifiers 3.4 Session Identifiers
Session identifiers are opaque strings of arbitrary length. Linear Session identifiers are opaque strings of arbitrary length. Linear
white space must be URL-escaped. A session identifier MUST be chosen 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 randomly and MUST be at least eight octets long to make guessing it
more difficult. (See Section 16.) more difficult. (See Section 16.)
session-id = 1*( ALPHA | DIGIT | safe ) session-id = 1*( ALPHA | DIGIT | safe )
3.5 SMPTE Relative Timestamps 3.5 SMPTE Relative Timestamps
A SMPTE relative timestamp expresses time relative to the start of A SMPTE relative timestamp expresses time relative to the start of
the clip. Relative timestamps are expressed as SMPTE time codes for the clip. Relative timestamps are expressed as SMPTE time codes for
frame-level access accuracy. The time code has the format frame-level access accuracy. The time code has the format
hours:minutes:seconds:frames.subframes, with the origin at the start hours:minutes:seconds:frames.subframes, with the origin at the start
of the clip. The default smpte format is``SMPTE 30 drop'' format, with 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 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 supported (such as "SMPTE 25") through the use of alternative use of
"smpte time". For the ``frames'' field in the time value can assume "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 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 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 00 and 01) of every minute, except every tenth minute. If the frame
value is zero, it may be omitted. Subframes are measured in value is zero, it may be omitted. Subframes are measured in
one-hundredth of a frame. one-hundredth of a frame.
smpte-range = smpte-type "=" smpte-time "-" [ smpte-time ] smpte-range = smpte-type "=" smpte-time "-" [ smpte-time ]
smpte-type = "smpte" | "smpte-30-drop" | "smpte-25" smpte-type = "smpte" | "smpte-30-drop" | "smpte-25"
; other timecodes may be added ; other timecodes may be added
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ] smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ]
[ "." 1*2DIGIT ] [ "." 1*2DIGIT ]
Examples: Examples:
smpte=10:12:33:20- smpte=10:12:33:20-
smpte=10:07:33- smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01 smpte=10:07:00-10:07:33:05.01
smpte-25=10:07:00-10:07:33:05.01 smpte-25=10:07:00-10:07:33:05.01
3.6 Normal Play Time 3.6 Normal Play Time
Normal play time (NPT) indicates the stream absolute position Normal play time (NPT) indicates the stream absolute position
relative to the beginning of the presentation. The timestamp consists relative to the beginning of the presentation. The timestamp consists
of a decimal fraction. The part left of the decimal may be expressed 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 in either seconds or hours, minutes, and seconds. The part right of
the decimal point measures fractions of a second. the decimal point measures fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative The beginning of a presentation corresponds to 0.0 seconds. Negative
values are not defined. The special constant now is defined as the 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. 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 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 viewer associates with a program. It is often digitally displayed on
VCR. NPT advances normally when in normal play mode (scale = 1), a VCR. NPT advances normally when in normal play mode (scale = 1),
advances at a faster rate when in fast scan forward (high positive advances at a faster rate when in fast scan forward (high positive
scale ratio), decrements when in scan reverse (high negative scale scale ratio), decrements when in scan reverse (high negative scale
ratio) and is fixed in pause mode. NPT is (logically) equivalent to ratio) and is fixed in pause mode. NPT is (logically) equivalent to
SMPTE time codes.'' [5] SMPTE time codes." [5]
npt-range = ( npt-time "-" [ npt-time ] ) | ( "-" npt-time ) npt-range = ( npt-time "-" [ npt-time ] ) | ( "-" npt-time )
npt-time = "now" | npt-sec | npt-hhmmss npt-time = "now" | npt-sec | npt-hhmmss
npt-sec = 1*DIGIT [ "." *DIGIT ] npt-sec = 1*DIGIT [ "." *DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ] npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
npt-hh = 1*DIGIT ; any positive number npt-hh = 1*DIGIT ; any positive number
npt-mm = 1*2DIGIT ; 0-59 npt-mm = 1*2DIGIT ; 0-59
npt-ss = 1*2DIGIT ; 0-59 npt-ss = 1*2DIGIT ; 0-59
Examples: Examples:
skipping to change at page 19, line 4 skipping to change at page 17, line 49
npt-sec = 1*DIGIT [ "." *DIGIT ] npt-sec = 1*DIGIT [ "." *DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ] npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
npt-hh = 1*DIGIT ; any positive number npt-hh = 1*DIGIT ; any positive number
npt-mm = 1*2DIGIT ; 0-59 npt-mm = 1*2DIGIT ; 0-59
npt-ss = 1*2DIGIT ; 0-59 npt-ss = 1*2DIGIT ; 0-59
Examples: Examples:
npt=123.45-125 npt=123.45-125
npt=12:05:35.3- npt=12:05:35.3-
npt=now- npt=now-
The syntax conforms to ISO 8601. The npt-sec notation is optimized The syntax conforms to ISO 8601. The npt-sec notation is optimized
for automatic generation, the ntp-hhmmss notation for consumption for automatic generation, the ntp-hhmmss notation for consumption
by human readers. The ``now'' constant allows clients to request to by human readers. The "now" constant allows clients to request to
receive the live feed rather than the stored or time-delayed receive the live feed rather than the stored or time-delayed
version. This is needed since neither absolute time nor zero time version. This is needed since neither absolute time nor zero time
are appropriate for this case. are appropriate for this case.
3.7 Absolute Time 3.7 Absolute Time
Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
Fractions of a second may be indicated. Fractions of a second may be indicated.
utc-range = "clock" "=" utc-time "-" [ utc-time ] utc-range = "clock" "=" utc-time "-" [ utc-time ]
utc-time = utc-date "T" utc-time "Z" utc-time = utc-date "T" utc-time "Z"
utc-date = 8DIGIT ; < YYYYMMDD > utc-date = 8DIGIT ; < YYYYMMDD >
utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction > utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC: UTC:
19961108T143720.25Z 19961108T143720.25Z
3.8 Option Tags 3.8 Option Tags
Option tags are unique identifiers used to designate new options in Option tags are unique identifiers used to designate new options in
RTSP. These tags are used in in Require (Section 12.32) and RTSP. These tags are used in Require (Section 12.32) and Proxy-
Proxy-Require (Section 12.27) header fields. Require (Section 12.27) header fields.
Syntax: Syntax:
option-tag = 1*xchar option-tag = 1*xchar
The creator of a new RTSP option should either prefix the option with 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 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 for a feature whose inventor can be reached at "foo.com"), or
register the new option with the Internet Assigned Numbers Authority register the new option with the Internet Assigned Numbers Authority
(IANA). (IANA).
3.8.1 Registering New Option Tags with IANA 3.8.1 Registering New Option Tags with IANA
When registering a new RTSP option, the following information should When registering a new RTSP option, the following information should
be provided: be provided:
* Name and description of option. The name may be of any length, but * Name and description of option. The name may be of any length,
SHOULD be no more than twenty characters long. The name MUST not but SHOULD be no more than twenty characters long. The name MUST
contain any spaces, control characters or periods. not contain any spaces, control characters or periods.
* Indication of who has change control over the option (for example, * Indication of who has change control over the option (for
IETF, ISO, ITU-T, other international standardization bodies, a example, IETF, ISO, ITU-T, other international standardization
consortium or a particular company or group of companies); bodies, a consortium or a particular company or group of
companies);
* A reference to a further description, if available, for example * A reference to a further description, if available, for example
(in order of preference) an RFC, a published paper, a patent (in order of preference) an RFC, a published paper, a patent
filing, a technical report, documented source code or a computer filing, a technical report, documented source code or a computer
manual; manual;
* For proprietary options, contact information (postal and email * For proprietary options, contact information (postal and email
address); address);
4 RTSP Message 4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set RTSP is a text-based protocol and uses the ISO 10646 character set in
in UTF-8 encoding (RFC XXXX [21]). Lines are terminated by CRLF, but UTF-8 encoding (RFC 2279 [21]). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by themselves receivers should be prepared to also interpret CR and LF by
as line terminators. themselves as line terminators.
Text-based protocols make it easier to add optional parameters in a Text-based protocols make it easier to add optional parameters in a
self-describing manner. Since the number of parameters and the self-describing manner. Since the number of parameters and the
frequency of commands is low, processing efficiency is not a frequency of commands is low, processing efficiency is not a
concern. Text-based protocols, if done carefully, also allow easy concern. Text-based protocols, if done carefully, also allow easy
implementation of research prototypes in scripting languages such implementation of research prototypes in scripting languages such
as Tcl, Visual Basic and Perl. as Tcl, Visual Basic and Perl.
The 10646 character set avoids tricky character set switching, but The 10646 character set avoids tricky character set switching, but
is invisible to the application as long as US-ASCII is being used. 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 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 directly into Unicode with a high-order octet of zero. ISO 8859-1
characters with the most-significant bit set are represented as characters with the most-significant bit set are represented as
1100001x 10xxxxxx. (See RFC XXXX [21]) 1100001x 10xxxxxx. (See RFC 2279 [21])
RTSP messages can be carried over any lower-layer transport protocol RTSP messages can be carried over any lower-layer transport protocol
that is 8-bit clean. that is 8-bit clean.
Requests contain methods, the object the method is operating upon and Requests contain methods, the object the method is operating upon and
parameters to further describe the method. Methods are idempotent, parameters to further describe the method. Methods are idempotent,
unless otherwise noted. Methods are also designed to require little or unless otherwise noted. Methods are also designed to require little
no state maintenance at the media server. or no state maintenance at the media server.
4.1 Message Types 4.1 Message Types
See [H4.1] See [H4.1]
4.2 Message Headers 4.2 Message Headers
See [H4.2] See [H4.2]
4.3 Message Body 4.3 Message Body
skipping to change at page 21, line 38 skipping to change at page 20, line 26
2. If a Content-Length header field (section 12.14) is present, 2. If a Content-Length header field (section 12.14) is present,
its value in bytes represents the length of the message-body. its value in bytes represents the length of the message-body.
If this header field is not present, a value of zero is If this header field is not present, a value of zero is
assumed. assumed.
3. By the server closing the connection. (Closing the connection 3. By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a that would leave no possibility for the server to send back a
response.) response.)
Note that RTSP does not (at present) support the HTTP/1.1 ``chunked'' Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
transfer coding(see [H3.6]) and requires the presence of the transfer coding(see [H3.6]) and requires the presence of the
Content-Length header field. Content-Length header field.
Given the moderate length of presentation descriptions returned, Given the moderate length of presentation descriptions returned,
the server should always be able to determine its length, even if the server should always be able to determine its length, even if
it is generated dynamically, making the chunked transfer encoding it is generated dynamically, making the chunked transfer encoding
unnecessary. Even though Content-Length must be present if there is unnecessary. Even though Content-Length must be present if there is
any entity body, the rules ensure reasonable behavior even if the any entity body, the rules ensure reasonable behavior even if the
length is not given explicitly. length is not given explicitly.
5 General Header Fields 5 General Header Fields
See [H4.5], except that Pragma, Transfer-Encoding and Upgrade See [H4.5], except that Pragma, Transfer-Encoding and Upgrade headers
headers are not defined: are not defined:
general-header = Cache-Control ; Section 12.8 general-header = Cache-Control ; Section 12.8
| Connection ; Section 12.10 | Connection ; Section 12.10
| Date ; Section 12.18 | Date ; Section 12.18
| Via ; Section 12.43 | Via ; Section 12.43
6 Request 6 Request
A request message from a client to a server or vice versa includes, 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 within the first line of that message, the method to be applied to
resource, the identifier of the resource, and the protocol version in the resource, the identifier of the resource, and the protocol
use. version in use.
Request = Request-Line ; Section 6.1 Request = Request-Line ; Section 6.1
*( general-header ; Section 5 *( general-header ; Section 5
| request-header ; Section 6.2 | request-header ; Section 6.2
| entity-header ) ; Section 8.1 | entity-header ) ; Section 8.1
CRLF CRLF
[ message-body ] ; Section 4.3 [ message-body ] ; Section 4.3
6.1 Request Line 6.1 Request Line
skipping to change at page 23, line 24 skipping to change at page 21, line 48
| Accept-Encoding ; Section 12.2 | Accept-Encoding ; Section 12.2
| Accept-Language ; Section 12.3 | Accept-Language ; Section 12.3
| Authorization ; Section 12.5 | Authorization ; Section 12.5
| From ; Section 12.20 | From ; Section 12.20
| If-Modified-Since ; Section 12.23 | If-Modified-Since ; Section 12.23
| Range ; Section 12.29 | Range ; Section 12.29
| Referer ; Section 12.30 | Referer ; Section 12.30
| User-Agent ; Section 12.41 | User-Agent ; Section 12.41
Note that in contrast to HTTP/1.1 [2], RTSP requests always contain 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 the absolute URL (that is, including the scheme, host and port)
than just the absolute path. rather than just the absolute path.
HTTP/1.1 requires servers to understand the absolute URL, but HTTP/1.1 requires servers to understand the absolute URL, but
clients are supposed to use the Host request header. This is purely clients are supposed to use the Host request header. This is purely
needed for backward-compatibility with HTTP/1.0 servers, a needed for backward-compatibility with HTTP/1.0 servers, a
consideration that does not apply to RTSP. consideration that does not apply to RTSP.
The asterisk "*" in the Request-URI means that the request does not 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 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. allowed when the method used does not necessarily apply to a
One example would be: resource. One example would be:
OPTIONS * RTSP/1.0 OPTIONS * RTSP/1.0
7 Response 7 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version. [H6] applies except that HTTP-Version is replaced by RTSP-Version.
Also, RTSP defines additional status codes and does not define some Also, RTSP defines additional status codes and does not define some
HTTP codes. The valid response codes and the methods they can be used HTTP codes. The valid response codes and the methods they can be used
with are defined in Table 1. with are defined in Table 1.
After receiving and interpreting a request message, the recipient After receiving and interpreting a request message, the recipient
responds with an RTSP response message. responds with an RTSP response message.
Response = Status-Line ; Section 7.1 Response = Status-Line ; Section 7.1
*( general-header ; Section 5 *( general-header ; Section 5
| response-header ; Section 7.1.2 | response-header ; Section 7.1.2
| entity-header ) ; Section 8.1 | entity-header ) ; Section 8.1
CRLF CRLF
[ message-body ] ; Section 4.3 [ message-body ] ; Section 4.3
7.1 Status-Line 7.1 Status-Line
The first line of a Response message is the Status-Line, consisting The first line of a Response message is the Status-Line, consisting
of the protocol version followed by a numeric status code, and the of the protocol version followed by a numeric status code, and the
textual phrase associated with the status code, with each element textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the final separated by SP characters. No CR or LF is allowed except in the
CRLF sequence. final CRLF sequence.
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
7.1.1 Status Code and Reason Phrase 7.1.1 Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully attempt to understand and satisfy the request. These codes are fully
defined in Section 11. The Reason-Phrase is intended to give a short defined in Section 11. The Reason-Phrase is intended to give a short
textual description of the Status-Code. The Status-Code is intended textual description of the Status-Code. The Status-Code is intended
for use by automata and the Reason-Phrase is intended for the human for use by automata and the Reason-Phrase is intended for the human
user. The client is not required to examine or display the user. The client is not required to examine or display the Reason-
Reason-Phrase. Phrase.
The first digit of the Status-Code defines the class of response. The 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 last two digits do not have any categorization role. There are 5
values for the first digit: values for the first digit:
* 1xx: Informational - Request received, continuing process * 1xx: Informational - Request received, continuing process
* 2xx: Success - The action was successfully received, understood, * 2xx: Success - The action was successfully received, understood,
and accepted and accepted
* 3xx: Redirection - Further action must be taken in order to * 3xx: Redirection - Further action must be taken in order to
complete the request complete the request
* 4xx: Client Error - The request contains bad syntax or cannot be * 4xx: Client Error - The request contains bad syntax or cannot be
fulfilled fulfilled
* 5xx: Server Error - The server failed to fulfill an apparently * 5xx: Server Error - The server failed to fulfill an apparently
valid request valid request
The individual values of the numeric status codes defined for The individual values of the numeric status codes defined for
RTSP/1.0, and an example set of corresponding Reason-Phrase's, are RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
presented below. The reason phrases listed here are only recommended - presented below. The reason phrases listed here are only recommended
they may be replaced by local equivalents without affecting the - they may be replaced by local equivalents without affecting the
protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and
adds RTSP-specific status codes starting at x50 to avoid conflicts adds RTSP-specific status codes starting at x50 to avoid conflicts
with newly defined HTTP status codes. with newly defined HTTP status codes.
Status-Code = "100" ; Continue Status-Code = "100" ; Continue
| "200" ; OK | "200" ; OK
| "201" ; Created | "201" ; Created
| "250" ; Low on Storage Space | "250" ; Low on Storage Space
| "300" ; Multiple Choices | "300" ; Multiple Choices
| "301" ; Moved Permanently | "301" ; Moved Permanently
skipping to change at page 27, line 14 skipping to change at page 25, line 14
extension-code = 3DIGIT extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF> Reason-Phrase = *<TEXT, excluding CR, LF>
RTSP status codes are extensible. RTSP applications are not required RTSP status codes are extensible. RTSP applications are not required
to understand the meaning of all registered status codes, though such to understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an unrecognized x00 status code of that class, with the exception that an
response MUST NOT be cached. For example, if an unrecognized status unrecognized response MUST NOT be cached. For example, if an
code of 431 is received by the client, it can safely assume that there unrecognized status code of 431 is received by the client, it can
was something wrong with its request and treat the response as if it safely assume that there was something wrong with its request and
had received a 400 status code. In such cases, user agents SHOULD treat the response as if it had received a 400 status code. In such
present to the user the entity returned with the response, since that cases, user agents SHOULD present to the user the entity returned
entity is likely to include human-readable information which will with the response, since that entity is likely to include human-
explain the unusual status. readable information which will explain the unusual status.
Code reason Code reason
100 Continue all 100 Continue all
200 OK all 200 OK all
201 Created RECORD 201 Created RECORD
250 Low on Storage Space RECORD 250 Low on Storage Space RECORD
300 Multiple Choices all 300 Multiple Choices all
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500 Internal Server Error all 500 Internal Server Error all
501 Not Implemented all 501 Not Implemented all
502 Bad Gateway all 502 Bad Gateway all
503 Service Unavailable all 503 Service Unavailable all
504 Gateway Timeout all 504 Gateway Timeout all
505 RTSP Version Not Supported all 505 RTSP Version Not Supported all
551 Option not support all 551 Option not support all
Table 1: Status codes and their usage with RTSP methods Table 1: Status codes and their usage with RTSP methods
7.1.2 Response Header Fields 7.1.2 Response Header Fields
The response-header fields allow the request recipient to pass The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the server the Status-Line. These header fields give information about the
and about further access to the resource identified by the server and about further access to the resource identified by the
Request-URI. Request-URI.
response-header = Location ; Section 12.25 response-header = Location ; Section 12.25
| Proxy-Authenticate ; Section 12.26 | Proxy-Authenticate ; Section 12.26
| Public ; Section 12.28 | Public ; Section 12.28
| Retry-After ; Section 12.31 | Retry-After ; Section 12.31
| Server ; Section 12.36 | Server ; Section 12.36
| Vary ; Section 12.42 | Vary ; Section 12.42
| WWW-Authenticate ; Section 12.44 | WWW-Authenticate ; Section 12.44
Response-header field names can be extended reliably only in Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of experimental header fields MAY be given the semantics of response-
response-header fields if all parties in the communication recognize header fields if all parties in the communication recognize them to
them to be response-header fields. Unrecognized header fields are be response-header fields. Unrecognized header fields are treated as
treated as entity-header fields. entity-header fields.
8 Entity 8 Entity
Request and Response messages MAY transfer an entity if not Request and Response messages MAY transfer an entity if not otherwise
otherwise restricted by the request method or response status code. An restricted by the request method or response status code. An entity
entity consists of entity-header fields and an entity-body, although consists of entity-header fields and an entity-body, although some
some responses will only include the entity-headers. responses will only include the entity-headers.
In this section, both sender and recipient refer to either the client In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity. or the server, depending on who sends and who receives the entity.
8.1 Entity Header Fields 8.1 Entity Header Fields
Entity-header fields define optional metainformation about the Entity-header fields define optional metainformation about the
entity-body or, if no body is present, about the resource identified entity-body or, if no body is present, about the resource identified
by the request. by the request.
entity-header = Allow ; Section 12.4 entity-header = Allow ; Section 12.4
| Content-Base ; Section 12.11 | Content-Base ; Section 12.11
| Content-Encoding ; Section 12.12 | Content-Encoding ; Section 12.12
| Content-Language ; Section 12.13 | Content-Language ; Section 12.13
| Content-Length ; Section 12.14 | Content-Length ; Section 12.14
| Content-Location ; Section 12.15 | Content-Location ; Section 12.15
| Content-Type ; Section 12.16 | Content-Type ; Section 12.16
skipping to change at page 30, line 16 skipping to change at page 28, line 11
to be defined without changing the protocol, but these fields cannot to be defined without changing the protocol, but these fields cannot
be assumed to be recognizable by the recipient. Unrecognized header be assumed to be recognizable by the recipient. Unrecognized header
fields SHOULD be ignored by the recipient and forwarded by proxies. fields SHOULD be ignored by the recipient and forwarded by proxies.
8.2 Entity Body 8.2 Entity Body
See [H7.2] See [H7.2]
9 Connections 9 Connections
RTSP requests can be transmitted in several different ways: RTSP requests can be transmitted in several different ways:
* persistent transport connections used for several request-response * persistent transport connections used for several
transactions; request-response transactions;
* one connection per request/response transaction; * one connection per request/response transaction;
* connectionless mode. * connectionless mode.
The type of transport connection is defined by the RTSP URI The type of transport connection is defined by the RTSP URI (Section
(Section 3.2). For the scheme ``rtsp'', a persistent connection is 3.2). For the scheme "rtsp", a persistent connection is assumed,
assumed, while the scheme ``rtspu'' calls for RTSP requests to be sent while the scheme "rtspu" calls for RTSP requests to be sent without
without setting up a connection. setting up a connection.
Unlike HTTP, RTSP allows the media server to send requests to the Unlike HTTP, RTSP allows the media server to send requests to the
media client. However, this is only supported for persistent media client. However, this is only supported for persistent
connections, as the media server otherwise has no reliable way of connections, as the media server otherwise has no reliable way of
reaching the client. Also, this is the only way that requests from reaching the client. Also, this is the only way that requests from
media server to client are likely to traverse firewalls. media server to client are likely to traverse firewalls.
9.1 Pipelining 9.1 Pipelining
A client that supports persistent connections or connectionless mode A client that supports persistent connections or connectionless mode
MAY ``pipeline'' its requests (i.e., send multiple requests without MAY "pipeline" its requests (i.e., send multiple requests without
waiting for each response). A server MUST send its responses to those waiting for each response). A server MUST send its responses to those
requests in the same order that the requests were received. requests in the same order that the requests were received.
9.2 Reliability and Acknowledgements 9.2 Reliability and Acknowledgements
Requests are acknowledged by the receiver unless they are sent to a Requests are acknowledged by the receiver unless they are sent to a
multicast group. If there is no acknowledgement, the sender may resend multicast group. If there is no acknowledgement, the sender may
the same message after a timeout of one round-trip time (RTT). The resend the same message after a timeout of one round-trip time (RTT).
round-trip time is estimated as in TCP (RFC 1123) [18], with an 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 initial round-trip value of 500 ms. An implementation MAY cache the
last RTT measurement as the initial value for future connections. last RTT measurement as the initial value for future connections.
If a reliable transport protocol is used to carry RTSP, requests If a reliable transport protocol is used to carry RTSP, requests MUST
SHOULD NOT be retransmitted; the RTSP application SHOULD instead rely NOT be retransmitted; the RTSP application MUST instead rely on the
on the underlying transport to provide reliability. underlying transport to provide reliability.
If both the underlying reliable transport such as TCP and the RTSP If both the underlying reliable transport such as TCP and the RTSP
application retransmit requests, it is possible that each packet application retransmit requests, it is possible that each packet
loss results in two retransmissions. The receiver cannot typically loss results in two retransmissions. The receiver cannot typically
take advantage of the application-layer retransmission since the take advantage of the application-layer retransmission since the
transport stack will not deliver the application-layer transport stack will not deliver the application-layer
retransmission before the first attempt has reached the receiver. retransmission before the first attempt has reached the receiver.
If the packet loss is caused by congestion, multiple If the packet loss is caused by congestion, multiple
retransmissions at different layers will exacerbate the congestion. retransmissions at different layers will exacerbate the congestion.
If RTSP is used over a small-RTT LAN, standard procedures for If RTSP is used over a small-RTT LAN, standard procedures for
optimizing inital TCP round trip estimates, such as those used in optimizing initial TCP round trip estimates, such as those used in
T/TCP (RFC 1644) [22], can be beneficial. T/TCP (RFC 1644) [22], can be beneficial.
The Timestamp header (Section 12.38) is used to avoid the The Timestamp header (Section 12.38) is used to avoid the
retransmission ambiguity problem [23, p. 301] and obviates the need retransmission ambiguity problem [23, p. 301] and obviates the need
for Karn's algorithm. for Karn's algorithm.
Each request carries a sequence number in the CSeq header Each request carries a sequence number in the CSeq header (Section
(Section 12.17), which is incremented by one for each distinct request 12.17), which is incremented by one for each distinct request
transmitted. If a request is repeated because of lack of transmitted. If a request is repeated because of lack of
acknowledgement, the request MUST carry the original sequence number acknowledgement, the request MUST carry the original sequence number
(i.e. sequence number is not incremented). (i.e., the sequence number is not incremented).
Systems implementing RTSP MUST support carrying RTSP over TCP and MAY 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 support UDP. The default port for the RTSP server is 554 for both UDP
and TCP. and TCP.
A number of RTSP packets destined for the same control end point may 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 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 stream. RTSP data MAY be interleaved with RTP and RTCP packets.
HTTP, an RTSP message MUST contain a Content-Length header whenever Unlike HTTP, an RTSP message MUST contain a Content-Length header
that message contains a payload. Otherwise, an RTSP packet is whenever that message contains a payload. Otherwise, an RTSP packet
terminated with an empty line immediately following the last message is terminated with an empty line immediately following the last
header. message header.
10 Method Definitions 10 Method Definitions
The method token indicates the method to be performed on the The method token indicates the method to be performed on the resource
resource identified by the Request-URI. The method is case-sensitive. identified by the Request-URI. The method is case-sensitive. New
New methods may be defined in the future. Method names may not start methods may be defined in the future. Method names may not start with
with a $ character (decimal 24) and must be a token. Methods are a $ character (decimal 24) and must be a token. Methods are
summarized in Table 2. summarized in Table 2.
method direction object requirement method direction object requirement
DESCRIBE C->S P,S recommended DESCRIBE C->S P,S recommended
ANNOUNCE C->S, S->C P,S optional ANNOUNCE C->S, S->C P,S optional
GET_PARAMETER C->S, S->C P,S optional GET_PARAMETER C->S, S->C P,S optional
OPTIONS C->S, S->C P,S required OPTIONS C->S, S->C P,S required
(S->C: optional) (S->C: optional)
PAUSE C->S P,S recommended PAUSE C->S P,S recommended
PLAY C->S P,S required PLAY C->S P,S required
skipping to change at page 32, line 30 skipping to change at page 30, line 30
objects (P: presentation, S: stream) they operate on objects (P: presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommended, but not required in that a Notes on Table 2: PAUSE is recommended, but not required in that a
fully functional server can be built that does not support this fully functional server can be built that does not support this
method, for example, for live feeds. If a server does not support a method, for example, for live feeds. If a server does not support a
particular method, it MUST return "501 Not Implemented" and a client particular method, it MUST return "501 Not Implemented" and a client
SHOULD not try this method again for this server. SHOULD not try this method again for this server.
10.1 OPTIONS 10.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An 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 request may be issued at any time, e.g., if the client is about to
a nonstandard request. It does not influence server state. try a nonstandard request. It does not influence server state.
Example: Example:
C->S: OPTIONS * RTSP/1.0 C->S: OPTIONS * RTSP/1.0
CSeq: 1 CSeq: 1
Require: implicit-play Require: implicit-play
Proxy-Require: gzipped-messages Proxy-Require: gzipped-messages
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are necessarily fictional features (one would hope Note that these are necessarily fictional features (one would hope
that we would not purposefully overlook a truly useful feature just so that we would not purposefully overlook a truly useful feature just
that we could have a strong example in this section). so that we could have a strong example in this section).
10.2 DESCRIBE 10.2 DESCRIBE
The DESCRIBE method retrieves the description of a presentation or The DESCRIBE method retrieves the description of a presentation or
media object identified by the request URL from a server. It may use media object identified by the request URL from a server. It may use
the Accept header to specify the description formats that the client the Accept header to specify the description formats that the client
understands. The server responds with a description of the requested understands. The server responds with a description of the requested
resource. The DESCRIBE reply-response pair constitutes the media resource. The DESCRIBE reply-response pair constitutes the media
initialization phase of RTSP. initialization phase of RTSP.
Example: Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312 CSeq: 312
skipping to change at page 34, line 27 skipping to change at page 32, line 23
* via RTSP's DESCRIBE method; * via RTSP's DESCRIBE method;
* via some other protocol (HTTP, email attachment, etc.); * via some other protocol (HTTP, email attachment, etc.);
* via the command line or standard input (thus working as a browser * via the command line or standard input (thus working as a browser
helper application launched with an SDP file or other media helper application launched with an SDP file or other media
initialization format). initialization format).
In the interest of practical interoperability, it is highly In the interest of practical interoperability, it is highly
recommended that minimal servers support the DESCRIBE method, and recommended that minimal servers support the DESCRIBE method, and
highly recommended that minimal clients support the ability to act highly recommended that minimal clients support the ability to act
as a ``helper application'' that accepts a media initialization as a "helper application" that accepts a media initialization file
file from standard input, command line, and/or other means that are from standard input, command line, and/or other means that are
appropriate to the operating environment of the client. appropriate to the operating environment of the client.
10.3 ANNOUNCE 10.3 ANNOUNCE
The ANNOUNCE method serves two purposes: The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a When sent from client to server, ANNOUNCE posts the description of a
presentation or media object identified by the request URL to a presentation or media object identified by the request URL to a
server. When sent from server to client, ANNOUNCE updates the session server. When sent from server to client, ANNOUNCE updates the session
description in real-time. description in real-time.
If a new media stream is added to a presentation (e.g., during a live If a new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent presentation), the whole presentation description should be sent
again, rather than just the additional components, so that components again, rather than just the additional components, so that components
can be deleted. can be deleted.
Example: Example:
C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0 C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312 CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Session: 4711 Session: 47112344
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 332 Content-Length: 332
v=0 v=0
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4 o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
s=SDP Seminar s=SDP Seminar
i=A Seminar on the session description protocol i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley) e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127 c=IN IP4 224.2.17.12/127
t=2873397496 2873404696 t=2873397496 2873404696
a=recvonly a=recvonly
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31 m=video 2232 RTP/AVP 31
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 312 CSeq: 312
10.4 SETUP 10.4 SETUP
The SETUP request for a URI specifies the transport mechanism to be 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 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 stream that is already playing to change transport parameters, which
server MAY allow. If it does not allow this, it MUST respond with 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 error "455 Method Not Valid In This State". For the benefit of any
intervening firewalls, a client must indicate the transport parameters intervening firewalls, a client must indicate the transport
even if it has no influence over these parameters, for example, where parameters even if it has no influence over these parameters, for
the server advertises a fixed multicast address. example, where the server advertises a fixed multicast address.
Since SETUP includes all transport initialization information, Since SETUP includes all transport initialization information,
firewalls and other intermediate network devices (which need this firewalls and other intermediate network devices (which need this
information) are spared the more arduous task of parsing the information) are spared the more arduous task of parsing the
DESCRIBE response, which has been reserved for media DESCRIBE response, which has been reserved for media
initialization. initialization.
The Transport header specifies the transport parameters acceptable to The Transport header specifies the transport parameters acceptable to
the client for data transmission; the response will contain the the client for data transmission; the response will contain the
transport parameters selected by the server. transport parameters selected by the server.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302 CSeq: 302
Transport: RTP/AVP;unicast;client_port=4588-4589 Transport: RTP/AVP;unicast;client_port=4588-4589
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 302 CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Session: 4711 Session: 47112344
Transport: RTP/AVP;unicast; Transport: RTP/AVP;unicast;
client_port=4588-4589;server_port=6256-6257 client_port=4588-4589;server_port=6256-6257
The server generates session identifiers in response to SETUP
requests. If a SETUP request to a server includes a session
identifier, the server MUST bundle this setup request into the
existing session or return error "459 Aggregate Operation Not
Allowed" (see Section 11.3.10).
10.5 PLAY 10.5 PLAY
The PLAY method tells the server to start sending data via the The PLAY method tells the server to start sending data via the
mechanism specified in SETUP. A client MUST NOT issue a PLAY request mechanism specified in SETUP. A client MUST NOT issue a PLAY request
until any outstanding SETUP requests have been acknowledged as until any outstanding SETUP requests have been acknowledged as
successful. successful.
The PLAY request positions the normal play time to the beginning of The PLAY request positions the normal play time to the beginning of
the range specified and delivers stream data until the end of the the range specified and delivers stream data until the end of the
range is reached. PLAY requests may be pipelined (queued); a server range is reached. PLAY requests may be pipelined (queued); a server
MUST queue PLAY requests to be executed in order. That is, a PLAY MUST queue PLAY requests to be executed in order. That is, a PLAY
request arriving while a previous PLAY request is still active is request arriving while a previous PLAY request is still active is
delayed until the first has been completed. delayed until the first has been completed.
This allows precise editing. This allows precise editing.
For example, regardless of how closely spaced the two PLAY requests in For example, regardless of how closely spaced the two PLAY requests
the example below arrive, the server will first play seconds 10 in the example below arrive, the server will first play seconds 10
through 15, then, immediately following, seconds 20 to 25, and finally through 15, then, immediately following, seconds 20 to 25, and
seconds 30 through the end. finally seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 835 CSeq: 835
Session: 12345678
Range: npt=10-15 Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 836 CSeq: 836
Session: 12345678
Range: npt=20-25 Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 837 CSeq: 837
Session: 12345678
Range: npt=30- Range: npt=30-
See the description of the PAUSE request for further examples. See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It starts playing a 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 from the beginning unless the stream has been paused. If a
stream has been paused via PAUSE, stream delivery resumes at the pause stream has been paused via PAUSE, stream delivery resumes at the
point. If a stream is playing, such a PLAY request causes no further pause point. If a stream is playing, such a PLAY request causes no
action and can be used by the client to test server liveness. further action and can be used by the client to test server liveness.
The Range header may also contain a time parameter. This parameter The Range header may also contain a time parameter. This parameter
specifies a time in UTC at which the playback should start. If the specifies a time in UTC at which the playback should start. If the
message is received after the specified time, playback is started message is received after the specified time, playback is started
immediately. The time parameter may be used to aid in synchronization immediately. The time parameter may be used to aid in synchronization
of streams obtained from different sources. of streams obtained from different sources.
For a on-demand stream, the server replies with the actual range that 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 will be played back. This may differ from the requested range if
alignment of the requested range to valid frame boundaries is required alignment of the requested range to valid frame boundaries is
for the media source. If no range is specified in the request, the required for the media source. If no range is specified in the
current position is returned in the reply. The unit of the range in request, the current position is returned in the reply. The unit of
the reply is the same as that in the request. the range in the reply is the same as that in the request.
After playing the desired range, the presentation is automatically After playing the desired range, the presentation is automatically
paused, as if a PAUSE request had been issued. paused, as if a PAUSE request had been issued.
The following example plays the whole presentation starting at SMPTE 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 time code 0:10:20 until the end of the clip. The playback is to start
at 15:36 on 23 Jan 1997. at 15:36 on 23 Jan 1997.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
CSeq: 833 CSeq: 833
Session: 12345678
Range: smpte=0:10:20-;time=19970123T153600Z Range: smpte=0:10:20-;time=19970123T153600Z
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 833 CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Range: smpte=0:10:22-;time=19970123T153600Z Range: smpte=0:10:22-;time=19970123T153600Z
For playing back a recording of a live presentation, it may be For playing back a recording of a live presentation, it may be
desirable to use clock units: desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0 C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
CSeq: 835 CSeq: 835
Session: 12345678
Range: clock=19961108T142300Z-19961108T143520Z Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 835 CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the npt format A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats. and MAY support the clock and smpte formats.
10.6 PAUSE 10.6 PAUSE
The PAUSE request causes the stream delivery to be interrupted The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only playback (halted) temporarily. If the request URL names a stream, only
and recording of that stream is halted. For example, for audio, this playback and recording of that stream is halted. For example, for
is equivalent to muting. If the request URL names a presentation or audio, this is equivalent to muting. If the request URL names a
group of streams, delivery of all currently active streams within the presentation or group of streams, delivery of all currently active
presentation or group is halted. After resuming playback or recording, streams within the presentation or group is halted. After resuming
synchronization of the tracks MUST be maintained. Any server resources playback or recording, synchronization of the tracks MUST be
are kept, though servers MAY close the session and free resources maintained. Any server resources are kept, though servers MAY close
after being paused for the duration specified with the timeout the session and free resources after being paused for the duration
parameter of the Session header in the SETUP message. specified with the timeout parameter of the Session header in the
SETUP message.
Example: Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834 CSeq: 834
Session: 1234 Session: 12345678
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 834 CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
The PAUSE request may contain a Range header specifying when the The PAUSE request may contain a Range header specifying when the
stream or presentation is to be halted. The header must contain stream or presentation is to be halted. We refer to this point as the
exactly one value rather than a time range. The normal play time for "pause point". The header must contain exactly one value rather than
the stream is set to that value. The pause request becomes effective a time range. The normal play time for the stream is set to the pause
the first time the server is encountering the time point specified in point. The pause request becomes effective the first time the server
any of the currently pending PLAY requests. If the Range header is encountering the time point specified in any of the currently
specifies a time outside any currently pending PLAY requests, the pending PLAY requests. If the Range header specifies a time outside
error ``457 Invalid Range'' is returned. If this header is missing, any currently pending PLAY requests, the error "457 Invalid Range" is
stream delivery is interrupted immediately on receipt of the message. 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 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 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 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 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 first play request, the server stops immediately. If the pause
is for NPT 16, the server stops after completing the first play request is for NPT 16, the server stops after completing the first
request and discards the second play request. play request and discards the second play request.
As another example, if a server has received requests to play ranges 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 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 request for NPT=14 would take effect while the server plays the first
range, with the second PLAY request effectively being ignored, range, with the second PLAY request effectively being ignored,
assuming the PAUSE request arrives before the server has started assuming the PAUSE request arrives before the server has started
playing the second, overlapping range. Regardless of when the PAUSE playing the second, overlapping range. Regardless of when the PAUSE
request arrives, it sets the NPT to 14. request arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the 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 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 is assumed that the client has discarded data after that point. This
ensures continuous pause/play cycling without gaps. ensures continuous pause/play cycling without gaps.
10.7 TEARDOWN 10.7 TEARDOWN
The TEARDOWN request stops the stream delivery for the given URI, The TEARDOWN request stops the stream delivery for the given URI,
freeing the resources associated with it. If the URI is the freeing the resources associated with it. If the URI is the
presentation URI for this presentation, any RTSP session identifier presentation URI for this presentation, any RTSP session identifier
associated with the session is no longer valid. Unless all transport associated with the session is no longer valid. Unless all transport
parameters are defined by the session description, a SETUP request has parameters are defined by the session description, a SETUP request
to be issued before the session can be played again. has to be issued before the session can be played again.
Example: Example:
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 892 CSeq: 892
Session: 1234 Session: 12345678
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 892 CSeq: 892
10.8 GET_PARAMETER 10.8 GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a The GET_PARAMETER request retrieves the value of a parameter of a
presentation or stream specified in the URI. The content of the reply presentation or stream specified in the URI. The content of the reply
and response is left to the implementation. GET_PARAMETER with no and response is left to the implementation. GET_PARAMETER with no
entity body may be used to test client or server liveness (``ping''). entity body may be used to test client or server liveness ("ping").
Example: Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 431 CSeq: 431
Content-Type: text/parameters Content-Type: text/parameters
Session: 1234 Session: 12345678
Content-Length: 15 Content-Length: 15
packets_received packets_received
jitter jitter
C->S: RTSP/1.0 200 OK C->S: RTSP/1.0 200 OK
CSeq: 431 CSeq: 431
Content-Length: 46 Content-Length: 46
Content-Type: text/parameters Content-Type: text/parameters
packets_received: 10 packets_received: 10
jitter: 0.3838 jitter: 0.3838
The ``text/parameters'' section is only an example type for The "text/parameters" section is only an example type for
parameter. This method is intentionally loosely defined with the parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be intention that the reply content and response content will be
defined after further experimentation. defined after further experimentation.
10.9 SET_PARAMETER 10.9 SET_PARAMETER
This method requests to set the value of a parameter for a This method requests to set the value of a parameter for a
presentation or stream specified by the URI. presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client A request SHOULD only contain a single parameter to allow the client
to determine why a particular request failed. If the request contains to determine why a particular request failed. If the request contains
several parameters, the server MUST only act on the request if all of several parameters, the server MUST only act on the request if all of
the parameters can be set successfully. A server MUST allow a the parameters can be set successfully. A server MUST allow a
parameter to be set repeatedly to the same value, but it MAY disallow parameter to be set repeatedly to the same value, but it MAY disallow
changing parameter values. changing parameter values.
Note: transport parameters for the media stream MUST only be set with Note: transport parameters for the media stream MUST only be set with
the SETUP command. the SETUP command.
Restricting setting transport parameters to SETUP is for the Restricting setting transport parameters to SETUP is for the
benefit of firewalls. benefit of firewalls.
The parameters are split in a fine-grained fashion so that there The parameters are split in a fine-grained fashion so that there
can be more meaningful error indications. However, it may make can be more meaningful error indications. However, it may make
sense to allow the setting of several parameters if an atomic sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client does setting is desirable. Imagine device control where the client does
not want the camera to pan unless it can also tilt to the right not want the camera to pan unless it can also tilt to the right
angle at the same time. angle at the same time.
skipping to change at page 41, line 28 skipping to change at page 39, line 10
barparam: barstuff barparam: barstuff
S->C: RTSP/1.0 451 Invalid Parameter S->C: RTSP/1.0 451 Invalid Parameter
CSeq: 421 CSeq: 421
Content-length: 10 Content-length: 10
Content-type: text/parameters Content-type: text/parameters
barparam barparam
The ``text/parameters'' section is only an example type for The "text/parameters" section is only an example type for
parameter. This method is intentionally loosely defined with the parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be intention that the reply content and response content will be
defined after further experimentation. defined after further experimentation.
10.10 REDIRECT 10.10 REDIRECT
A redirect request informs the client that it must connect to A redirect request informs the client that it must connect to another
another server location. It contains the mandatory header Location, server location. It contains the mandatory header Location, which
which indicates that the client should issue requests for that URL. It indicates that the client should issue requests for that URL. It may
may contain the parameter Range, which indicates when the redirection contain the parameter Range, which indicates when the redirection
takes effect. If the client wants to continue to send or receive media takes effect. If the client wants to continue to send or receive
for this URI, the client MUST issue a TEARDOWN request for the current media for this URI, the client MUST issue a TEARDOWN request for the
session and a SETUP for the new session at the designated host. 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 This example request redirects traffic for this URI to the new server
at the given play time: at the given play time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0 S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 732 CSeq: 732
Location: rtsp://bigserver.com:8001 Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z- Range: clock=19960213T143205Z-
10.11 RECORD 10.11 RECORD
This method initiates recording a range of media data according to This method initiates recording a range of media data according to
the presentation description. The timestamp reflects start and end the presentation description. The timestamp reflects start and end
time (UTC). If no time range is given, use the start or end time time (UTC). If no time range is given, use the start or end time
provided in the presentation description. If the session has already provided in the presentation description. If the session has already
started, commence recording immediately. started, commence recording immediately.
The server decides whether to store the recorded data under the 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 or another URI. If the server does not use the request-
request-URI, the response SHOULD be 201 (Created) and contain an URI, the response SHOULD be 201 (Created) and contain an entity which
entity which describes the status of the request and refers to the new describes the status of the request and refers to the new resource,
resource, and a Location header. and a Location header.
A media server supporting recording of live presentations MUST support A media server supporting recording of live presentations MUST
the clock range format; the smpte format does not make sense. support the clock range format; the smpte format does not make sense.
In this example, the media server was previously invited to the In this example, the media server was previously invited to the
conference indicated. conference indicated.
C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0 C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
CSeq: 954 CSeq: 954
Session: 1234 Session: 12345678
Conference: 128.16.64.19/32492374 Conference: 128.16.64.19/32492374
10.12 Embedded (Interleaved) Binary Data 10.12 Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server Certain firewall designs and other circumstances may force a server
to interleave RTSP methods and stream data. This interleaving should to interleave RTSP methods and stream data. This interleaving should
generally be avoided unless necessary since it complicates client and generally be avoided unless necessary since it complicates client and
server operation and imposes additional overhead. Interleaved binary server operation and imposes additional overhead. Interleaved binary
data SHOULD only be used if RTSP is carried over TCP. data SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier, followed sign (24 hexadecimal), followed by a one-byte channel identifier,
by the length of the encapsulated binary data as a binary, two-byte followed by the length of the encapsulated binary data as a binary,
integer in network byte order. The stream data follows immediately two-byte integer in network byte order. The stream data follows
afterwards, without a CRLF, but including the upper-layer protocol immediately afterwards, without a CRLF, but including the upper-layer
headers. Each $ block contains exactly one upper-layer protocol data protocol headers. Each $ block contains exactly one upper-layer
unit, e.g., one RTP packet. protocol data unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header with the The channel identifier is defined in the Transport header with the
interleaved parameter(Section 12.39). interleaved parameter(Section 12.39).
When the transport choice is RTP, RTCP messages are also interleaved When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. As a default, RTCP packets are 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 sent on the first available channel higher than the RTP channel. The
client MAY explicitly request RTCP packets on another channel. This is client MAY explicitly request RTCP packets on another channel. This
done by specifying two channels in the interleaved parameter of the is done by specifying two channels in the interleaved parameter of
Transport header(Section 12.39). the Transport header(Section 12.39).
RTCP is needed for synchronization when two or more streams are RTCP is needed for synchronization when two or more streams are
interleaved in such a fashion. Also, this provides a convenient way interleaved in such a fashion. Also, this provides a convenient way
to tunnel RTP/RTCP packets through the TCP control connection when to tunnel RTP/RTCP packets through the TCP control connection when
required by the network configuration and transfer them onto UDP required by the network configuration and transfer them onto UDP
when possible. when possible.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
CSeq: 2 CSeq: 2
Transport: RTP/AVP/TCP;interleaved=0-1 Transport: RTP/AVP/TCP;interleaved=0-1
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;interleaved=0-1 Transport: RTP/AVP/TCP;interleaved=0-1
Session: 12345 Session: 12345678
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0 C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
CSeq: 3 CSeq: 3
Session: 12345 Session: 12345678
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
Session: 12345 Session: 12345678
Date: 05 Jun 1997 18:59:15 GMT Date: 05 Jun 1997 18:59:15 GMT
RTP-Info: url=rtsp://foo.com/bar.file; RTP-Info: url=rtsp://foo.com/bar.file;
seq=232433;rtptime=972948234 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: $\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} S->C: $\001{2 byte length}{"length" bytes RTCP packet}
11 Status Code Definitions 11 Status Code Definitions
Where applicable, HTTP status [H10] codes are reused. Status codes Where applicable, HTTP status [H10] codes are reused. Status codes
that have the same meaning are not repeated here. See Table 1 for a 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. listing of which status codes may be returned by which requests.
11.1 Success 2xx 11.1 Success 2xx
11.1.1 250 Low on Storage Space 11.1.1 250 Low on Storage Space
The server returns this warning after receiving a RECORD request that The server returns this warning after receiving a RECORD request that
it may not be able to fulfill completely due to insufficient storage it may not be able to fulfill completely due to insufficient storage
space. If possible, the server should use the Range header to indicate space. If possible, the server should use the Range header to
what time period it may still be able to record. Since other processes indicate what time period it may still be able to record. Since other
on the server may be consuming storage space simultaneously, a client processes on the server may be consuming storage space
should take this only as an estimate. simultaneously, a client should take this only as an estimate.
11.2 Redirection 3xx 11.2 Redirection 3xx
See [H10.3]. See [H10.3].
Within RTSP, redirection may be used for load balancing or redirecting Within RTSP, redirection may be used for load balancing or
stream requests to a server topologically closer to the client. redirecting stream requests to a server topologically closer to the
Mechanisms to determine topological proximity are beyond the scope of client. Mechanisms to determine topological proximity are beyond the
this specification. scope of this specification.
11.3 Client Error 4xx 11.3 Client Error 4xx
11.3.1 405 Method Not Allowed 11.3.1 405 Method Not Allowed
The method specified in the request is not allowed for the resource The method specified in the request is not allowed for the resource
identified by the request URI. The response MUST include an Allow identified by the request URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource. 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 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 method not indicated during SETUP, e.g., if a RECORD request is
even though the mode parameter in the Transport header only specified issued even though the mode parameter in the Transport header only
PLAY. specified PLAY.
11.3.2 451 Parameter Not Understood 11.3.2 451 Parameter Not Understood
The recipient of the request does not support one or more parameters The recipient of the request does not support one or more parameters
contained in the request. contained in the request.
11.3.3 452 Conference Not Found 11.3.3 452 Conference Not Found
The conference indicated by a Conference header field is unknown to The conference indicated by a Conference header field is unknown to
the media server. the media server.
11.3.4 453 Not Enough Bandwidth 11.3.4 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth. This The request was refused because there was insufficient bandwidth.
may, for example, be the result of a resource reservation failure. This may, for example, be the result of a resource reservation
failure.
11.3.5 454 Session Not Found 11.3.5 454 Session Not Found
The RTSP session identifier in the Session header is missing, invalid, The RTSP session identifier in the Session header is missing,
or has timed out. invalid, or has timed out.
11.3.6 455 Method Not Valid in This State 11.3.6 455 Method Not Valid in This State
The client or server cannot process this request in its current state. The client or server cannot process this request in its current
The response SHOULD contain an Allow header to make error recovery state. The response SHOULD contain an Allow header to make error
easier. recovery easier.
11.3.7 456 Header Field Not Valid for Resource 11.3.7 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example, if The server could not act on a required request header. For example,
PLAY contains the Range header field but the stream does not allow if PLAY contains the Range header field but the stream does not allow
seeking. seeking.
11.3.8 457 Invalid Range 11.3.8 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the The Range value given is out of bounds, e.g., beyond the end of the
presentation. presentation.
11.3.9 458 Parameter Is Read-Only 11.3.9 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not modified. The parameter to be set by SET_PARAMETER can be read but not
modified.
11.3.10 459 Aggregate Operation Not Allowed 11.3.10 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URL in question since 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 it is an aggregate (presentation) URL. The method may be applied on a
stream URL. stream URL.
11.3.11 460 Only Aggregate Operation Allowed 11.3.11 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URL in question since 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 it is not an aggregate (presentation) URL. The method may be applied
on the presentation URL. on the presentation URL.
11.3.12 461 Unsupported Transport 11.3.12 461 Unsupported Transport
The Transport field did not contain a supported transport The Transport field did not contain a supported transport
specification. specification.
11.3.13 462 Destination Unreachable 11.3.13 462 Destination Unreachable
The data transmission channel could not be established because the The data transmission channel could not be established because the
client address could not be reached. This error will most likely be client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid Destination the result of a client attempt to place an invalid Destination
parameter in the Transport field. parameter in the Transport field.
11.3.14 551 Option not supported 11.3.14 551 Option not supported
An option given in the Require or the Proxy-Require fields was not An option given in the Require or the Proxy-Require fields was not
supported. The Unsupported header should be returned stating the supported. The Unsupported header should be returned stating the
option for which there is no support. option for which there is no support.
12 Header Field Definitions 12 Header Field Definitions
HTTP/1.1 [2] or other, non-standard header fields not listed here 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 currently have no well-defined meaning and SHOULD be ignored by the
recipient. recipient.
Table 3 summarizes the header fields used by RTSP. Type ``g'' Table 3 summarizes the header fields used by RTSP. Type "g"
designates general request headers to be found in both requests and designates general request headers to be found in both requests and
responses, type ``R'' designates request headers, type ``r'' responses, type "R" designates request headers, type "r" designates
designates response headers, and type ``e'' designates entity header response headers, and type "e" designates entity header fields.
fields. Fields marked with ``req.'' in the column labeled ``support'' Fields marked with "req." in the column labeled "support" MUST be
MUST be implemented by the recipient for a particular method, while implemented by the recipient for a particular method, while fields
fields marked ``opt.'' are optional. Note that not all fields marked marked "opt." are optional. Note that not all fields marked "req."
``req.'' will be sent in every request of this type. The ``req.'' will be sent in every request of this type. The "req." means only
means only that client (for response headers) and server (for request that client (for response headers) and server (for request headers)
headers) MUST implement the fields. The last column lists the method MUST implement the fields. The last column lists the method for which
for which this header field is meaningful; the designation ``entity'' this header field is meaningful; the designation "entity" refers to
refers to all methods that return a message body. Within this all methods that return a message body. Within this specification,
specification, DESCRIBE and GET_PARAMETER fall into this class. DESCRIBE and GET_PARAMETER fall into this class.
Header type support methods Header type support methods
Accept R opt. entity Accept R opt. entity
Accept-Encoding R opt. entity Accept-Encoding R opt. entity
Accept-Language R opt. all Accept-Language R opt. all
Allow r opt. all Allow r opt. all
Authorization R opt. all Authorization R opt. all
Bandwidth R opt. all Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control g opt. SETUP Cache-Control g opt. SETUP
skipping to change at page 48, line 8 skipping to change at page 46, line 8
Speed Rr opt. PLAY Speed Rr opt. PLAY
Transport Rr req. SETUP Transport Rr req. SETUP
Unsupported r req. all Unsupported r req. all
User-Agent R opt. all User-Agent R opt. all
Via g opt. all Via g opt. all
WWW-Authenticate r opt. all WWW-Authenticate r opt. all
Overview of RTSP header fields Overview of RTSP header fields
12.1 Accept 12.1 Accept
The Accept request-header field can be used to specify certain The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the presentation description content types which are acceptable for the
response. response.
The ``level'' parameter for presentation descriptions is properly The "level" parameter for presentation descriptions is properly
defined as part of the MIME type registration, not here. defined as part of the MIME type registration, not here.
See [H14.1] for syntax. See [H14.1] for syntax.
Example of use: Example of use:
Accept: application/rtsl, application/sdp;level=2 Accept: application/rtsl, application/sdp;level=2
12.2 Accept-Encoding 12.2 Accept-Encoding
See [H14.3] See [H14.3]
12.3 Accept-Language 12.3 Accept-Language
See [H14.4]. Note that the language specified applies to the See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media presentation description and any reason phrases, not the media
content. content.
12.4 Allow 12.4 Allow
The Allow response header field lists the methods supported by the The Allow response header field lists the methods supported by the
resource identified by the request-URI. The purpose of this field is resource identified by the request-URI. The purpose of this field is
to strictly inform the recipient of valid methods associated with the to strictly inform the recipient of valid methods associated with the
resource. An Allow header field must be present in a 405 (Method not resource. An Allow header field must be present in a 405 (Method not
allowed) response. allowed) response.
Example of use: Example of use:
Allow: SETUP, PLAY, RECORD, SET_PARAMETER Allow: SETUP, PLAY, RECORD, SET_PARAMETER
12.5 Authorization 12.5 Authorization
See [H14.8] See [H14.8]
12.6 Bandwidth 12.6 Bandwidth
The Bandwidth request header field describes the estimated bandwidth The Bandwidth request header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured available to the client, expressed as a positive integer and measured
in bits per second. The bandwidth available to the client may change in bits per second. The bandwidth available to the client may change
during an RTSP session, e.g., due to modem retraining. during an RTSP session, e.g., due to modem retraining.
Bandwidth = "Bandwidth" ":" 1*DIGIT Bandwidth = "Bandwidth" ":" 1*DIGIT
Example: Example:
Bandwidth: 4000 Bandwidth: 4000
12.7 Blocksize 12.7 Blocksize
This request header field is sent from the client to the media This request header field is sent from the client to the media server
server asking the server for a particular media packet size. This asking the server for a particular media packet size. This packet
packet size does not include lower-layer headers such as IP, UDP, or size does not include lower-layer headers such as IP, UDP, or RTP.
RTP. The server is free to use a blocksize which is lower than the one 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 requested. The server MAY truncate this packet size to the closest
multiple of the minimum, media-specific block size, or override it multiple of the minimum, media-specific block size, or override it
with the media-specific size if necessary. The block size MUST be a with the media-specific size if necessary. The block size MUST be a
positive decimal number, measured in octets. The server only returns positive decimal number, measured in octets. The server only returns
an error (416) if the value is syntactically invalid. an error (416) if the value is syntactically invalid.
12.8 Cache-Control 12.8 Cache-Control
The Cache-Control general header field is used to specify directives The Cache-Control general header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the that MUST be obeyed by all caching mechanisms along the
request/response chain. request/response chain.
Cache directives must be passed through by a proxy or gateway Cache directives must be passed through by a proxy or gateway
application, regardless of their significance to that application, application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a cache- request/response chain. It is not possible to specify a cache-
directive for a specific cache. directive for a specific cache.
Cache-Control should only be specified in a SETUP request and its Cache-Control should only be specified in a SETUP request and its
response. Note: Cache-Control does not govern the caching of responses response. Note: Cache-Control does not govern the caching of
as for HTTP, but rather of the stream identified by the SETUP request. responses as for HTTP, but rather of the stream identified by the
Responses to RTSP requests are not cacheable, except for responses to SETUP request. Responses to RTSP requests are not cacheable, except
DESCRIBE. for responses to DESCRIBE.
Cache-Control = "Cache-Control" ":" 1#cache-directive Cache-Control = "Cache-Control" ":" 1#cache-directive
cache-directive = cache-request-directive cache-directive = cache-request-directive
| cache-response-directive | cache-response-directive
cache-request-directive = "no-cache" cache-request-directive = "no-cache"
| "max-stale" | "max-stale"
| "min-fresh" | "min-fresh"
| "only-if-cached" | "only-if-cached"
| cache-extension | cache-extension
cache-response-directive = "public" cache-response-directive = "public"
skipping to change at page 50, line 34 skipping to change at page 48, line 20
Indicates that the media stream MUST NOT be cached anywhere. Indicates that the media stream MUST NOT be cached anywhere.
This allows an origin server to prevent caching even by caches This allows an origin server to prevent caching even by caches
that have been configured to return stale responses to client that have been configured to return stale responses to client
requests. requests.
public: public:
Indicates that the media stream is cacheable by any cache. Indicates that the media stream is cacheable by any cache.
private: private:
Indicates that the media stream is intended for a single user Indicates that the media stream is intended for a single user
and MUST NOT be cached by a shared cache. A private and MUST NOT be cached by a shared cache. A private (non-
(non-shared) cache may cache the media stream. shared) cache may cache the media stream.
no-transform: no-transform:
An intermediate cache (proxy) may find it useful to convert the An intermediate cache (proxy) may find it useful to convert
media type of a certain stream. A proxy might, for example, the media type of a certain stream. A proxy might, for
convert between video formats to save cache space or to reduce example, convert between video formats to save cache space or
the amount of traffic on a slow link. Serious operational to reduce the amount of traffic on a slow link. Serious
problems may occur, however, when these transformations have operational problems may occur, however, when these
been applied to streams intended for certain kinds of transformations have been applied to streams intended for
applications. For example, applications for medical imaging, certain kinds of applications. For example, applications for
scientific data analysis and those using end-to-end medical imaging, scientific data analysis and those using
authentication all depend on receiving a stream that is end-to-end authentication all depend on receiving a stream
bit-for-bit identical to the original entity-body. Therefore, that is bit-for-bit identical to the original entity-body.
if a response includes the no-transform directive, an Therefore, if a response includes the no-transform directive,
intermediate cache or proxy MUST NOT change the encoding of the an intermediate cache or proxy MUST NOT change the encoding of
stream. Unlike HTTP, RTSP does not provide for partial the stream. Unlike HTTP, RTSP does not provide for partial
transformation at this point, e.g., allowing translation into a transformation at this point, e.g., allowing translation into
different language. a different language.
only-if-cached: only-if-cached:
In some cases, such as times of extremely poor network In some cases, such as times of extremely poor network
connectivity, a client may want a cache to return only those connectivity, a client may want a cache to return only those
media streams that it currently has stored, and not to receive media streams that it currently has stored, and not to receive
these from the origin server. To do this, the client may these from the origin server. To do this, the client may
include the only-if-cached directive in a request. If it include the only-if-cached directive in a request. If it
receives this directive, a cache SHOULD either respond using a receives this directive, a cache SHOULD either respond using a
cached media stream that is consistent with the other cached media stream that is consistent with the other
constraints of the request, or respond with a 504 (Gateway constraints of the request, or respond with a 504 (Gateway
Timeout) status. However, if a group of caches is being Timeout) status. However, if a group of caches is being
operated as a unified system with good internal connectivity, operated as a unified system with good internal connectivity,
such a request MAY be forwarded within that group of caches. such a request MAY be forwarded within that group of caches.
max-stale: max-stale:
Indicates that the client is willing to accept a media stream Indicates that the client is willing to accept a media stream
that has exceeded its expiration time. If max-stale is assigned that has exceeded its expiration time. If max-stale is
a value, then the client is willing to accept a response that assigned a value, then the client is willing to accept a
has exceeded its expiration time by no more than the specified response that has exceeded its expiration time by no more than
number of seconds. If no value is assigned to max-stale, then the specified number of seconds. If no value is assigned to
the client is willing to accept a stale response of any age. max-stale, then the client is willing to accept a stale
response of any age.
min-fresh: min-fresh:
Indicates that the client is willing to accept a media stream Indicates that the client is willing to accept a media stream
whose freshness lifetime is no less than its current age plus whose freshness lifetime is no less than its current age plus
the specified time in seconds. That is, the client wants a the specified time in seconds. That is, the client wants a
response that will still be fresh for at least the specified response that will still be fresh for at least the specified
number of seconds. number of seconds.
must-revalidate: must-revalidate:
When the must-revalidate directive is present in a SETUP When the must-revalidate directive is present in a SETUP
response received by a cache, that cache MUST NOT use the entry response received by a cache, that cache MUST NOT use the
after it becomes stale to respond to a subsequent request entry after it becomes stale to respond to a subsequent
without first revalidating it with the origin server. That is, request without first revalidating it with the origin server.
the cache must do an end-to-end revalidation every time, if, That is, the cache must do an end-to-end revalidation every
based solely on the origin server's Expires, the cached time, if, based solely on the origin server's Expires, the
response is stale.) cached response is stale.)
12.9 Conference 12.9 Conference
This request header field establishes a logical connection between a This request header field establishes a logical connection between a
pre-established conference and an RTSP stream. The conference-id must pre-established conference and an RTSP stream. The conference-id must
not be changed for the same RTSP session. not be changed for the same RTSP session.
Conference = "Conference" ":" conference-id Conference = "Conference" ":" conference-id Example:
Example:
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
A response code of 452 (452 Conference Not Found) is returned if the A response code of 452 (452 Conference Not Found) is returned if the
conference-id is not valid. conference-id is not valid.
12.10 Connection 12.10 Connection
See [H14.10] See [H14.10]
12.11 Content-Base 12.11 Content-Base
See [H14.11] See [H14.11]
12.12 Content-Encoding 12.12 Content-Encoding
See [H14.12] See [H14.12]
12.13 Content-Language 12.13 Content-Language
See [H14.13] See [H14.13]
12.14 Content-Length 12.14 Content-Length
This field contains the length of the content of the method (i.e. 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 after the double CRLF following the last header). Unlike HTTP, it
be included in all messages that carry content beyond the header 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 portion of the message. If it is missing, a default value of zero is
assumed. It is interpreted according to [H14.14]. assumed. It is interpreted according to [H14.14].
12.15 Content-Location 12.15 Content-Location
See [H14.15] See [H14.15]
12.16 Content-Type 12.16 Content-Type
See [H14.18]. Note that the content types suitable for RTSP are See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and likely to be restricted in practice to presentation descriptions and
parameter-value types. parameter-value types.
12.17 CSeq 12.17 CSeq
The CSeq field specifies the sequence number for an RTSP The CSeq field specifies the sequence number for an RTSP request-
request-response pair. This field MUST be present in all requests and response pair. This field MUST be present in all requests and
responses. For every RTSP request containing the given sequence responses. For every RTSP request containing the given sequence
number, there will be a corresponding response having the same number. number, there will be a corresponding response having the same
Any retransmitted request must contain the same sequence number as the number. Any retransmitted request must contain the same sequence
original (i.e. the sequence number is not incremented for number as the original (i.e. the sequence number is not incremented
retransmissions of the same request). for retransmissions of the same request).
12.18 Date 12.18 Date
See [H14.19]. See [H14.19].
12.19 Expires 12.19 Expires
The Expires entity-header field gives a date and time after which The Expires entity-header field gives a date and time after which the
the description or media-stream should be considered stale. The description or media-stream should be considered stale. The
interpretation depends on the method: interpretation depends on the method:
DESCRIBE response: DESCRIBE response:
The Expires header indicates a date and time after which the The Expires header indicates a date and time after which the
description should be considered stale. description should be considered stale.
A stale cache entry may not normally be returned by a cache (either a 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 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 the origin server (or with an intermediate cache that has a fresh
of the entity). See section 13 for further discussion of the copy of the entity). See section 13 for further discussion of the
expiration model. expiration model.
The presence of an Expires field does not imply that the original 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. 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 The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format: [H3.3]; it MUST be in RFC1123-date format:
Expires = "Expires" ":" HTTP-date Expires = "Expires" ":" HTTP-date
An example of its use is An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats, RTSP/1.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occured in the past especially including the value "0", as having occurred in the past
(i.e., ``already expired''). (i.e., "already expired").
To mark a response as ``already expired,'' an origin server should use 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 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 response as "never expires," an origin server should use an Expires
date approximately one year from the time the response is sent. 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 RTSP/1.0 servers should not send Expires dates more than one year in
the future. the future.
The presence of an Expires header field with a date value of some time The presence of an Expires header field with a date value of some
in the future on a media stream that otherwise would by default be time in the future on a media stream that otherwise would by default
non-cacheable indicates that the media stream is cacheable, unless be non-cacheable indicates that the media stream is cacheable, unless
indicated otherwise by a Cache-Control header field (Section 12.8). indicated otherwise by a Cache-Control header field (Section 12.8).
12.20 From 12.20 From
See [H14.22]. See [H14.22].
12.21 Host 12.21 Host
This HTTP request header field is not needed for RTSP. It should be This HTTP request header field is not needed for RTSP. It should be
silently ignored if sent. silently ignored if sent.
12.22 If-Match 12.22 If-Match
See [H14.25]. See [H14.25].
This field is especially useful for ensuring the integrity of the This field is especially useful for ensuring the integrity of the
presentation description, in both the case where it is fetched via 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 means external to RTSP (such as HTTP), or in the case where the
implementation is guaranteeing the integrity of the description server implementation is guaranteeing the integrity of the
between the time of the DESCRIBE message and the SETUP message. description between the time of the DESCRIBE message and the SETUP
message.
The identifier is an opaque identifier, and thus is not specific to The identifier is an opaque identifier, and thus is not specific to
any particular session description language. any particular session description language.
12.23 If-Modified-Since 12.23 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional. If the requested variant and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (not modified) stream will not be set up (SETUP). Instead, a 304 (not modified)
response will be returned without any message-body. response will be returned without any message-body.
If-Modified-Since = "If-Modified-Since" ":" HTTP-date If-Modified-Since = "If-Modified-Since" ":" HTTP-date
An example of the field is: An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
12.24 Last-Modified 12.24 Last-Modified
The Last-Modified entity-header field indicates the date and time at The Last-Modified entity-header field indicates the date and time at
which the origin server believes the presentation description or media which the origin server believes the presentation description or
stream was last modified. See [H14.29]. For the methods DESCRIBE or media stream was last modified. See [H14.29]. For the methods
ANNOUNCE, the header field indicates the last modification date and DESCRIBE or ANNOUNCE, the header field indicates the last
time of the description, for SETUP that of the media stream. modification date and time of the description, for SETUP that of the
media stream.
12.25 Location 12.25 Location
See [H14.30]. See [H14.30].
12.26 Proxy-Authenticate 12.26 Proxy-Authenticate
See [H14.33]. See [H14.33].
12.27 Proxy-Require 12.27 Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive The Proxy-Require header is used to indicate proxy-sensitive features
features that MUST be supported by the proxy. Any Proxy-Require header that MUST be supported by the proxy. Any Proxy-Require header
features that are not supported by the proxy MUST be negatively features that are not supported by the proxy MUST be negatively
acknowledged by the proxy to the client if not supported. Servers acknowledged by the proxy to the client if not supported. Servers
should treat this field identically to the Require field. should treat this field identically to the Require field.
See Section 12.32 for more details on the mechanics of this message See Section 12.32 for more details on the mechanics of this message
and a usage example. and a usage example.
12.28 Public 12.28 Public
See [H14.35]. See [H14.35].
12.29 Range 12.29 Range
This request and response header field specifies a range of time. This request and response header field specifies a range of time.
The range can be specified in a number of units. This specification The range can be specified in a number of units. This specification
defines the smpte (Section 3.5), npt (Section 3.6), and clock 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 (Section 3.7) range units. Within RTSP, byte ranges [H14.36.1] are
meaningful and MUST NOT be used. The header may also contain a time not meaningful and MUST NOT be used. The header may also contain a
parameter in UTC, specifying the time at which the operation is to be time parameter in UTC, specifying the time at which the operation is
made effective. Servers supporting the Range header MUST understand to be made effective. Servers supporting the Range header MUST
the NPT range format and SHOULD understand the SMPTE range format. The understand the NPT range format and SHOULD understand the SMPTE range
Range response header indicates what range of time is actually being format. The Range response header indicates what range of time is
played or recorded. If the Range header is given in a time format that actually being played or recorded. If the Range header is given in a
is not understood, the recipient should return ``501 Not time format that is not understood, the recipient should return "501
Implemented''. Not Implemented".
Range = "Range" ":" 1#ranges-specifier Ranges are half-open intervals, including the lower point, but
[ ";" "time" "=" utc-time ] 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 ranges-specifier = npt-range | utc-range | smpte-range
Example: Example:
Range: clock=19960213T143205Z-;time=19970123T143720Z Range: clock=19960213T143205Z-;time=19970123T143720Z
The notation is similar to that used for the HTTP/1.1 [2] The notation is similar to that used for the HTTP/1.1 [2] byte-
byte-range header. It allows clients to select an excerpt from the range header. It allows clients to select an excerpt from the media
media object, and to play from a given point to the end as well as object, and to play from a given point to the end as well as from
from the current location to a given point. The start of playback the current location to a given point. The start of playback can be
can be scheduled for any time in the future, although a server may scheduled for any time in the future, although a server may refuse
refuse to keep server resources for extended idle periods. to keep server resources for extended idle periods.
12.30 Referer 12.30 Referer
See [H14.37]. The URL refers to that of the presentation See [H14.37]. The URL refers to that of the presentation description,
description, typically retrieved via HTTP. typically retrieved via HTTP.
12.31 Retry-After 12.31 Retry-After
See [H14.38]. See [H14.38].
12.32 Require 12.32 Require
The Require header is used by clients to query the server about 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 options that it may or may not support. The server MUST respond to
this header by using the Unsupported header to negatively acknowledge this header by using the Unsupported header to negatively acknowledge
those options which are NOT supported. those options which are NOT supported.
This is to make sure that the client-server interaction will This is to make sure that the client-server interaction will
proceed without delay when all options are understood by both proceed without delay when all options are understood by both
sides, and only slow down if options are not understood (as in the sides, and only slow down if options are not understood (as in the
case above). For a well-matched client-server pair, the interaction case above). For a well-matched client-server pair, the interaction
proceeds quickly, saving a round-trip often required by negotiation proceeds quickly, saving a round-trip often required by negotiation
mechanisms. In addition, it also removes state ambiguity when the mechanisms. In addition, it also removes state ambiguity when the
skipping to change at page 57, line 42 skipping to change at page 54, line 47
S->C: RTSP/1.0 551 Option not supported S->C: RTSP/1.0 551 Option not supported
CSeq: 302 CSeq: 302
Unsupported: funky-feature Unsupported: funky-feature
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0 C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 303 CSeq: 303
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 303 CSeq: 303
In this example, ``funky-feature'' is the feature tag which indicates In this example, "funky-feature" is the feature tag which indicates
to the client that the fictional Funky-Parameter field is required. to the client that the fictional Funky-Parameter field is required.
The relationship between ``funky-feature'' and Funky-Parameter is not The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an communicated via the RTSP exchange, since that relationship is an
immutable property of ``funky-feature'' and thus should not be immutable property of "funky-feature" and thus should not be
transmitted with every exchange. transmitted with every exchange.
Proxies and other intermediary devices SHOULD ignore features that are Proxies and other intermediary devices SHOULD ignore features that
not understood in this field. If a particular extension requires that are not understood in this field. If a particular extension requires
intermediate devices support it, the extension should be tagged in the that intermediate devices support it, the extension should be tagged
Proxy-Require field instead (see Section 3.4). in the Proxy-Require field instead (see Section 12.27).
12.33 RTP-Info 12.33 RTP-Info
This field is used to set RTP-specific parameters in the PLAY This field is used to set RTP-specific parameters in the PLAY
response. response.
url: url:
Indicates the stream URL which for which the following RTP Indicates the stream URL which for which the following RTP
parameters correspond. parameters correspond.
seq: seq:
Indicates the sequence number of the first packet of the Indicates the sequence number of the first packet of the
stream. This allows clients to gracefully deal with packets stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that packets that originated before the seek from packets that
originated after the seek. originated after the seek.
rtptime: rtptime:
Indicates the RTP timestamp of the first packet of the stream. Indicates the RTP timestamp corresponding to the time value in
The client uses this value to calculate the mapping of RTP time the Range response header. (Note: For aggregate control, a
to NPT. 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 A mapping from RTP timestamps to NTP timestamps (wall clock) is
available via RTCP. However, this information is not sufficient to available via RTCP. However, this information is not sufficient to
generate a mapping from RTP timestamps to NPT. Furthermore, in generate a mapping from RTP timestamps to NPT. Furthermore, in
order to ensure that this information is available at the necessary order to ensure that this information is available at the necessary
time (immediately at startup or after a seek), and that it is time (immediately at startup or after a seek), and that it is
delivered reliably, this mapping is placed in the RTSP control delivered reliably, this mapping is placed in the RTSP control
channel. channel.
In order to compensate for drift for long, uninterrupted In order to compensate for drift for long, uninterrupted
skipping to change at page 59, line 12 skipping to change at page 56, line 19
parameter = ";" "seq" "=" 1*DIGIT parameter = ";" "seq" "=" 1*DIGIT
| ";" "rtptime" "=" 1*DIGIT | ";" "rtptime" "=" 1*DIGIT
Example: Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102, RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211 url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
12.34 Scale 12.34 Scale
A scale value of 1 indicates normal play or record at the normal 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 forward viewing rate. If not 1, the value corresponds to the rate
respect to normal viewing rate. For example, a ratio of 2 indicates with respect to normal viewing rate. For example, a ratio of 2
twice the normal viewing rate (``fast forward'') and a ratio of 0.5 indicates twice the normal viewing rate ("fast forward") and a ratio
indicates half the normal viewing rate. In other words, a ratio of 2 of 0.5 indicates half the normal viewing rate. In other words, a
has normal play time increase at twice the wallclock rate. For every ratio of 2 has normal play time increase at twice the wallclock rate.
second of elapsed (wallclock) time, 2 seconds of content will be For every second of elapsed (wallclock) time, 2 seconds of content
delivered. A negative value indicates reverse direction. will be delivered. A negative value indicates reverse direction.
Unless requested otherwise by the Speed parameter, the data rate Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on the SHOULD not be changed. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver 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 only key frames or selected key frames. For audio, it may time-scale
the audio while preserving pitch or, less desirably, deliver fragments the audio while preserving pitch or, less desirably, deliver
of audio. fragments of audio.
The server should try to approximate the viewing rate, but may The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response MUST restrict the range of scale values that it supports. The response
contain the actual scale value chosen by the server. MUST contain the actual scale value chosen by the server.
If the request contains a Range parameter, the new scale value will If the request contains a Range parameter, the new scale value will
take effect at that time. take effect at that time.
Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ] Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
Example of playing in reverse at 3.5 times normal rate: Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5 Scale: -3.5
12.35 Speed 12.35 Speed
This request header fields parameter requests the server to deliver This request header fields parameter requests the server to deliver
data to the client at a particular speed, contingent on the server's 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. ability and desire to serve the media stream at the given speed.
Implementation by the server is OPTIONAL. The default is the bit rate Implementation by the server is OPTIONAL. The default is the bit rate
of the stream. of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of 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 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, speed of zero is invalid. If the request contains a Range parameter,
the new speed value will take effect at that time. the new speed value will take effect at that time.
Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ] Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]
Example: Example:
Speed: 2.5 Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It is Use of this field changes the bandwidth used for data delivery. It is
meant for use in specific circumstances where preview of the meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. Implementors presentation at a higher or lower rate is necessary. Implementors
should keep in mind that bandwidth for the session may be negotiated should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. When data is delivered over UDP, it is highly may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss recommended that means such as RTCP be used to track packet loss
rates. rates.
12.36 Server 12.36 Server
See [H14.39] See [H14.39]
12.37 Session 12.37 Session
This request and response header field identifies an RTSP session This request and response header field identifies an RTSP session
started by the media server in a SETUP response and concluded by started by the media server in a SETUP response and concluded by
TEARDOWN on the presentation URL. The session identifier is chosen 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 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. identifier, it MUST return it for any request related to that
A server does not have to set up a session identifier if it has other session. A server does not have to set up a session identifier if it
means of identifying a session, such as dynamically generated URLs. has other means of identifying a session, such as dynamically
generated URLs.
Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ] Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
The timeout parameter is only allowed in a response header. The server The timeout parameter is only allowed in a response header. The
uses it to indicate to the client how long the server is prepared to server uses it to indicate to the client how long the server is
wait between RTSP commands before closing the session due to lack of prepared to wait between RTSP commands before closing the session due
activity (see Section A). The timeout is measured in seconds, with a to lack of activity (see Section A). The timeout is measured in
default of 60 seconds (1 minute). seconds, with a default of 60 seconds (1 minute).
Note that a session identifier identifies a RTSP session across Note that a session identifier identifies a RTSP session across
transport sessions or connections. Control messages for more than one transport sessions or connections. Control messages for more than one
RTSP URL may be sent within a single RTSP session. Hence, it is RTSP URL may be sent within a single RTSP session. Hence, it is
possible that clients use the same session for controlling many possible that clients use the same session for controlling many
streams constituting a presentation, as long as all the streams come streams constituting a presentation, as long as all the streams come
from the same server. (See example in Section 14). However, multiple from the same server. (See example in Section 14). However, multiple
``user'' sessions for the same URL from the same client MUST use "user" sessions for the same URL from the same client MUST use
different session identifiers. different session identifiers.
The session identifier is needed to distinguish several delivery The session identifier is needed to distinguish several delivery
requests for the same URL coming from the same client. requests for the same URL coming from the same client.
The response 454 (Session Not Found) is returned if the session The response 454 (Session Not Found) is returned if the session
identifier is invalid. identifier is invalid.
12.38 Timestamp 12.38 Timestamp
The timestamp general header describes when the client sent the The timestamp general header describes when the client sent the
request to the server. The value of the timestamp is of significance 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 only to the client and may use any timescale. The server MUST echo
exact same value and MAY, if it has accurate information about this, the exact same value and MAY, if it has accurate information about
add a floating point number indicating the number of seconds that has this, add a floating point number indicating the number of seconds
elapsed since it has received the request. The timestamp is used by that has elapsed since it has received the request. The timestamp is
the client to compute the round-trip time to the server so that it can used by the client to compute the round-trip time to the server so
adjust the timeout value for retransmissions. that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ] Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ] delay = *(DIGIT) [ "." *(DIGIT) ]
12.39 Transport 12.39 Transport
This request header indicates which transport protocol is to be used This request header indicates which transport protocol is to be used
and configures its parameters such as destination address, and configures its parameters such as destination address,
compression, multicast time-to-live and destination port for a single compression, multicast time-to-live and destination port for a single
stream. It sets those values not already determined by a presentation stream. It sets those values not already determined by a presentation
description. description.
Transports are comma separated, listed in order of preference. Transports are comma separated, listed in order of preference.
Parameters may be added to each transport, separated by a semicolon. Parameters may be added to each transport, separated by a semicolon.
The Transport header MAY also be used to change certain transport The Transport header MAY also be used to change certain transport
parameters. A server MAY refuse to change parameters of an existing parameters. A server MAY refuse to change parameters of an existing
stream. stream.
The server MAY return a Transport response header in the response to The server MAY return a Transport response header in the response to
indicate the values actually chosen. indicate the values actually chosen.
A Transport request header field may contain a list of transport A Transport request header field may contain a list of transport
options acceptable to the client. In that case, the server MUST return options acceptable to the client. In that case, the server MUST
a single option which was actually chosen. return a single option which was actually chosen.
The syntax for the transport specifier is The syntax for the transport specifier is
transport/profile/lower-transport. transport/profile/lower-transport.
The default value for the ``lower-transport'' parameters is specific The default value for the "lower-transport" parameters is specific to
to the profile. For RTP/AVP, the default is UDP. the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport: Below are the configuration parameters associated with transport:
General parameters: General parameters:
unicast | multicast unicast | multicast:
: mutually exclusive indication of whether unicast or multicast mutually exclusive indication of whether unicast or multicast
delivery will be attempted. Default value is multicast. Clients delivery will be attempted. Default value is multicast.
that are capable of handling both unicast and multicast Clients that are capable of handling both unicast and
transmission MUST indicate such capability by including two multicast transmission MUST indicate such capability by
full transport-specs with separate parameters for each. including two full transport-specs with separate parameters
for each.
destination: destination:
The address to which a stream will be sent. The client may The address to which a stream will be sent. The client may
specify the multicast address with the destination parameter. specify the multicast address with the destination parameter.
To avoid becoming the unwitting perpetrator of a To avoid becoming the unwitting perpetrator of a remote-
remote-controlled denial-of-service attack, a server SHOULD controlled denial-of-service attack, a server SHOULD
authenticate the client and SHOULD log such attempts before authenticate the client and SHOULD log such attempts before
allowing the client to direct a media stream to an address not allowing the client to direct a media stream to an address not
chosen by the server. This is particularly important if RTSP chosen by the server. This is particularly important if RTSP
commands are issued via UDP, but implementations cannot rely on commands are issued via UDP, but implementations cannot rely
TCP as reliable means of client identification by itself. A on TCP as reliable means of client identification by itself. A
server SHOULD not allow a client to direct media streams to an server SHOULD not allow a client to direct media streams to an
address that differs from the address commands are coming from. address that differs from the address commands are coming
from.
source: source:
If the source address for the stream is different than can be If the source address for the stream is different than can be
derived from the RTSP endpoint address (the server in playback derived from the RTSP endpoint address (the server in playback
or the client in recording), the source MAY be specified. or the client in recording), the source MAY be specified.
This information may also be available through SDP. However, since This information may also be available through SDP. However, since
this is more a feature of transport than media initialization, the this is more a feature of transport than media initialization, the
authoritative source for this information should be in the SETUP authoritative source for this information should be in the SETUP
response. response.
skipping to change at page 63, line 21 skipping to change at page 60, line 21
The mode parameter indicates the methods to be supported for The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY. provided, the default is PLAY.
append: append:
If the mode parameter includes RECORD, the append parameter If the mode parameter includes RECORD, the append parameter
indicates that the media data should append to the existing indicates that the media data should append to the existing
resource rather than overwrite it. If appending is requested resource rather than overwrite it. If appending is requested
and the server does not support this, it MUST refuse the and the server does not support this, it MUST refuse the
request rather than overwrite the resource identified by the request rather than overwrite the resource identified by the
URI. The append parameter is ignored if the mode parameter does URI. The append parameter is ignored if the mode parameter
not contain RECORD. does not contain RECORD.
interleaved: interleaved:
The interleaved parameter implies mixing the media stream with The interleaved parameter implies mixing the media stream with
the control stream in whatever protocol is being used by the the control stream in whatever protocol is being used by the
control stream, using the mechanism defined in Section 10.12. control stream, using the mechanism defined in Section 10.12.
The argument provides the channel number to be used in the $ The argument provides the channel number to be used in the $
statement. This parameter may be specified as a range, e.g., statement. This parameter may be specified as a range, e.g.,
interleaved=4-5 in cases where the transport choice for the interleaved=4-5 in cases where the transport choice for the
media stream requires it. media stream requires it.
skipping to change at page 63, line 48 skipping to change at page 60, line 48
ttl: ttl:
multicast time-to-live multicast time-to-live
RTP Specific: RTP Specific:
port: port:
This parameter provides the RTP/RTCP port pair for a multicast This parameter provides the RTP/RTCP port pair for a multicast
session. It is specified as a range, e.g., port=3456-3457. session. It is specified as a range, e.g., port=3456-3457.
client_port: client_port:
This parameter provides the unicast RTP/RTCP port pair on the This parameter provides the unicast RTP/RTCP port pair on
client where media data and control information is to be sent. which the client has chosen to receive media data and control
It is specified as a range, e.g., port=3456-3457. information. It is specified as a range, e.g.,
client_port=3456-3457.
server_port: server_port:
This parameter provides the unicast RTP/RTCP port pair on the This parameter provides the unicast RTP/RTCP port pair on
server where media data and control information is to be sent. which the server has chosen to receive media data and control
It is specified as a range, e.g., port=3456-3457. information. It is specified as a range, e.g.,
server_port=3456-3457.
ssrc: ssrc:
The ssrc parameter indicates the RTP SSRC [24, Sec. 3] value The ssrc parameter indicates the RTP SSRC [24, Sec. 3] value
that should be (request) or will be (response) used by the that should be (request) or will be (response) used by the
media server. This parameter is only valid for unicast media server. This parameter is only valid for unicast
transmission. It identifies the synchronization source to be transmission. It identifies the synchronization source to be
associated with the media stream. associated with the media stream.
Transport = "Transport" ":" Transport = "Transport" ":"
1\#transport-spec 1\#transport-spec
skipping to change at page 65, line 7 skipping to change at page 62, line 7
RTP/AVP;unicast;client_port=3456-3457;mode="PLAY" RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"
The Transport header is restricted to describing a single RTP The Transport header is restricted to describing a single RTP
stream. (RTSP can also control multiple streams as a single stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of of session description formats greatly simplifies designs of
firewalls. firewalls.
12.40 Unsupported 12.40 Unsupported
The Unsupported response header lists the features not supported by The Unsupported response header lists the features not supported by
the server. In the case where the feature was specified via the 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 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 between the client and the server, the proxy MUST insert a message
reply with an error message ``551 Option Not Supported''. reply with an error message "551 Option Not Supported".
See Section 12.32 for a usage example. See Section 12.32 for a usage example.
12.41 User-Agent 12.41 User-Agent
See [H14.42] See [H14.42]
12.42 Vary 12.42 Vary
See [H14.43] See [H14.43]
12.43 Via 12.43 Via
See [H14.44]. See [H14.44].
12.44 WWW-Authenticate 12.44 WWW-Authentica
See [H14.46]. See [H14.46].
13 Caching 13 Caching
In HTTP, response-request pairs are cached. RTSP differs In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE or exception of the presentation description returned by DESCRIBE or
included with ANNOUNCE. (Since the responses for anything but DESCRIBE included with ANNOUNCE. (Since the responses for anything but
and GET_PARAMETER do not return any data, caching is not really an DESCRIBE and GET_PARAMETER do not return any data, caching is not
issue for these requests.) However, it is desirable for the continuous really an issue for these requests.) However, it is desirable for the
media data, typically delivered out-of-band with respect to RTSP, to continuous media data, typically delivered out-of-band with respect
be cached, as well as the session description. to RTSP, to be cached, as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it 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 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 description. It can determine whether the copy is up-to-date by
issuing a SETUP or DESCRIBE request, respectively, and comparing the issuing a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is not 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 up-to-date, it modifies the SETUP transport parameters as appropriate
and forwards the request to the origin server. Subsequent control and forwards the request to the origin server. Subsequent control
commands such as PLAY or PAUSE then pass the proxy unmodified. The commands such as PLAY or PAUSE then pass the proxy unmodified. The
proxy delivers the continuous media data to the client, while possibly proxy delivers the continuous media data to the client, while
making a local copy for later reuse. The exact behavior allowed to the possibly making a local copy for later reuse. The exact behavior
cache is given by the cache-response directives described in allowed to the cache is given by the cache-response directives
Section 12.8. A cache MUST answer any DESCRIBE requests if it is described in Section 12.8. A cache MUST answer any DESCRIBE requests
currently serving the stream to the requestor, as it is possible that if it is currently serving the stream to the requestor, as it is
low-level details of the stream description may have changed on the possible that low-level details of the stream description may have
origin-server. changed on the origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
``cut-through'' variety. Rather than retrieving the whole resource through" variety. Rather than retrieving the whole resource from the
from the origin server, the cache simply copies the streaming data as origin server, the cache simply copies the streaming data as it
it passes by on its way to the client. Thus, it does not introduce passes by on its way to the client. Thus, it does not introduce
additional latency. additional latency.
To the client, an RTSP proxy cache appears like a regular media 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 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 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 the objects it caches, a media cache has to store the presentation
description. Typically, a cache eliminates all transport-references description. Typically, a cache eliminates all transport-references
(that is, multicast information) from the presentation description, (that is, multicast information) from the presentation description,
since these are independent of the data delivery from the cache to the since these are independent of the data delivery from the cache to
client. Information on the encodings remains the same. If the cache is the client. Information on the encodings remains the same. If the
able to translate the cached media data, it would create a new cache is able to translate the cached media data, it would create a
presentation description with all the encoding possibilities it can new presentation description with all the encoding possibilities it
offer. can offer.
14 Examples 14 Examples
The following examples refer to stream description formats that are The following examples refer to stream description formats that are
not standards, such as RTSL. The following examples are not to be used not standards, such as RTSL. The following examples are not to be
as a reference for those formats. used as a reference for those formats.
14.1 Media on Demand (Unicast) 14.1 Media on Demand (Unicast)
Client C requests a movie from media servers A ( audio.example.com) 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 and V (video.example.com). The media description is stored on a web
server W . The media description contains descriptions of the server W . The media description contains descriptions of the
presentation and all its streams, including the codecs that are presentation and all its streams, including the codecs that are
available, dynamic RTP payload types, the protocol stack, and content available, dynamic RTP payload types, the protocol stack, and content
information such as language or copyright restrictions. It may also information such as language or copyright restrictions. It may also
give an indication about the timeline of the movie. give an indication about the timeline of the movie.
In this example, the client is only interested in the last part of the In this example, the client is only interested in the last part of
movie. the movie.
C->W: GET /twister.sdp HTTP/1.1 C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com Host: www.example.com
Accept: application/sdp Accept: application/sdp
W->C: HTTP/1.0 200 OK W->C: HTTP/1.0 200 OK
Content-Type: application/sdp Content-Type: application/sdp
v=0 v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202 o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session s=RTSP Session
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31 m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1 CSeq: 1
skipping to change at page 67, line 39 skipping to change at page 64, line 18
a=control:rtsp://audio.example.com/twister/audio.en a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31 m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1 CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057 Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
A->C: RTSP/1.0 200 OK A->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Session: 1234 Session: 12345678
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057; Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
server_port=5000-5001 server_port=5000-5001
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 1 CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059 Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
V->C: RTSP/1.0 200 OK V->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Session: 1235 Session: 23456789
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059; Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
server_port=5002-5003 server_port=5002-5003
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 2 CSeq: 2
Session: 1235 Session: 23456789
Range: smpte=0:10:00- Range: smpte=0:10:00-
V->C: RTSP/1.0 200 OK V->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Session: 1235 Session: 23456789
Range: smpte=0:10:00-0:20:00 Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://video.example.com/twister/video; RTP-Info: url=rtsp://video.example.com/twister/video;
seq=12312232;rtptime=78712811 seq=12312232;rtptime=78712811
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 2 CSeq: 2
Session: 1234 Session: 12345678
Range: smpte=0:10:00- Range: smpte=0:10:00-
A->C: RTSP/1.0 200 OK A->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Session: 1234 Session: 12345678
Range: smpte=0:10:00-0:20:00 Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://audio.example.com/twister/audio.en; RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
seq=876655;rtptime=1032181 seq=876655;rtptime=1032181
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 3 CSeq: 3
Session: 1234 Session: 12345678
A->C: RTSP/1.0 200 OK A->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 3 CSeq: 3
Session: 1235 Session: 23456789
V->C: RTSP/1.0 200 OK V->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
Even though the audio and video track are on two different servers, Even though the audio and video track are on two different servers,
and may start at slightly different times and may drift with respect and may start at slightly different times and may drift with respect
to each other, the client can synchronize the two using standard RTP to each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender methods, in particular the time scale contained in the RTCP sender
reports. reports.
14.2 Streaming of a Container file 14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in For purposes of this example, a container file is a storage entity in
which multiple continuous media types pertaining to the same end-user which multiple continuous media types pertaining to the same end-user
presentation are present. In effect, the container file represents a presentation are present. In effect, the container file represents an
RTSP presentation, with each of its components being RTSP streams. RTSP presentation, with each of its components being RTSP streams.
Container files are a widely used means to store such presentations. Container files are a widely used means to store such presentations.
While the components are transported as independent streams, it is While the components are transported as independent streams, it is
desirable to maintain a common context for those streams at the server desirable to maintain a common context for those streams at the
end. server end.
This enables the server to keep a single storage handle open This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case of easily. It also allows treating all the streams equally in case of
any prioritization of streams by the server. any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by the client in order to preserve selective retrieval of the streams by the client in order to preserve
the artistic effect of the combined media presentation. Similarly, in the artistic effect of the combined media presentation. Similarly, in
such a tightly bound presentation, it is desirable to be able to such a tightly bound presentation, it is desirable to be able to
control all the streams via a single control message using an control all the streams via a single control message using an
aggregate URL. aggregate URL.
The following is an example of using a single RTSP session to control The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URLs. multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M . The movie is Client C requests a presentation from media server M . The movie is
stored in a container file. The client has obtained a RTSP URL to the stored in a container file. The client has obtained an RTSP URL to
container file. the container file.
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0 C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
CSeq: 1 CSeq: 1
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 164 Content-Length: 164
v=0 v=0
o=- 2890844256 2890842807 IN IP4 172.16.2.93 o=- 2890844256 2890842807 IN IP4 172.16.2.93
s=RTSP Session s=RTSP Session
i=An Example of RTSP Session Usage i=An Example of RTSP Session Usage
a=control:rtsp://foo/twister a=control:rtsp://foo/twister
t=0 0 t=0 0
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=control:rtsp://foo/twister/audio a=control:rtsp://foo/twister/audio
m=video 0 RTP/AVP 26 m=video 0 RTP/AVP 26
a=control:rtsp://foo/twister/video a=control:rtsp://foo/twister/video
skipping to change at page 70, line 23 skipping to change at page 66, line 36
a=control:rtsp://foo/twister/video a=control:rtsp://foo/twister/video
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0 C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
CSeq: 2 CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001 Transport: RTP/AVP;unicast;client_port=8000-8001
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001; Transport: RTP/AVP;unicast;client_port=8000-8001;
server_port=9000-9001 server_port=9000-9001
Session: 1234 Session: 12345678
C->M: SETUP rtsp://foo/twister/video RTSP/1.0 C->M: SETUP rtsp://foo/twister/video RTSP/1.0
CSeq: 3 CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003 Transport: RTP/AVP;unicast;client_port=8002-8003
Session: 1234 Session: 12345678
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003; Transport: RTP/AVP;unicast;client_port=8002-8003;
server_port=9004-9005 server_port=9004-9005
Session: 1234 Session: 12345678
C->M: PLAY rtsp://foo/twister RTSP/1.0 C->M: PLAY rtsp://foo/twister RTSP/1.0
CSeq: 4 CSeq: 4
Range: npt=0- Range: npt=0-
Session: 1234 Session: 12345678
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 4 CSeq: 4
Session: 1234 Session: 12345678
RTP-Info: url=rtsp://foo/twister/video; RTP-Info: url=rtsp://foo/twister/video;
seq=9810092;rtptime=3450012 seq=9810092;rtptime=3450012
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0 C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
CSeq: 5 CSeq: 5
Session: 1234 Session: 12345678
M->C: RTSP/1.0 460 Only aggregate operation allowed M->C: RTSP/1.0 460 Only aggregate operation allowed
CSeq: 5 CSeq: 5
C->M: PAUSE rtsp://foo/twister RTSP/1.0 C->M: PAUSE rtsp://foo/twister RTSP/1.0
CSeq: 6 CSeq: 6
Session: 1234 Session: 12345678
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 6 CSeq: 6
Session: 1234 Session: 12345678
C->M: SETUP rtsp://foo/twister RTSP/1.0 C->M: SETUP rtsp://foo/twister RTSP/1.0
CSeq: 7 CSeq: 7
Transport: RTP/AVP;unicast;client_port=10000 Transport: RTP/AVP;unicast;client_port=10000
M->C: RTSP/1.0 459 Aggregate operation not allowed M->C: RTSP/1.0 459 Aggregate operation not allowed
CSeq: 7 CSeq: 7
In the first instance of failure, the client tries to pause one stream In the first instance of failure, the client tries to pause one
(in this case video) of the presentation. This is disallowed for that stream (in this case video) of the presentation. This is disallowed
presentation by the server. In the second instance, the aggregate URL for that presentation by the server. In the second instance, the
may not be used for SETUP and one control message is required per aggregate URL may not be used for SETUP and one control message is
stream to set up transport parameters. required per stream to set up transport parameters.
This keeps the syntax of the Transport header simple and allows This keeps the syntax of the Transport header simple and allows
easy parsing of transport information by firewalls. easy parsing of transport information by firewalls.
14.3 Single Stream Container Files 14.3 Single Stream Container Files
Some RTSP servers may treat all files as though they are ``container Some RTSP servers may treat all files as though they are "container
files'', yet other servers may not support such a concept. Because of files", yet other servers may not support such a concept. Because of
this, clients SHOULD use the rules set forth in the session this, clients SHOULD use the rules set forth in the session
description for request URLs, rather than assuming that a consistant description for request URLs, rather than assuming that a consistent
URL may always be used throughout. Here's an example of how a URL may always be used throughout. Here's an example of how a multi-
multi-stream server might expect a single-stream file to be served: stream server might expect a single-stream file to be served:
C->S DESCRIBE rtsp://foo.com/test.wav RTSP/1.0
Accept: application/x-rtsp-mh, application/sdp Accept: application/x-rtsp-mh, application/sdp
CSeq: 1 CSeq: 1
S->C RTSP/1.0 200 OK S->C RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Content-base: rtsp://foo.com/test.wav/ Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp Content-type: application/sdp
Content-length: 48 Content-length: 48
v=0 v=0
o=- 872653257 872653257 IN IP4 172.16.2.187 o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file s=mu-law wave file
i=audio test i=audio test
t=0 0 t=0 0
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=control:streamid=0 a=control:streamid=0
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0 C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast; Transport: RTP/AVP/UDP;unicast;
skipping to change at page 72, line 36 skipping to change at page 68, line 44
S->C RTSP/1.0 200 OK S->C RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
Session: 2034820394 Session: 2034820394
RTP-Info: url=rtsp://foo.com/test.wav/streamid=0; RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
seq=981888;rtptime=3781123 seq=981888;rtptime=3781123
Note the different URL in the SETUP command, and then the switch back 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 to the aggregate URL in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but is less when there are multiple streams with aggregate control, but is less
than intuitive in the special case where the number of streams is one. 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 In this special case, it is recommended that servers be forgiving of
implementations that send: implementations that send:
C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0 C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
CSeq: 3 CSeq: 3
In the worst case, servers should send back: In the worst case, servers should send back:
S->C RTSP/1.0 460 Only aggregate operation allowed S->C RTSP/1.0 460 Only aggregate operation allowed
skipping to change at page 75, line 23 skipping to change at page 71, line 19
CSeq: 3 CSeq: 3
Session: 91389234234 Session: 91389234234
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
14.6 Recording 14.6 Recording
The conference participant client C asks the media server M to record The conference participant client C asks the media server M to record
the audio and video portions of a meeting. The client uses the the audio and video portions of a meeting. The client uses the
ANNOUNCE method to provide meta-information about the recorded session ANNOUNCE method to provide meta-information about the recorded
to the server. session to the server.
C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0 C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
CSeq: 90 CSeq: 90
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 121 Content-Length: 121
v=0 v=0
o=camera1 3080117314 3080118787 IN IP4 195.27.192.36 o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
s=IETF Meeting, Munich - 1 s=IETF Meeting, Munich - 1
i=The thirty-ninth IETF meeting will be held in Munich, Germany i=The thirty-ninth IETF meeting will be held in Munich, Germany
skipping to change at page 76, line 12 skipping to change at page 72, line 7
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 90 CSeq: 90
C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0 C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
CSeq: 91 CSeq: 91
Transport: RTP/AVP;multicast;destination=224.0.1.11; Transport: RTP/AVP;multicast;destination=224.0.1.11;
port=21010-21011;mode=record;ttl=127 port=21010-21011;mode=record;ttl=127
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 91 CSeq: 91
Session: 508876 Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.11; Transport: RTP/AVP;multicast;destination=224.0.1.11;
port=21010-21011;mode=record;ttl=127 port=21010-21011;mode=record;ttl=127
C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0 C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
CSeq: 92 CSeq: 92
Session: 508876 Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.12; Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127 port=61010-61011;mode=record;ttl=127
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 92 CSeq: 92
Transport: RTP/AVP;multicast;destination=224.0.1.12; Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127 port=61010-61011;mode=record;ttl=127
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
CSeq: 93 CSeq: 93
Session: 508876 Session: 50887676
Range: clock=19961110T1925-19961110T2015 Range: clock=19961110T1925-19961110T2015
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 93 CSeq: 93
15 Syntax 15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF) The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 [2]. as used in RFC 2068 [2].
15.1 Base Syntax 15.1 Base Syntax
OCTET = <any 8-bit sequence of data> OCTET = <any 8-bit sequence of data>
CHAR = <any US-ASCII character (octets 0 - 127)> CHAR = <any US-ASCII character (octets 0 - 127)>
UPALPHA = <any US-ASCII uppercase letter "A".."Z"> UPALPHA = <any US-ASCII uppercase letter "A".."Z">
LOALPHA = <any US-ASCII lowercase letter "a".."z"> LOALPHA = <any US-ASCII lowercase letter "a".."z">
ALPHA = UPALPHA | LOALPHA ALPHA = UPALPHA | LOALPHA
DIGIT = <any US-ASCII digit "0".."9"> DIGIT = <any US-ASCII digit "0".."9">
CTL = <any US-ASCII control character CTL = <any US-ASCII control character
(octets 0 - 31) and DEL (127)> (octets 0 - 31) and DEL (127)>
CR = <US-ASCII CR, carriage return (13)> CR = <US-ASCII CR, carriage return (13)>
LF = <US-ASCII LF, linefeed (10)> LF = <US-ASCII LF, linefeed (10)>
SP = <US-ASCII SP, space (32)> SP = <US-ASCII SP, space (32)>
HT = <US-ASCII HT, horizontal-tab (9)> HT = <US-ASCII HT, horizontal-tab (9)>
<"> = <US-ASCII double-quote mark (34)> <"> = <US-ASCII double-quote mark (34)>
CRLF = CR LF CRLF = CR LF
skipping to change at page 78, line 7 skipping to change at page 73, line 37
hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" | hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
"a" | "b" | "c" | "d" | "e" | "f" "a" | "b" | "c" | "d" | "e" | "f"
escape = "\%" hex hex escape = "\%" hex hex
reserved = ";" | "/" | "?" | ":" | "@" | "&" | "=" reserved = ";" | "/" | "?" | ":" | "@" | "&" | "="
unreserved = alpha | digit | safe | extra unreserved = alpha | digit | safe | extra
xchar = unreserved | reserved | escape xchar = unreserved | reserved | escape
16 Security Considerations 16 Security Considerations
Because of the similarity in syntax and usage between RTSP servers Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [H15] apply. and HTTP servers, the security considerations outlined in [H15]
Specifically, please note the following: apply. Specifically, please note the following:
Authentication Mechanisms: Authentication Mechanisms:
RTSP and HTTP share common authentication schemes, and thus RTSP and HTTP share common authentication schemes, and thus
should follow the same prescriptions with regards to should follow the same prescriptions with regards to
authentication. See [H15.1] for client authentication issues, authentication. See [H15.1] for client authentication issues,
and [H15.2] for issues regarding support for multiple and [H15.2] for issues regarding support for multiple
authentication mechanisms. authentication mechanisms.
Abuse of Server Log Information: Abuse of Server Log Information:
RTSP and HTTP servers will presumably have similar logging RTSP and HTTP servers will presumably have similar logging
mechanisms, and thus should be equally guarded in protecting mechanisms, and thus should be equally guarded in protecting
the contents of those logs, thus protecting the privacy of the the contents of those logs, thus protecting the privacy of the
users of the servers. See [H15.3] for HTTP server users of the servers. See [H15.3] for HTTP server
recommendations regarding server logs. recommendations regarding server logs.
Transfer of Sensitive Information: Transfer of Sensitive Information:
There is no reason to believe that information transferred via There is no reason to believe that information transferred via
RTSP may be any less sensitive than that normally transmitted RTSP may be any less sensitive than that normally transmitted
via HTTP. Therefore, all of the precautions regarding the via HTTP. Therefore, all of the precautions regarding the
protection of data privacy and user privacy apply to protection of data privacy and user privacy apply to
implementors of RTSP clients, servers, and proxies. See [H15.4] implementors of RTSP clients, servers, and proxies. See
for further details. [H15.4] for further details.
Attacks Based On File and Path Names: Attacks Based On File and Path Names:
Though RTSP URLs are opaque handles that do not necessarily Though RTSP URLs are opaque handles that do not necessarily
have file system semantics, it is anticipated that many have file system semantics, it is anticipated that many
implementations will translate portions of the request URLs implementations will translate portions of the request URLs
directly to file system calls. In such cases, file systems directly to file system calls. In such cases, file systems
SHOULD follow the precautions outlined in [H15.5], such as SHOULD follow the precautions outlined in [H15.5], such as
checking for ``..'' in path components. checking for ".." in path components.
Personal Information: Personal Information:
RTSP clients are often privy to the same information that HTTP RTSP clients are often privy to the same information that HTTP
clients are (user name, location, etc.) and thus should be clients are (user name, location, etc.) and thus should be
equally. See [H15.6] for further recommendations. equally. See [H15.6] for further recommendations.
Privacy Issues Connected to Accept Headers: Privacy Issues Connected to Accept Headers:
Since may of the same ``Accept'' headers exist in RTSP as in Since may of the same "Accept" headers exist in RTSP as in
HTTP, the same caveats outlined in [H15.7] with regards to HTTP, the same caveats outlined in [H15.7] with regards to
their use should be followed. their use should be followed.
DNS Spoofing: DNS Spoofing:
Presumably, given the longer connection times typically Presumably, given the longer connection times typically
associated to RTSP sessions relative to HTTP sessions, RTSP associated to RTSP sessions relative to HTTP sessions, RTSP
client DNS optimizations should be less prevalent. Nonetheless, client DNS optimizations should be less prevalent.
the recommendations provided in [H15.8] are still relevant to Nonetheless, the recommendations provided in [H15.8] are still
any implementation which attempts to rely on a DNS-to-IP relevant to any implementation which attempts to rely on a
mapping to hold beyond a single use of the mapping. DNS-to-IP mapping to hold beyond a single use of the mapping.
Location Headers and Spoofing: Location Headers and Spoofing:
If a single server supports multiple organizations that do not If a single server supports multiple organizations that do not
trust one another, then it must check the values of Location trust one another, then it must check the values of Location
and Content-Location headers in responses that are generated and Content-Location headers in responses that are generated
under control of said organizations to make sure that they do under control of said organizations to make sure that they do
not attempt to invalidate resources over which they have no not attempt to invalidate resources over which they have no
authority. ([H15.9]) authority. ([H15.9])
In addition to the recommendations in the current HTTP specification In addition to the recommendations in the current HTTP specification
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The following are added considerations for RTSP implementations. The following are added considerations for RTSP implementations.
Concentrated denial-of-service attack: Concentrated denial-of-service attack:
The protocol offers the opportunity for a remote-controlled The protocol offers the opportunity for a remote-controlled
denial-of-service attack. The attacker may initiate traffic denial-of-service attack. The attacker may initiate traffic
flows to one or more IP addresses by specifying them as the flows to one or more IP addresses by specifying them as the
destination in SETUP requests. While the attacker's IP address destination in SETUP requests. While the attacker's IP address
may be known in this case, this is not always useful in may be known in this case, this is not always useful in
prevention of more attacks or ascertaining the attackers prevention of more attacks or ascertaining the attackers
identity. Thus, an RTSP server SHOULD only allow identity. Thus, an RTSP server SHOULD only allow client-
client-specified destinations for RTSP-initiated traffic flows specified destinations for RTSP-initiated traffic flows if the
if the server has verified the client's identity, either server has verified the client's identity, either against a
against a database of known users using RTSP authentication database of known users using RTSP authentication mechanisms
mechanisms (preferrably digest authentication or stronger), or (preferably digest authentication or stronger), or other
other secure means. secure means.
Session hijacking: Session hijacking:
Since there is no relation between a transport layer connection Since there is no relation between a transport layer
and an RTSP session, it is possible for a malicious client to connection and an RTSP session, it is possible for a malicious
issue requests with random session identifiers which would client to issue requests with random session identifiers which
affect unsuspecting clients. The server SHOULD use a large, would affect unsuspecting clients. The server SHOULD use a
random and non-sequential session identifier to minimize the large, random and non-sequential session identifier to
possibility of this kind of attack. minimize the possibility of this kind of attack.
Authentication: Authentication:
Servers SHOULD implement both basic and digest [8] Servers SHOULD implement both basic and digest [8]
authentication. In environments requiring tighter security for authentication. In environments requiring tighter security for
the control messages, transport layer mechanisms such as TLS the control messages, the RTSP control stream may be
(RFC XXXX [7]) SHOULD be used. encrypted.
Stream issues: Stream issues:
RTSP only provides for stream control. Stream delivery issues RTSP only provides for stream control. Stream delivery issues
are not covered in this section, nor in the rest of this draft. are not covered in this section, nor in the rest of this memo.
RTSP implementations will most likely rely on other protocols RTSP implementations will most likely rely on other protocols
such as RTP, IP multicast, RSVP and IGMP, and should address such as RTP, IP multicast, RSVP and IGMP, and should address
security considerations brought up in those and other security considerations brought up in those and other
applicable specifications. applicable specifications.
Persistently suspicious behavior: Persistently suspicious behavior:
RTSP servers SHOULD return error code 403 (Forbidden) upon RTSP servers SHOULD return error code 403 (Forbidden) upon
receiving a single instance of behavior which is deemed a receiving a single instance of behavior which is deemed a
security risk. RTSP servers SHOULD also be aware of attempts to security risk. RTSP servers SHOULD also be aware of attempts
probe the server for weaknesses and entry points and MAY to probe the server for weaknesses and entry points and MAY
arbitrarily disconnect and ignore further requests clients arbitrarily disconnect and ignore further requests clients
which are deemed to be in violation of local security policy. which are deemed to be in violation of local security policy.
Appendix A: RTSP Protocol State Machines Appendix A: RTSP Protocol State Machines
The RTSP client and server state machines describe the behavior of The RTSP client and server state machines describe the behavior of
the protocol from RTSP session initialization through RTSP session the protocol from RTSP session initialization through RTSP session
termination. termination.
State is defined on a per object basis. An object is uniquely State is defined on a per object basis. An object is uniquely
identified by the stream URL and the RTSP session identifier. Any identified by the stream URL and the RTSP session identifier. Any
request/reply using aggregate URLs denoting RTSP presentations request/reply using aggregate URLs denoting RTSP presentations
composed of multiple streams will have an effect on the individual composed of multiple streams will have an effect on the individual
states of all the streams. For example, if the presentation /movie states of all the streams. For example, if the presentation /movie
contains two streams, /movie/audio and /movie/video, then the contains two streams, /movie/audio and /movie/video, then the
following command: following command:
PLAY rtsp://foo.com/movie RTSP/1.0 PLAY rtsp://foo.com/movie RTSP/1.0
CSeq: 559 CSeq: 559
Session: 12345 Session: 12345678
will have an effect on the states of movie/audio and movie/video. 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 This example does not imply a standard way to represent streams in
URLs or a relation to the filesystem. See Section 3.2. URLs or a relation to the filesystem. See Section 3.2.
The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER, SET_PARAMETER The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER,
do not have any effect on client or server state and are therefore not SET_PARAMETER do not have any effect on client or server state and
listed in the state tables. are therefore not listed in the state tables.
A.1 Client State Machine A.1 Client State Machine
The client can assume the following states: The client can assume the following states:
Init: Init:
SETUP has been sent, waiting for reply. SETUP has been sent, waiting for reply.
Ready: Ready:
SETUP reply received or PAUSE reply received while in Playing SETUP reply received or PAUSE reply received while in Playing
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Playing: Playing:
PLAY reply received PLAY reply received
Recording: Recording:
RECORD reply received RECORD reply received
In general, the client changes state on receipt of replies to In general, the client changes state on receipt of replies to
requests. Note that some requests are effective at a future time or requests. Note that some requests are effective at a future time or
position (such as a PAUSE), and state also changes accordingly. If no position (such as a PAUSE), and state also changes accordingly. If no
explicit SETUP is required for the object (for example, it is explicit SETUP is required for the object (for example, it is
available via a multicast group), state begins at Ready. In this case, available via a multicast group), state begins at Ready. In this
there are only two states, Ready and Playing. The client also changes case, there are only two states, Ready and Playing. The client also
state from Playing/Recording to Ready when the end of the requested changes state from Playing/Recording to Ready when the end of the
range is reached. requested range is reached.
The ``next state'' column indicates the state assumed after receiving The "next state" column indicates the state assumed after receiving a
a success response (2xx). If a request yields a status code of 3xx, success response (2xx). If a request yields a status code of 3xx, the
the state becomes Init, and a status code of 4xx yields no change in 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 state. Messages not listed for each state MUST NOT be issued by the
client in that state, with the exception of messages not affecting client in that state, with the exception of messages not affecting
state, as listed above. Receiving a REDIRECT from the server is state, as listed above. Receiving a REDIRECT from the server is
equivalent to receiving a 3xx redirect status from the server. equivalent to receiving a 3xx redirect status from the server.
state message sent next state after response state message sent next state after response
Init SETUP Ready Init SETUP Ready
TEARDOWN Init TEARDOWN Init
Ready PLAY Playing Ready PLAY Playing
RECORD Recording RECORD Recording
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Playing: Playing:
Last PLAY received was successful, reply sent. Data is being Last PLAY received was successful, reply sent. Data is being
sent. sent.
Recording: Recording:
The server is recording media data. The server is recording media data.
In general, the server changes state on receiving requests. If the In general, the server changes state on receiving requests. If the
server is in state Playing or Recording and in unicast mode, it MAY 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 revert to Init and tear down the RTSP session if it has not received
``wellness'' information, such as RTCP reports or RTSP commands, from "wellness" information, such as RTCP reports or RTSP commands, from
the client for a defined interval, with a default of one minute. The the client for a defined interval, with a default of one minute. The
server can declare another timeout value in the Session response server can declare another timeout value in the Session response
header (Section 12.37). If the server is in state Ready, it MAY revert header (Section 12.37). If the server is in state Ready, it MAY
to Init if it does not receive an RTSP request for an interval of more revert to Init if it does not receive an RTSP request for an interval
than one minute. Note that some requests (such as PAUSE) may be of more than one minute. Note that some requests (such as PAUSE) may
effective at a future time or position, and server state changes at be effective at a future time or position, and server state changes
the appropriate time. The server reverts from state Playing or at the appropriate time. The server reverts from state Playing or
Recording to state Ready at the end of the range requested by the Recording to state Ready at the end of the range requested by the
client. client.
The REDIRECT message, when sent, is effective immediately unless it The REDIRECT message, when sent, is effective immediately unless it
has a Range header specifying when the redirect is effective. In such has a Range header specifying when the redirect is effective. In such
a case, server state will also change at the appropriate time. a case, server state will also change at the appropriate time.
If no explicit SETUP is required for the object, the state starts at If no explicit SETUP is required for the object, the state starts at
Ready and there are only two states, Ready and Playing. Ready and there are only two states, Ready and Playing.
The ``next state'' column indicates the state assumed after sending a The "next state" column indicates the state assumed after sending a
success response (2xx). If a request results in a status code of 3xx, 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. the state becomes Init. A status code of 4xx results in no change.
state message received next state state message received next state
Init SETUP Ready Init SETUP Ready
TEARDOWN Init TEARDOWN Init
Ready PLAY Playing Ready PLAY Playing
SETUP Ready SETUP Ready
TEARDOWN Init TEARDOWN Init
RECORD Recording RECORD Recording
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PAUSE Ready PAUSE Ready
TEARDOWN Init TEARDOWN Init
SETUP Playing SETUP Playing
Recording RECORD Recording Recording RECORD Recording
PAUSE Ready PAUSE Ready
TEARDOWN Init TEARDOWN Init
SETUP Recording SETUP Recording
Appendix B: Interaction with RTP Appendix B: Interaction with RTP
RTSP allows media clients to control selected, non-contiguous RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP sections of media presentations, rendering those streams with an RTP
media layer[24]. The media layer rendering the RTP stream should not 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 be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
timestamps MUST be continuous and monotonic across jumps of NPT. timestamps MUST be continuous and monotonic across jumps of NPT.
As an example, assume a clock frequency of 8000 Hz, a packetization As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms and an initial sequence number and timestamp of 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 zero. First we play NPT 10 through 15, then skip ahead and play NPT
through 20. The first segment is presented as RTP packets with 18 through 20. The first segment is presented as RTP packets with
sequence numbers 0 through 49 and timestamp 0 through 39,200. The sequence numbers 0 through 49 and timestamp 0 through 39,200. The
second segment consists of RTP packets with sequence number 50 through second segment consists of RTP packets with sequence number 50
69, with timestamps 40,000 through 55,200. through 69, with timestamps 40,000 through 55,200.
We cannot assume that the RTSP client can communicate with the RTP We cannot assume that the RTSP client can communicate with the RTP
media agent, as the two may be independent processes. If 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 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 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 NPT is large enough, the RTP timestamp may roll over and the media
agent may believe later packets to be duplicates of packets just agent may believe later packets to be duplicates of packets just
played out. played out.
For certain datatypes, tight integration between the RTSP layer and For certain datatypes, tight integration between the RTSP layer and
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For scaling (see Section 12.34), RTP timestamps should correspond to For scaling (see Section 12.34), RTP timestamps should correspond to
the playback timing. For example, when playing video recorded at 30 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 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 server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame. would increase by 1/15 second for each video frame.
The client can maintain a correct display of NPT by noting the RTP The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning. The timestamp value of the first packet arriving after repositioning. The
sequence parameter of the RTP-Info (Section 12.33) header provides the sequence parameter of the RTP-Info (Section 12.33) header provides
first sequence number of the next segment. the first sequence number of the next segment.
Appendix C: Use of SDP for RTSP Session Descriptions Appendix C: Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, RFC XXXX [6]) may be used to The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
describe streams or presentations in RTSP. Such usage is limited to describe streams or presentations in RTSP. Such usage is limited to
specifying means of access and encoding(s) for: specifying means of access and encoding(s) for:
aggregate control: aggregate control:
A presentation composed of streams from one or more servers A presentation composed of streams from one or more servers
that are not available for aggregate control. Such a that are not available for aggregate control. Such a
description is typically retrieved by HTTP or other non-RTSP description is typically retrieved by HTTP or other non-RTSP
means. However, they may be received with ANNOUNCE methods. means. However, they may be received with ANNOUNCE methods.
non-aggregate control: non-aggregate control:
A presentation composed of multiple streams from a single A presentation composed of multiple streams from a single
server that are available for aggregate control. Such a server that are available for aggregate control. Such a
description is typically returned in reply to a DESCRIBE description is typically returned in reply to a DESCRIBE
request on a URL, or received in an ANNOUNCE method. request on a URL, or received in an ANNOUNCE method.
This appendix describes how an SDP file, retrieved, for example, This appendix describes how an SDP file, retrieved, for example,
through HTTP, determines the operation of an RTSP session. It also through HTTP, determines the operation of an RTSP session. It also
describes how a client should interpret SDP content returned in reply describes how a client should interpret SDP content returned in reply
to a DESCRIBE request. SDP provides no mechanism by which a client can to a DESCRIBE request. SDP provides no mechanism by which a client
distinguish, without human guidance, between several media streams to can distinguish, without human guidance, between several media
be rendered simultaneously and a set of alternatives (e.g., two audio streams to be rendered simultaneously and a set of alternatives
streams spoken in different languages). (e.g., two audio streams spoken in different languages).
C.1 Definitions C.1 Definitions
The terms ``session-level'', ``media-level'' and other key/attribute The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in names and values used in this appendix are to be used as defined in
SDP (RFC XXXX [6]): SDP (RFC 2327 [6]):
C.1.1 Control URL C.1.1 Control URL
The ``a=control:'' attribute is used to convey the control URL. This The "a=control:" attribute is used to convey the control URL. This
attribute is used both for the session and media descriptions. If used attribute is used both for the session and media descriptions. If
for individual media, it indicates the URL to be used for controlling used for individual media, it indicates the URL to be used for
that particular media stream. If found at the session level, the controlling that particular media stream. If found at the session
attribute indicates the URL for aggregate control. level, the attribute indicates the URL for aggregate control.
Example: Example:
a=control:rtsp://example.com/foo a=control:rtsp://example.com/foo
This attribute may contain either relative and absolute URLs, This attribute may contain either relative and absolute URLs,
following the rules and conventions set out in RFC 1808 [25]. following the rules and conventions set out in RFC 1808 [25].
Implementations should look for a base URL in the following order: Implementations should look for a base URL in the following order:
1. The RTSP Content-Base field 1. The RTSP Content-Base field
2. The RTSP Content-Location field 2. The RTSP Content-Location field
3. The RTSP request URL 3. The RTSP request URL
If this attribute contains only an asterisk (*), then the URL is If this attribute contains only an asterisk (*), then the URL is
treated as if it were an empty embedded URL, and thus inherits the treated as if it were an empty embedded URL, and thus inherits the
entire base URL. entire base URL.
C.1.2 Media streams C.1.2 Media streams
The ``m='' field is used to enumerate the streams. It is expected that The "m=" field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate all the specified streams will be rendered with appropriate
synchronization. If the session is unicast, the port number serves as synchronization. If the session is unicast, the port number serves as
a recommendation from the server to the client; the client still has a recommendation from the server to the client; the client still has
to include it in its SETUP request and may ignore this recommendation. to include it in its SETUP request and may ignore this
If the server has no preference, it SHOULD set the port number value recommendation. If the server has no preference, it SHOULD set the
to zero. port number value to zero.
Example: Example:
m=audio 0 RTP/AVP 31 m=audio 0 RTP/AVP 31
C.1.3 Payload type(s) C.1.3 Payload type(s)
The payload type(s) are specified in the ``m='' field. In case the 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 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 information is required. In case it is a dynamic payload type, the
media attribute ``rtpmap'' is used to specify what the media is. The media attribute "rtpmap" is used to specify what the media is. The
``encoding name'' within the ``rtpmap'' attribute may be one of those "encoding name" within the "rtpmap" attribute may be one of those
specified in RFC 1890 (Sections 5 and 6), or an experimental encoding specified in RFC 1890 (Sections 5 and 6), or an experimental encoding
with a ``X-'' prefix as specified in SDP (RFC XXXX [6]). with a "X-" prefix as specified in SDP (RFC 2327 [6]). Codec-
Codec-specific parameters are not specified in this field, but rather specific parameters are not specified in this field, but rather in
in the ``fmtp'' attribute described below. Implementors seeking to the "fmtp" attribute described below. Implementors seeking to
register new encodings should follow the procedure in RFC 1890 [1]. If register new encodings should follow the procedure in RFC 1890 [1].
the media type is not suited to the RTP AV profile, then it is 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 recommended that a new profile be created and the appropriate profile
name be used in lieu of ``RTP/AVP'' in the ``m='' field. name be used in lieu of "RTP/AVP" in the "m=" field.
C.1.4 Format-specific parameters C.1.4 Format-specific parameters
Format-specific parameters are conveyed using the ``fmtp'' media Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the ``fmtp'' attribute is specific to the attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) that the attribute refers to. Note that the packetization encoding(s) that the attribute refers to. Note that the packetization
interval is conveyed using the ``ptime'' attribute. interval is conveyed using the "ptime" attribute.
C.1.5 Range of presentation C.1.5 Range of presentation
The ``a=range'' attribute defines the total time range of the stored The "a=range" attribute defines the total time range of the stored
session. (The length of live sessions can be deduced from the ``t'' session. (The length of live sessions can be deduced from the "t" and
and ``r'' parameters.) Unless the presentation contains media streams "r" parameters.) Unless the presentation contains media streams of
of different durations, the length attribute is a session-level different durations, the range attribute is a session-level
attribute. The unit is specified first, followed by the value range. 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. The units and their values are as defined in Section 3.5, 3.6 and
3.7.
Examples: Examples:
a=range:npt=0-34.4368 a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203 a=range:clock=19971113T2115-19971113T2203
C.1.6 Time of availability C.1.6 Time of availability
The ``t='' field MUST contain suitable values for the start and stop The "t=" field MUST contain suitable values for the start and stop
times for both aggregate and non-aggregate stream control. With times for both aggregate and non-aggregate stream control. With
aggregate control, the server SHOULD indicate a stop time value for aggregate control, the server SHOULD indicate a stop time value for
which it guarantees the description to be valid, and a start time that which it guarantees the description to be valid, and a start time
is equal to or before the time at which the DESCRIBE request was 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 received. It MAY also indicate start and stop times of 0, meaning
the session is always available. With non-aggregate control, the that the session is always available. With non-aggregate control, the
values should reflect the actual period for which the session is values should reflect the actual period for which the session is
available in keeping with SDP semantics, and not depend on other means available in keeping with SDP semantics, and not depend on other
(such as the life of the web page containing the description) for this means (such as the life of the web page containing the description)
purpose. for this purpose.
C.1.7 Connection Information C.1.7 Connection Information
In SDP, the ``c='' field contains the destination address for the In SDP, the "c=" field contains the destination address for the media
media stream. However, for on-demand unicast streams and some stream. However, for on-demand unicast streams and some multicast
multicast streams, the destination address is specified by the client streams, the destination address is specified by the client via the
via the SETUP request. Unless the media content has a fixed SETUP request. Unless the media content has a fixed destination
destination address, the ``c='' field is to be set to a suitable null address, the "c=" field is to be set to a suitable null value. For
value. For addresses of type ``IP4'', this value is ``0.0.0.0''. addresses of type "IP4", this value is "0.0.0.0".
C.1.8 Entity Tag C.1.8 Entity Tag
The optional ``a=etag'' attribute identifies a version of the session The optional "a=etag" attribute identifies a version of the session
description. It is opaque to the client. SETUP requests may include description. It is opaque to the client. SETUP requests may include
this identifier in the If-Match field (see section 12.22) to only this identifier in the If-Match field (see section 12.22) to only
allow session establishment if this attribute value still corresponds allow session establishment if this attribute value still corresponds
to that of the current description. The attribute value is opaque and to that of the current description. The attribute value is opaque and
may contain any character allowed within SDP attribute values. may contain any character allowed within SDP attribute values.
Example: Example:
a=etag:158bb3e7c7fd62ce67f12b533f06b83a a=etag:158bb3e7c7fd62ce67f12b533f06b83a
One could argue that the "o=" field provides identical
One could argue that the ``o='' field provides identical
functionality. However, it does so in a manner that would put functionality. However, it does so in a manner that would put
constraints on servers that need to support multiple session constraints on servers that need to support multiple session
description types other than SDP for the same piece of media description types other than SDP for the same piece of media
content. content.
C.2 Aggregate Control Not Available C.2 Aggregate Control Not Available
If a presentation does not support aggregate control and multiple If a presentation does not support aggregate control and multiple
media sections are specified, each section MUST have the control URL media sections are specified, each section MUST have the control URL
specified via the ``a=control:'' attribute. specified via the "a=control:" attribute.
Example: Example:
v=0 v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32 o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I came from a web page s=I came from a web page
t=0 0 t=0 0
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
m=video 8002 RTP/AVP 31 m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3 m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid a=control:rtsp://video.com/movie.vid
Note that the position of the control URL in the description implies Note that the position of the control URL in the description implies
that the client establishes separate RTSP control sessions to the that the client establishes separate RTSP control sessions to the
servers audio.com and video.com. servers audio.com and video.com.
It is recommended that an SDP file contains the complete media It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media client initialization information even if it is delivered to the media
through non-RTSP means. This is necessary as there is no mechanism to client through non-RTSP means. This is necessary as there is no
indicate that the client should request more detailed media stream mechanism to indicate that the client should request more detailed
information via DESCRIBE. media stream information via DESCRIBE.
C.3 Aggregate Control Available C.3 Aggregate Control Available
In this scenario, the server has multiple streams that can be In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level controlled as a whole. In this case, there are both media-level
``a=control:'' attributes, which are used to specify the stream URLs, "a=control:" attributes, which are used to specify the stream URLs,
and a session-level ``a=control:'' attribute which is used as the and a session-level "a=control:" attribute which is used as the
request URL for aggregate control. If the media-level URL is relative, request URL for aggregate control. If the media-level URL is
it is resolved to absolute URLs according to Section C.1.1 above. 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 If the presentation comprises only a single stream, the media-level
``a=control:'' attribute may be omitted altogether. However, if the "a=control:" attribute may be omitted altogether. However, if the
presentation contains more than one stream, each media stream section presentation contains more than one stream, each media stream section
MUST contain its own ``a=control'' attribute. MUST contain its own "a=control" attribute.
Example: Example:
v=0 v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32 o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I contain s=I contain
i=<more info> i=<more info>
t=0 0 t=0 0
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=control:rtsp://example.com/movie/ a=control:rtsp://example.com/movie/
m=video 8002 RTP/AVP 31 m=video 8002 RTP/AVP 31
skipping to change at page 91, line 11 skipping to change at page 85, line 11
rtsp://example.com/movie/trackID=2 to set up the video and audio rtsp://example.com/movie/trackID=2 to set up the video and audio
streams, respectively. The URL rtsp://example.com/movie/ controls the streams, respectively. The URL rtsp://example.com/movie/ controls the
whole movie. whole movie.
Appendix D: Minimal RTSP implementation Appendix D: Minimal RTSP implementation
D.1 Client D.1 Client
A client implementation MUST be able to do the following : A client implementation MUST be able to do the following :
* Generate the following requests : * Generate the following requests: SETUP, TEARDOWN, and one of PLAY
SETUP, TEARDOWN, and one of PLAY (i.e., a minimal playback client) (i.e., a minimal playback client) or RECORD (i.e., a minimal
or RECORD (i.e., a minimal recording client). If RECORD is recording client). If RECORD is implemented, ANNOUNCE must be
implemented, ANNOUNCE must be implemented as well. implemented as well.
* Include the following headers in requests: * Include the following headers in requests: CSeq, Connection,
CSeq, Connection, Session, Transport. If ANNOUNCE is implemented, Session, Transport. If ANNOUNCE is implemented, the capability to
the capability to include headers Content-Language, include headers Content-Language, Content-Encoding, Content-
Content-Encoding, Content-Length, and Content-Type should be as Length, and Content-Type should be as well.
well.
* Parse and understand the following headers in responses: CSeq, * Parse and understand the following headers in responses: CSeq,
Connection, Session, Transport, Content-Language, Connection, Session, Transport, Content-Language, Content-
Content-Encoding, Content-Length, Content-Type. If RECORD is Encoding, Content-Length, Content-Type. If RECORD is implemented,
implemented, the Location header must be understood as well. the Location header must be understood as well. RTP-compliant
RTP-compliant implementations should also implement RTP-Info. implementations should also implement RTP-Info.
* Understand the class of each error code received and notify the * Understand the class of each error code received and notify the
end-user, if one is present, of error codes in classes 4xx and end-user, if one is present, of error codes in classes 4xx and
5xx. The notification requirement may be relaxed if the end-user 5xx. The notification requirement may be relaxed if the end-user
explicitly does not want it for one or all status codes. explicitly does not want it for one or all status codes.
* Expect and respond to asynchronous requests from the server, such * Expect and respond to asynchronous requests from the server, such
as ANNOUNCE. This does not necessarily mean that it should as ANNOUNCE. This does not necessarily mean that it should
implement the ANNOUNCE method, merely that it MUST respond implement the ANNOUNCE method, merely that it MUST respond
positively or negatively to any request received from the server. positively or negatively to any request received from the server.
Though not required, the following are highly recommended at the time Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial of publication for practical interoperability with initial
implementations and/or to be a ``good citizen''. implementations and/or to be a "good citizen".
* Implement RTP/AVP/UDP as a valid transport. * Implement RTP/AVP/UDP as a valid transport.
* Inclusion of the User-Agent header. * Inclusion of the User-Agent header.
* Understand SDP session descriptions as defined in Appendix C * Understand SDP session descriptions as defined in Appendix C
* Accept media initialization formats (such as SDP) from standard * Accept media initialization formats (such as SDP) from standard
input, command line, or other means appropriate to the operating input, command line, or other means appropriate to the operating
environment to act as a ``helper application'' for other environment to act as a "helper application" for other
applications (such as web browsers). applications (such as web browsers).
There may be RTSP applications different from those initially There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for which envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict recommendations above serve only as guidelines instead of strict
requirements. requirements.
D.1.1 Basic Playback D.1.1 Basic Playback
To support on-demand playback of media streams, the client MUST To support on-demand playback of media streams, the client MUST
additionally be able to do the following: additionally be able to do the following:
* generate the PAUSE request; * generate the PAUSE request;
* implement the REDIRECT method, and the Location header. * implement the REDIRECT method, and the Location header.
D.1.2 Authentication-enabled D.1.2 Authentication-enabled
In order to access media presentations from RTSP servers that require In order to access media presentations from RTSP servers that require
authentication, the client MUST additionally be able to do the authentication, the client MUST additionally be able to do the
following: following:
* recognize the 401 status code; * recognize the 401 status code;
* parse and include the WWW-Authenticate header; * parse and include the WWW-Authenticate header;
* implement Basic Authentication and Digest Authentication. * implement Basic Authentication and Digest Authentication.
D.2 Server D.2 Server
A minimal server implementation MUST be able to do the following: A minimal server implementation MUST be able to do the following:
* Implement the following methods: SETUP, TEARDOWN, OPTIONS and * Implement the following methods: SETUP, TEARDOWN, OPTIONS and
either PLAY (for a minimal playback server) or RECORD (for a either PLAY (for a minimal playback server) or RECORD (for a
minimal recording server). minimal recording server). If RECORD is implemented, ANNOUNCE
If RECORD is implemented, ANNOUNCE should be implemented as well. should be implemented as well.
* Include the following headers in responses: Connection, * Include the following headers in responses: Connection,
Content-Length, Content-Type, Content-Language, Content-Encoding, Content-Length, Content-Type, Content-Language, Content-Encoding,
Transport, Public. The capability to include the Location header Transport, Public. The capability to include the Location header
should be implemented if the RECORD method is. RTP-compliant should be implemented if the RECORD method is. RTP-compliant
implementations should also implement the RTP-Info field. implementations should also implement the RTP-Info field.
* Parse and respond appropriately to the following headers in * Parse and respond appropriately to the following headers in
requests: Connection, Session, Transport, Require. requests: Connection, Session, Transport, Require.
Though not required, the following are highly recommended at the time Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial of publication for practical interoperability with initial
implementations and/or to be a ``good citizen''. implementations and/or to be a "good citizen".
* Implement RTP/AVP/UDP as a valid transport. * Implement RTP/AVP/UDP as a valid transport.
* Inclusion of the Server header. * Inclusion of the Server header.
* Implement the DESCRIBE method. * Implement the DESCRIBE method.
* Generate SDP session descriptions as defined in Appendix C * Generate SDP session descriptions as defined in Appendix C
There may be RTSP applications different from those initially There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for which envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict recommendations above serve only as guidelines instead of strict
requirements. requirements.
D.2.1 Basic Playback D.2.1 Basic Playback
To support on-demand playback of media streams, the server MUST To support on-demand playback of media streams, the server MUST
additionally be able to do the following: additionally be able to do the following:
* Recognize the Range header, and return an error if seeking is not * Recognize the Range header, and return an error if seeking is not
supported. supported.
* Implement the PAUSE method. * Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface In addition, in order to support commonly-accepted user interface
features, the following are highly recommended for on-demand media features, the following are highly recommended for on-demand media
servers: servers:
* Include and parse the Range header, with NPT units. Implementation * Include and parse the Range header, with NPT units.
of SMPTE units is recommended. Implementation of SMPTE units is recommended.
* Include the length of the media presentation in the media * Include the length of the media presentation in the media
initialization information. initialization information.
* Include mappings from data-specific timestamps to NPT. When RTP is * Include mappings from data-specific timestamps to NPT. When RTP
used, the rtptime portion of the RTP-Info field may be used to map is used, the rtptime portion of the RTP-Info field may be used to
RTP timestamps to NPT. map RTP timestamps to NPT.
Client implementations may use the presence of length information Client implementations may use the presence of length information
to determine if the clip is seekable, and visably disable seeking to determine if the clip is seekable, and visibly disable seeking
features for clips for which the length information is unavailable. features for clips for which the length information is unavailable.
A common use of the presentation length is to implement a ``slider A common use of the presentation length is to implement a "slider
bar'' which serves as both a progress indicator and a timeline bar" which serves as both a progress indicator and a timeline
positioning tool. positioning tool.
Mappings from RTP timestamps to NPT are necessary to ensure correct Mappings from RTP timestamps to NPT are necessary to ensure correct
positioning of the slider bar. positioning of the slider bar.
D.2.2 Authentication-enabled D.2.2 Authentication-enabled
In order to correctly handle client authentication, the server MUST In order to correctly handle client authentication, the server MUST
additionally be able to do the following: additionally be able to do the following:
* Generate the 401 status code when authentication is required for * Generate the 401 status code when authentication is required for
the resource. the resource.
* Parse and include the WWW-Authenticate header * Parse and include the WWW-Authenticate header
* Implement Basic Authentication and Digest Authentication * Implement Basic Authentication and Digest Authentication
Appendix E: Author Addresses Appendix E: Authors' Addresses
Henning Schulzrinne Henning Schulzrinne
Dept. of Computer Science Dept. of Computer Science
Columbia University Columbia University
1214 Amsterdam Avenue 1214 Amsterdam Avenue
New York, NY 10027 New York, NY 10027
USA USA
electronic mail: schulzrinne@cs.columbia.edu
EMail: schulzrinne@cs.columbia.edu
Anup Rao Anup Rao
Netscape Communications Corp. Netscape Communications Corp.
501 E. Middlefield Road 501 E. Middlefield Road
Mountain View, CA 94043 Mountain View, CA 94043
USA USA
electronic mail: anup@netscape.com
EMail: anup@netscape.com
Robert Lanphier Robert Lanphier
RealNetworks RealNetworks
1111 Third Avenue Suite 2900 1111 Third Avenue Suite 2900
Seattle, WA 98101 Seattle, WA 98101
USA USA
electronic mail: robla@prognet.com
EMail: robla@real.com
Appendix F: Acknowledgements Appendix F: Acknowledgements
This draft is based on the functionality of the original RTSP draft This memo is based on the functionality of the original RTSP document
submitted in October 96. It also borrows format and descriptions from submitted in October 96. It also borrows format and descriptions from
HTTP/1.1. HTTP/1.1.
This document has benefited greatly from the comments of all those This document has benefited greatly from the comments of all those
participating in the MMUSIC-WG. In addition to those already participating in the MMUSIC-WG. In addition to those already
mentioned, the following individuals have contributed to this mentioned, the following individuals have contributed to this
specification: specification:
Rahul Agarwal, Torsten Braun, Brent Browning, Bruce Butterfield, Ema Rahul Agarwal, Torsten Braun, Brent Browning, Bruce Butterfield,
Patki, Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Dunsmuir,
Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter
Peter Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka,
Hoschka, Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan
Jonathan Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Papadopouli,
Papadopouli, Sujal Patel, Alagu Periyannan, Igor Plotnikov, Pinaki Sujal Patel, Ema Patki, Alagu Periyannan, Igor Plotnikov, Pinaki
Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and
John Francis Stracke. John Francis Stracke.
References References
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with minimal control,'' RFC 1890, Internet Engineering Task with minimal control", RFC 1890, January 1996.
Force, Jan. 1996.
2 R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. 2 Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC Berners-Lee, "Hypertext transfer protocol - HTTP/1.1", RFC
2068, Internet Engineering Task Force, Jan. 1997. 2068, January 1997.
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``Internationalization of the hypertext markup language,'' RFC "Internationalization of the hypertext markup language", RFC
2070, Internet Engineering Task Force, Jan. 1997. 2070, January 1997.
4 S. Bradner, ``Key words for use in RFCs to indicate requirement 4 Bradner, S., "Key words for use in RFCs to indicate
levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997. requirement levels", BCP 14, RFC 2119, March 1997.
5 ISO/IEC, ``Information technology - generic coding of moving 5 ISO/IEC, "Information technology - generic coding of moving
pictures and associated audio informaiton - part 6: extension pictures and associated audio information - part 6: extension
for digital storage media and control,'' Draft International for digital storage media and control," Draft International
Standard ISO 13818-6, International Organization for Standard ISO 13818-6, International Organization for
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Nov. 1995. Nov. 1995.
6 M. Handley and V. Jacobson, ``SDP: Session description 6 Handley, M., and V. Jacobson, "SDP: Session Description
protocol,'' Request for Comments XXXX, Internet Engineering Protocol", RFC 2327, April 1998.
Task Force, Feb. 1998.
7 A. Freier, P. Karlton, and P. Kocher, ``The TLS protocol,''
Request for Comments XXXX, Internet Engineering Task Force,
Feb. 1998.
8 J. Franks, P. Hallam-Baker, and J. Hostetler, ``An extension to 7 Franks, J., Hallam-Baker, P., and J. Hostetler, "An extension to
HTTP: digest access authentication,'' RFC 2069, Internet HTTP: digest access authentication", RFC 2069, January 1997.
Engineering Task Force, Jan. 1997.
9 J. Postel, ``User datagram protocol,'' RFC STD 6, 768, Internet 8 Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
Engineering Task Force, Aug. 1980. 1980.
10 B. Hinden and C. Partridge, ``Version 2 of the reliable data 9 Hinden, B. and C. Partridge, "Version 2 of the reliable data
protocol (RDP),'' RFC 1151, Internet Engineering Task Force, protocol (RDP)", RFC 1151, April 1990.
Apr. 1990.
11 J. Postel, ``Transmission control protocol,'' RFC STD 7, 793, 10 Postel, J., "Transmission control protocol", STD 7, RFC 793,
Internet Engineering Task Force, Sept. 1981. September 1981.
12 H. Schulzrinne, ``A comprehensive multimedia control 11 H. Schulzrinne, "A comprehensive multimedia control
architecture for the Internet,'' in Proc. International architecture for the Internet," in Proc. International
Workshop on Network and Operating System Support for Digital Workshop on Network and Operating System Support for Digital
Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997. Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997.
13 International Telecommunication Union, ``Visual telephone 12 International Telecommunication Union, "Visual telephone
systems and equipment for local area networks which provide a systems and equipment for local area networks which provide a
non-guaranteed quality of service,'' Recommendation H.323, non-guaranteed quality of service," Recommendation H.323,
Telecommunication Standardization Sector of ITU, Geneva, Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, May 1996. Switzerland, May 1996.
14 P. McMahon, ``GSS-API authentication method for SOCKS version 13 McMahon, P., "GSS-API authentication method for SOCKS version
5,'' RFC 1961, Internet Engineering Task Force, June 1996. 5", RFC 1961, June 1996.
15 J. Miller, P. Resnick, and D. Singer, ``Rating services and 14 J. Miller, P. Resnick, and D. Singer, "Rating services and
rating systems (and their machine readable descriptions),'' rating systems (and their machine readable descriptions),"
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Consortium), Boston, Massachusetts, Oct. 1996. Consortium), Boston, Massachusetts, Oct. 1996.
16 J. Miller, T. Krauskopf, P. Resnick, and W. Treese, ``PICS 15 J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS
label distribution label syntax and communication protocols,'' label distribution label syntax and communication protocols,"
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Consortium), Boston, Massachusetts, Oct. 1996. Consortium), Boston, Massachusetts, Oct. 1996.
17 D. Crocker and P. Overell, ``Augmented BNF for syntax 16 Crocker, D. and P. Overell, "Augmented BNF for syntax
specifications: ABNF,'' RFC 2234, Internet Engineering Task specifications: ABNF", RFC 2234, November 1997.
Force, Nov. 1997.
18 B. Braden, ``Requirements for internet hosts - application and 17 Braden, B., "Requirements for internet hosts - application and
support,'' RFC STD 3, 1123, Internet Engineering Task Force, support", STD 3, RFC 1123, October 1989.
Oct. 1989.
19 R. Elz, ``A compact representation of IPv6 addresses,'' RFC 18 Elz, R., "A compact representation of IPv6 addresses", RFC
1924, Internet Engineering Task Force, Apr. 1996. 1924, April 1996.
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resource locators (URL),'' RFC 1738, Internet Engineering Task resource locators (URL)", RFC 1738, December 1994.
Force, Dec. 1994.
21 F. Yergeau, ``Utf-8, a transformation format of iso 10646,'' 20 Yergeau, F., "UTF-8, a transformation format of ISO 10646",
Request for Comments XXXX, Internet Engineering Task Force, RFC 2279, January 1998.
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22 B. Braden, ``T/TCP - TCP extensions for transactions functional 22 Braden, B., "T/TCP - TCP extensions for transactions
specification,'' RFC 1644, Internet Engineering Task Force, functional specification", RFC 1644, July 1994.
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23 W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2. 22 W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
Reading, Massachusetts: Addison-Wesley, 1994. Reading, Massachusetts: Addison-Wesley, 1994.
24 H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, 23 Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
``RTP: a transport protocol for real-time applications,'' RFC "RTP: a transport protocol for real-time applications", RFC
1889, Internet Engineering Task Force, Jan. 1996. 1889, January 1996.
25 R. Fielding, ``Relative uniform resource locators,'' RFC 1808, 24 Fielding, R., "Relative uniform resource locators", RFC 1808,
Internet Engineering Task Force, June 1995. June 1995.
Full Copyright Statement Full Copyright Statement
Copyright (C) The Internet Society (1997). All Rights Reserved. Copyright (C) The Internet Society (1998). All Rights Reserved.
This document and translations of it may be copied and furnished to This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published and or assist in its implementation may be prepared, copied, published
distributed, in whole or in part, without restriction of any kind, and distributed, in whole or in part, without restriction of any
provided that the above copyright notice and this paragraph are kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing document itself may not be modified in any way, such as by removing
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