draft-ietf-mmusic-rtsp-02.txt   draft-ietf-mmusic-rtsp-03.txt 
Internet Engineering Task Force MMUSIC WG Internet Engineering Task Force MMUSIC WG
Internet Draft H. Schulzrinne, A. Rao, R. Lanphier Internet Draft H. Schulzrinne, A. Rao, R. Lanphier
ietf-mmusic-rtsp-02.txt Columbia U./Netscape/Progressive Networks draft-ietf-mmusic-rtsp-03.txt Columbia U./Netscape/Progressive Networks
March 27, 1997 July 30, 1997 Expires: January 30, 1998
Expires: September 26, 1997
Real Time Streaming Protocol (RTSP) Real Time Streaming Protocol (RTSP)
STATUS OF THIS MEMO STATUS OF THIS MEMO
This document is an Internet-Draft. Internet-Drafts are working This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas, documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts. working documents as Internet-Drafts.
skipping to change at page 1, line 31 skipping to change at line 28
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To learn the current status of any Internet-Draft, please check the To learn the current status of any Internet-Draft, please check the
``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow ``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
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ftp.isi.edu (US West Coast). ftp.isi.edu (US West Coast).
Distribution of this document is unlimited. Distribution of this document is unlimited.
ABSTRACT Abstract:
The Real Time Streaming Protocol, or RTSP, is an The Real Time Streaming Protocol, or RTSP, is an application-level
application-level protocol for control over the delivery protocol for control over the delivery of data with real-time
of data with real-time properties. RTSP provides an properties. RTSP provides an extensible framework to enable
extensible framework to enable controlled, on-demand controlled, on-demand delivery of real-time data, such as audio and
delivery of real-time data, such as audio and video. video. Sources of data can include both live data feeds and stored
Sources of data can include both live data feeds and clips. This protocol is intended to control multiple data delivery
stored clips. This protocol is intended to control sessions, provide a means for choosing delivery channels such as UDP,
multiple data delivery sessions, provide a means for multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC
choosing delivery channels such as UDP, multicast UDP and 1889).
TCP, and delivery mechanisms based upon RTP (RFC 1889).
H. Schulzrinne, A. Rao, R. Lanphier Page 1
Contents
* 1 Introduction
+ 1.1 Purpose
+ 1.2 Requirements
+ 1.3 Terminology
+ 1.4 Protocol Properties
+ 1.5 Extending RTSP
+ 1.6 Overall Operation
+ 1.7 RTSP States
+ 1.8 Relationship with Other Protocols
* 2 Notational Conventions
* 3 Protocol Parameters
+ 3.1 RTSP Version
+ 3.2 RTSP URL
+ 3.3 Conference Identifiers
+ 3.4 Session Identifiers
+ 3.5 SMPTE Relative Timestamps
+ 3.6 Normal Play Time
+ 3.7 Absolute Time
* 4 RTSP Message
+ 4.1 Message Types
+ 4.2 Message Headers
+ 4.3 Message Body
+ 4.4 Message Length
* 5 General Header Fields
* 6 Request
+ 6.1 Request Line
+ 6.2 Request Header Fields
* 7 Response
+ 7.1 Status-Line
o 7.1.1 Status Code and Reason Phrase
o 7.1.2 Response Header Fields
* 8 Entity
+ 8.1 Entity Header Fields
+ 8.2 Entity Body
* 9 Connections
+ 9.1 Pipelining
+ 9.2 Reliability and Acknowledgements
* 10 Method Definitions
+ 10.1 OPTIONS
+ 10.2 DESCRIBE
+ 10.3 ANNOUNCE
+ 10.4 SETUP
+ 10.5 PLAY
+ 10.6 PAUSE
+ 10.7 TEARDOWN
+ 10.8 GET_PARAMETER
+ 10.9 SET_PARAMETER
+ 10.10 REDIRECT
H. Schulzrinne, A. Rao, R. Lanphier Page 2
+ 10.11 RECORD
+ 10.12 Embedded (Interleaved) Binary Data
* 11 Status Code Definitions
+ 11.1 Redirection 3xx
+ 11.2 Client Error 4xx
o 11.2.1 405 Method Not Allowed
o 11.2.2 451 Parameter Not Understood
o 11.2.3 452 Conference Not Found
o 11.2.4 453 Not Enough Bandwidth
o 11.2.5 45x Session Not Found
o 11.2.6 45x Method Not Valid in This State
o 11.2.7 45x Header Field Not Valid for Resource
o 11.2.8 45x Invalid Range
o 11.2.9 45x Parameter Is Read-Only
o 11.2.10 45x Aggregate operation not allowed
o 11.2.11 45x Only aggregate operation allowed
* 12 Header Field Definitions
+ 12.1 Accept
+ 12.2 Accept-Encoding
+ 12.3 Accept-Language
+ 12.4 Allow
+ 12.5 Authorization
+ 12.6 Bandwidth
+ 12.7 Blocksize
+ 12.8 C-PEP
+ 12.9 C-PEP-Info
+ 12.10 Cache-Control
+ 12.11 Conference
+ 12.12 Connection
+ 12.13 Content-Encoding
+ 12.14 Content-Language
+ 12.15 Content-Length
+ 12.16 Content-Type
+ 12.17 Date
+ 12.18 Expires
+ 12.19 From
+ 12.20 Host
+ 12.21 If-Modified-Since
+ 12.22 Last-Modified
+ 12.23 Location
+ 12.24 PEP
+ 12.25 PEP-Info
+ 12.26 Proxy-Authenticate
+ 12.27 Public
+ 12.28 Range
+ 12.29 Referer
+ 12.30 Retry-After
+ 12.31 Scale
+ 12.32 Speed
+ 12.33 Server
H. Schulzrinne, A. Rao, R. Lanphier Page 3
+ 12.34 Session
+ 12.35 Transport
+ 12.36 Transport-Info
+ 12.37 User-Agent
+ 12.38 Vary
+ 12.39 Via
+ 12.40 WWW-Authenticate
* 13 Caching
* 14 Examples
+ 14.1 Media on Demand (Unicast)
+ 14.2 Streaming of a Container file
+ 14.3 Live Media Presentation Using Multicast
+ 14.4 Playing media into an existing session
+ 14.5 Recording
* 15 Syntax
+ 15.1 Base Syntax
* 16 Security Considerations
* A RTSP Protocol State Machines
+ A.1 Client State Machine
+ A.2 Server State Machine
* B Open Issues
* C Changes
* D Author Addresses
* E Acknowledgements
* References
1 Introduction 1 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 9.11). 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.
H. Schulzrinne, A. Rao, R. Lanphier Page 4
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 The protocol is intentionally similar in syntax and operation to
HTTP/1.1, so that extension mechanisms to HTTP can in most cases also HTTP/1.1, so that extension mechanisms to HTTP can in most cases also
be added to RTSP. However, RTSP differs in a number of important be added to RTSP. However, RTSP differs in a number of important
aspects from HTTP: aspects from HTTP:
o 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
cases, as opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests.
* Data is carried out-of-band, by a different protocol. (There is an
exception to this.)
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts [3].
* The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1 carries
only the absolute path in the request and puts the host name in a
separate header field.
o An RTSP server needs to maintain state by default in almost This makes ``virtual hosting'' easier, where a single host with one
all cases, as opposed to the stateless nature of HTTP. (RTSP IP address hosts several document trees.
servers and clients MAY use the HTTP state maintenance
mechanism [2].)
o Both an RTSP server and client can issue requests.
o Data is carried out-of-band, by a different protocol. (There
is an exception to this.)
o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
8859-1, consistent with current HTML internationalization
efforts [3].
o The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1
carries only the absolute path in the request
This makes virtual hosting easier. However, this is
incompatible with HTTP/1.1, which may be a bad idea.
The protocol supports the following operations: The protocol supports the following operations:
Retrieval of media from media server: The client can request a Retrieval of media from media server:
presentation description via HTTP or some other method. If the The client can request a presentation description via HTTP or
presentation is being multicast, the presentation description some other method. If the presentation is being multicast, the
contains the multicast addresses and ports to be used for the presentation description contains the multicast addresses and
continuous media. If the presentation is to be sent only to the ports to be used for the continuous media. If the presentation
client via unicast, the client provides the destination for is to be sent only to the client via unicast, the client
security reasons. provides the destination for security reasons.
Invitation of a media server to a conference: A media server can be H. Schulzrinne, A. Rao, R. Lanphier Page 5
"invited" to join an existing conference, either to play back Invitation of a media server to a conference:
media into the presentation or to record all or a subset of the A media server can be ``invited'' to join an existing
media in a presentation. This mode is useful for distributed conference, either to play back media into the presentation or
teaching applications. Several parties in the conference may to record all or a subset of the media in a presentation. This
take turns "pushing the remote control buttons". mode is useful for distributed teaching applications. Several
parties in the conference may take turns ``pushing the remote
control buttons''.
Addition of media to an existing presentation: Particularly for live Addition of media to an existing presentation:
presentations, it is useful if the server can tell the client Particularly for live presentations, it is useful if the server
about additional media becoming available. 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. HTTP/1.1.
1.2 Requirements 1.2 Requirements
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", The key words ``MUST'', ``MUST NOT'', ``REQUIRED'', ``SHALL'', ``SHALL
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this NOT'', ``SHOULD'', ``SHOULD NOT'', ``RECOMMENDED'', ``MAY'', and
document are to be interpreted as described in RFC xxxx [4]. ``OPTIONAL'' in this document are to be interpreted as described in
RFC 2119 [4].
1.3 Terminology 1.3 Terminology
Some of the terminology has been adopted from HTTP/1.1 [5]. Terms Some of the terminology has been adopted from HTTP/1.1 [5]. Terms not
not listed here are defined as in HTTP/1.1. listed here are defined as in HTTP/1.1.
Conference: a multiparty, multimedia presentation, where "multi" Conference:
implies greater than or equal to one. a multiparty, multimedia presentation, where ``multi'' implies
greater than or equal to one.
Client: The client requests continuous media data from the media Client:
The client requests continuous media data from the media
server. server.
Connection: A transport layer virtual circuit established between two Connection:
A transport layer virtual circuit established between two
programs for the purpose of communication. programs for the purpose of communication.
Continuous media: Data where there is a timing relationship between H. Schulzrinne, A. Rao, R. Lanphier Page 6
source and sink, that is, the sink must reproduce the timing Continuous media:
relationshop that existed at the source. The most common Data where there is a timing relationship between source and
examples of continuous media are audio and motion video. sink, that is, the sink must reproduce the timing relationshop
Continuous media can be realtime (interactive) , where there is that existed at the source. The most common examples of
a "tight" timing relationship between source and sink, or continuous media are audio and motion video. Continuous media
streaming (playback) , where the relationship is less strict. can be realtime (interactive), where there is a ``tight''
timing relationship between source and sink, or streaming
(playback), where the relationship is less strict.
Participant: Participants are members of conferences. A participant Participant:
may be a machine, e.g., a media record or playback server. Participants are members of conferences. A participant may be a
machine, e.g., a media record or playback server.
Media server: The network entity providing playback or recording Media server:
services for one or more media streams. Different media streams The network entity providing playback or recording services for
within a presentation may originate from different media one or more media streams. Different media streams within a
servers. A media server may reside on the same or a different presentation may originate from different media servers. A
host as the web server the presentation is invoked from. media server may reside on the same or a different host as the
web server the presentation is invoked from.
Media parameter: Parameter specific to a media type that may be Media parameter:
changed while the stream is being played or prior to it. Parameter specific to a media type that may be changed while
the stream is being played or prior to it.
(Media) stream: A single media instance, e.g., an audio stream or a (Media) stream:
video stream as well as a single whiteboard or shared A single media instance, e.g., an audio stream or a video
application group. When using RTP, a stream consists of all RTP stream as well as a single whiteboard or shared application
and RTCP packets created by a source within an RTP session. This group. When using RTP, a stream consists of all RTP and RTCP
is equivalent to the definition of a DSM-CC stream. packets created by a source within an RTP session. This is
equivalent to the definition of a DSM-CC stream([18]).
Message: The basic unit of RTSP communication, consisting of a Message:
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 14 and transmitted via a connection or a connectionless Section 15 and transmitted via a connection or a connectionless
protocol. protocol.
Presentation: A set of one or more streams which the server allows Presentation:
the client to manipulate together. A presentation has a single A set of one or more streams which the server allows the client
time axis for all streams belonging to it. Presentations are to manipulate together. A presentation has a single time axis
defined by presentation descriptions (see below). A presentation for all streams belonging to it. Presentations are defined by
presentation descriptions (see below). A presentation
description contains RTSP URIs that define which streams can be description contains RTSP URIs that define which streams can be
controlled individually and an RTSP URI to control the whole controlled individually and an RTSP URI to control the whole
presentation. A movie or live concert consisting of one or more presentation. A movie or live concert consisting of one or more
audio and video streams is be an example of a presentation. audio and video streams is an example of a presentation.
Presentation description: A presentation description contains H. Schulzrinne, A. Rao, R. Lanphier Page 7
information about one or more media streams within a Presentation description:
presentation, such as the set of encodings, network addresses A presentation description contains information about one or
and information about the content. Other IETF protocols such as more media streams within a presentation, such as the set of
SDP [6] use the term "session" for a live presentation. The encodings, network addresses and information about the content.
presentation description may take several different formats, Other IETF protocols such as SDP [6] use the term ``session''
including but not limited to the session description format SDP. for a live presentation. The presentation description may take
several different formats, including but not limited to the
session description format SDP.
Response: An RTSP response. If an HTTP response is meant, that is Response:
An RTSP response. If an HTTP response is meant, that is
indicated explicitly. indicated explicitly.
Request: An RTSP request. If an HTTP request is meant, that is Request:
indicated explicitly. An RTSP request. If an HTTP request is meant, that is indicated
explicitly.
RTSP session: A complete RTSP "transaction", e.g., the viewing of a RTSP session:
movie. A session typically consist of a client setting up a A complete RTSP ``transaction'', e.g., the viewing of a movie.
transport mechanism for the continuous media stream ( SETUP), A session typically consists of a client setting up a transport
starting the stream with PLAY or RECORD and closing the stream mechanism for the continuous media stream (SETUP), starting the
with TEARDOWN. stream with PLAY or RECORD and closing the stream with
TEARDOWN.
1.4 Protocol Properties 1.4 Protocol Properties
RTSP has the following properties: RTSP has the following properties:
Extendable: New methods and parameters can be easily added to RTSP. Extendable:
New methods and parameters can be easily added to RTSP.
Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers. Easy to parse:
RTSP can be parsed by standard HTTP or MIME parsers.
Secure: RTSP re-uses web security mechanisms, either at the transport Secure:
RTSP re-uses web security mechanisms, either at the transport
level (TLS [7]) or within the protocol itself. All HTTP level (TLS [7]) or within the protocol itself. All HTTP
authentication mechanisms such as basic [5] and digest authentication mechanisms such as basic [5, Section 11.1] and
authentication [8] are directly applicable. digest authentication [8] are directly applicable.
Transport-independent: RTSP may use either an unreliable datagram Transport-independent:
protocol (UDP) [9], a reliable datagram protocol (RDP, not RTSP may use either an unreliable datagram protocol (UDP) [9],
widely used [10]) or a reliable stream protocol such as TCP [11] a reliable datagram protocol (RDP, not widely used [10]) or a
as it implements application-level reliability. reliable stream protocol such as TCP [11] as it implements
application-level reliability.
Multi-server capable: Each media stream within a presentation can H. Schulzrinne, A. Rao, R. Lanphier Page 8
reside on a different server. The client automatically Multi-server capable:
establishes several concurrent control sessions with the Each media stream within a presentation can reside on a
different media servers. Media synchronization is performed at different server. The client automatically establishes several
the transport level. concurrent control sessions with the different media servers.
Media synchronization is performed at the transport level.
Control of recording devices: The protocol can control both recording Control of recording devices:
and playback devices, as well as devices that can alternate The protocol can control both recording and playback devices,
between the two modes ("VCR"). as well as devices that can alternate between the two modes
(``VCR'').
Separation of stream control and conference initiation: Stream Separation of stream control and conference initiation:
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
may be used to invite a server to a conference. may be used to invite a server to a conference.
Suitable for professional applications: RTSP supports frame-level Suitable for professional applications:
accuracy through SMPTE time stamps to allow remote digital RTSP supports frame-level accuracy through SMPTE time stamps to
editing. allow remote digital editing.
Presentation description neutral: The protocol does not impose a Presentation description neutral:
particular presentation description or metafile format and can The protocol does not impose a particular presentation
convey the type of format to be used. However, the presentation description or metafile format and can convey the type of
description must contain at least one RTSP URI. format to be used. However, the presentation description must
contain at least one RTSP URI.
Proxy and firewall friendly: The protocol should be readily handled Proxy and firewall friendly:
by both application and transport-layer (SOCKS [13]) firewalls. The protocol should be readily handled by both application and
A firewall may need to understand the SETUP method to open a transport-layer (SOCKS [13]) firewalls. A firewall may need to
"hole" for the UDP media stream. understand the SETUP method to open a ``hole'' for the UDP
media stream.
HTTP-friendly: Where sensible, RTSP re-uses HTTP concepts, so that HTTP-friendly:
the existing infrastructure can be re-used. This infrastructure Where sensible, RTSP re-uses HTTP concepts, so that the
includes JEPI (the Joint Electronic Payment Initiative) for existing infrastructure can be re-used. This infrastructure
electronic payments and PICS (Platform for Internet Content includes PICS (Platform for Internet Content Selection [20])
Selection) for associating labels with content. However, RTSP for associating labels with content. However, RTSP does not
does not just add methods to HTTP, since the controlling just add methods to HTTP, since the controlling continuous
continuous media requires server state in most cases. media requires server state in most cases.
Appropriate server control: If a client can start a stream, it must Appropriate server control:
be able to stop a stream. Servers should not start streaming to If a client can start a stream, it must be able to stop a
clients in such a way that clients cannot stop the stream. stream. Servers should not start streaming to clients in such a
way that clients cannot stop the stream.
Transport negotiation: The client can negotiate the transport method H. Schulzrinne, A. Rao, R. Lanphier Page 9
prior to actually needing to process a continuous media stream. Transport negotiation:
The client can negotiate the transport method prior to actually
needing to process a continuous media stream.
Capability negotiation: If basic features are disabled, there must be Capability negotiation:
some clean mechanism for the client to determine which methods If basic features are disabled, there must be some clean
are not going to be implemented. This allows clients to present mechanism for the client to determine which methods are not
the appropriate user interface. For example, if seeking is not going to be implemented. This allows clients to present the
appropriate user interface. For example, if seeking is not
allowed, the user interface must be able to disallow moving a 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 An earlier requirement in RTSP was multi-client capability.
capability. However, it was determined that a better However, it was determined that a better approach was to make sure
approach was to make sure that the protocol is easily that the protocol is easily extensible to the multi-client
extensible to the multi-client scenario. Stream identifiers scenario. Stream identifiers can be used by several control
can be used by several control streams, so that "passing streams, so that ``passing the remote'' would be possible. The
the remote" would be possible. The protocol would not protocol would not address how several clients negotiate access;
address how several clients negotiate access; this is left this is left to either a ``social protocol'' or some other floor
to either a "social protocol" or some other floor control control mechanism.
mechanism.
1.5 Extending RTSP 1.5 Extending RTSP
Since not all media servers have the same functionality, media Since not all media servers have the same functionality, media servers
servers by necessity will support different sets of requests. For by necessity will support different sets of requests. For example:
example: * A server may only be capable of playback, not recording and thus
has no need to support the RECORD request.
o A server may only be capable of playback, not recording and * A server may not be capable of seeking (absolute positioning),
thus has no need to support the RECORD request.
o A server may not be capable of seeking (absolute positioning),
say, if it is to support live events only. say, if it is to support live events only.
* Some servers may not support setting stream parameters and thus
not support GET_PARAMETER and SET_PARAMETER.
o Some servers may not support setting stream parameters and A server SHOULD implement all header fields described in Section 12.
thus not support GET_PARAMETER and SET_PARAMETER.
A server SHOULD implement all header fields described in Section 11.
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, where impossible of a server. This situation is similar in HTTP/1.1, where
the methods described in [H19.6] are not likely to be supported the methods described in [H19.6] are not likely to be supported across
across all servers. all servers.
RTSP can be extended in three ways, listed in order of the magnitude RTSP can be extended in three ways, listed in order of the magnitude
of changes supported: of changes supported:
o Existing methods can be extended with new parameters, as long * Existing methods can be extended with new parameters, as long as
as these parameters can be safely ignored by the recipient. these parameters can be safely ignored by the recipient. (This is
(This is equivalent to adding new parameters to an HTML tag.) equivalent to adding new parameters to an HTML tag.)
* New methods can be added. If the recipient of the message does not
o 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 implemented) and the sender should not attempt to use this method
(Not implemented) and the sender can then attempt an earlier, again. A client may also use the OPTIONS method to inquire about
less functional version. methods supported by the server. The server SHOULD list the
methods it supports using the Public response header.
o A new version of the protocol can be defined, allowing almost * A new version of the protocol can be defined, allowing almost all
all aspects (except the position of the protocol version aspects (except the position of the protocol version number) to
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 HTTP or other means such as email and may not necessarily be stored on
on the media server. 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 assumed to describe one or more presentations, each of which maintains
maintains a common time axis. For simplicity of exposition and a common time axis. For simplicity of exposition and without loss of
without loss of generality, it is assumed that the presentation generality, it is assumed that the presentation description contains
description contains exactly one such presentation. A presentation exactly one such presentation. A presentation may contain several
may contain several media streams. 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 Besides the media parameters, the network destination address and port
port need to be determined. Several modes of operation can be need to be determined. Several modes of operation can be
distinguished: distinguished:
Unicast: The media is transmitted to the source of the RTSP request, Unicast:
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: The media server picks the Multicast, server chooses address:
multicast address and port. This is the typical case for a live The media server picks the multicast address and port. This is
or near-media-on-demand transmission. the typical case for a live or near-media-on-demand
transmission.
Multicast, client chooses address: If the server is to participate in Multicast, client chooses address:
an existing multicast conference, the multicast address, port If the server is to participate in an existing multicast
and encryption key are given by the conference description, conference, the multicast address, port and encryption key are
established by means outside the scope of this specification. given by the conference description, established by means
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: Causes the server to allocate resources for a stream and start SETUP:
Causes the server to allocate resources for a stream and start
an RTSP session. an RTSP session.
PLAY and RECORD: Starts data transmission on a stream allocated via PLAY and RECORD:
SETUP. Starts data transmission on a stream allocated via SETUP.
PAUSE: Temporarily halts a stream, without freeing server resources. PAUSE:
Temporarily halts a stream, without freeing server resources.
TEARDOWN: Frees resources associated with the stream. The RTSP TEARDOWN:
session ceases to exist on the server. Frees resources associated with the stream. The RTSP session
ceases to exist on the server.
1.8 Relationship with Other Protocols 1.8 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may RTSP has some overlap in functionality with HTTP. It also may interact
interact with HTTP in that the initial contact with streaming content with HTTP in that the initial contact with streaming content is often
is often to be made through a web page. The current protocol to be made through a web page. The current protocol specification aims
specification aims to allow different hand-off points between a web to allow different hand-off points between a web server and the media
server and the media server implementing RTSP. For example, the server implementing RTSP. For example, the presentation description
presentation description can be retrieved using HTTP or RTSP. Having can be retrieved using HTTP or RTSP. Having the presentation
the presentation description be returned by the web server makes it description be returned by the web server makes it possible to have
possible to have the web server take care of authentication and the web server take care of authentication and billing, by handing out
billing, by handing out a presentation description whose media a presentation description whose media identifier includes an
identifier includes an encrypted version of the requestor's IP encrypted version of the requestor's IP address and a timestamp, with
address and a timestamp, with a shared secret between web and media a shared secret between web and media server.
server.
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 takes place out-of-band, in a different protocol. HTTP is an
asymmetric protocol, where the client issues requests and the server asymmetric protocol, where the client issues requests and the server
responds. In RTSP, both the media client and media server can issue responds. In RTSP, both the media client and media server can issue
requests. RTSP requests are also not stateless, in that they may set requests. RTSP requests are also not stateless, in that they may set
parameters and continue to control a media stream long after the parameters and continue to control a media stream long after the
request has been acknowledged. request has been acknowledged.
Re-using HTTP functionality has advantages in at least two Re-using HTTP functionality has advantages in at least two areas,
areas, namely security and proxies. The requirements are namely security and proxies. The requirements are very similar, so
very similar, so having the ability to adopt HTTP work on having the ability to adopt HTTP work on caches, proxies and
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 rather than copying it. For brevity, [HX.Y] is to be taken to refer to
to Section X.Y of the current HTTP/1.1 specification (RFC 2068). Section X.Y of the current HTTP/1.1 specification (RFC 2068).
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
RFC 2068 [H2.1]. It is described in detail in [14]. RFC 2068 [H2.1]. It is described in detail in [14].
In this draft, we use indented and smaller-type paragraphs to provide In this draft, we use indented and smaller-type paragraphs to provide
background and motivation. Some of these paragraphs are marked with background and motivation. Some of these paragraphs are marked with
HS, AR and RL, designating opinions and comments by the individual HS, AR and RL, designating opinions and comments by the individual
authors which may not be shared by the co-authors and require authors which may not be shared by the co-authors and require
resolution. resolution.
3 Protocol Parameters 3 Protocol Parameters
3.1 RTSP Version 3.1 RTSP Version
applies, with HTTP replaced by RTSP. [H3.1] applies, with HTTP replaced by RTSP.
3.2 RTSP URL 3.2 RTSP URL
The "rtsp" and "rtspu" schemes are used to refer to network resources The ``rtsp'', ``rtspu'' and ``rtsps'' schemes are used to refer to
via the RTSP protocol. This section defines the scheme-specific network resources via the RTSP protocol. This section defines the
syntax and semantics for RTSP URLs. scheme-specific syntax and semantics for RTSP URLs.
rtsp_URL = ( "rtsp:" | "rtspu:" ) "//" host [ ":" port ] [abs_path] rtsp_URL = ( "rtsp:" | "rtspu:" | "rtsps:" )
"//" 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> of RFC 1123>
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 Note that fragment and query identifiers do not have a well-defined
well-defined meaning at this time, with the interpretation meaning at this time, with the interpretation left to the RTSP
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 protocol (within the Internet, TCP), while the scheme rtspu identifies
identifies an unreliable protocol (within the Internet, UDP). an unreliable protocol (within the Internet, UDP). The scheme rtsps
indicates that a TCP connection secured by TLS [7] must be used.
If the port is empty or not given, port 554 is assumed. The semantics If the port is empty or not given, port 554 is assumed. The semantics
are that the identified resource can be controlled be RTSP at the are that the identified resource can be controlled be 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 [15]). (see RFC 1924 [15]).
A presentation or a stream is identified by an textual media A presentation or a stream is identified by an 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 [16]. Requests described in Section 9 can refer to either the URLs [16]. URLs may refer to a stream or an aggregate of streams ie. a
whole presentation or an individual stream within the presentation. presentation. Accordingly, requests described in Section 10 can apply
Note that some methods can only be applied to streams, not to either the whole presentation or an individual stream within the
presentations and vice versa. A specific instance of a presentation presentation. Note that some request methods can only be applied to
or stream, e.g., one of several concurrent transmissions of the same streams, not presentations and vice versa.
content, an RTSP session , is indicated by the Session header field
(Section 11.26) where needed.
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.
This does not imply a standard way to reference streams in Also, the RTSP URL
URLs. The presentation description defines the hierarchical rtsp://media.example.com:554/twister
relationships in the presentation and the URLs for the
individual streams. A presentation description may name a identifies the presentation ``twister'', which may be composed of
stream 'a.mov' and the whole presentation 'b.mov'. audio and video streams.
This does not imply a standard way to reference streams in URLs.
The presentation description defines the hierarchical relationships
in the presentation and the URLs for the individual streams. A
presentation description may name a stream 'a.mov' and the whole
presentation 'b.mov'.
The path components of the RTSP URL are opaque to the client and do 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 This decoupling also allows presentation descriptions to be used
used with non-RTSP media control protocols, simply by with non-RTSP media control protocols, simply by replacing the
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*OCTET ; LWS must be URL-escaped conference-id = 1*OCTET ; LWS must be URL-escaped
Conference identifiers are used to allow to allow RTSP Conference identifiers are used to allow to allow RTSP sessions to
sessions to obtain parameters from multimedia conferences obtain parameters from multimedia conferences the media server is
the media server is participating in. These conferences are participating in. These conferences are created by protocols
created by protocols outside the scope of this outside the scope of this specification, e.g., H.323 [17] or SIP
specification, e.g., H.323 [17] or SIP [12]. Instead of the [12]. Instead of the RTSP client explicitly providing transport
RTSP client explicitly providing transport information, for information, for example, it asks the media server to use the
example, it asks the media server to use the values in the values in the conference description instead. If the conference
conference description instead. If the conference
participant inviting the media server would only supply a participant inviting the media server would only supply a
conference identifier which is unique for that inviting conference identifier which is unique for that inviting party, the
party, the media server could add an internal identifier media server could add an internal identifier for that party, e.g.,
for that party, e.g., its Internet address. However, this its Internet address. However, this would prevent that the
would prevent that the conference participant and the conference participant and the initiator of the RTSP commands are
initiator of the RTSP commands are two different entities. two different entities.
3.4 SMPTE Relative Timestamps 3.4 Session Identifiers
Session identifiers are opaque strings of arbitrary length. Linear
white space must be URL-escaped. A session identifier SHOULD be chosen
randomly and SHOULD be at least eight octets long to make guessing it
more difficult. (See Section 16).
session-id = 1*OCTET ; LWS must be URL-escaped
3.5 SMPTE Relative Timestamps
A SMPTE relative time-stamp expresses time relative to the start of A SMPTE relative time-stamp 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 hours:minutes:seconds:frames.subframes, with the origin at the start
, of the clip. For NTSC, the frame rate is 29.97 frames per second. This
with the origin at the start of the clip. For NTSC, the frame rate is is handled by dropping the first two frame indices (values 00 and 01)
29.97 frames per second. This is handled by dropping the first frame of every minute, except every tenth minute. If the frame value is
index of every minute, except every tenth minute. If the frame value zero, it may be omitted. Subframes are measured in one-hundredth of a
is zero, it may be omitted. frame.
smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ] smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ]
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ "." 1*2DIGIT ] smpte-time = 2DIGIT ":" 2DIGIT ":" 2DIGIT [ ":" 2DIGIT ] [ "." 2DIGIT]
Examples: Examples:
smpte=10:12:33:20-
smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01
smpte=10:12:33.40- 3.6 Normal Play Time
smpte=10:7:33-
smpte=10:7:0-10:7:33
3.5 Normal Play Time
Normal play time (NPT) indicates the stream absolute position Normal play time (NPT) indicates the stream absolute position relative
relative to the beginning of the presentation, measured in seconds to the beginning of the presentation, measured in seconds and
and microseconds. The beginning of a presentation corresponds to 0 microseconds. The beginning of a presentation corresponds to 0 seconds
seconds and 0 microseconds. Negative values are not defined. The and 0 microseconds. Negative values are not defined. The microsecond
microsecond field is always less than 1,000,000. NPT is defined as in field is always less than 1,000,000. NPT is defined as in DSM-CC [18]:
DSM-CC: "Intuitively, NPT is the clock the viewer associates with a ``Intuitively, NPT is the clock the viewer associates with a program.
program. It is often digitally displayed on a VCR. NPT advances It is often digitally displayed on a VCR. NPT advances normally when
normally when in normal play mode (scale = 1), advances at a faster in normal play mode (scale = 1), advances at a faster rate when in
rate when in fast scan forward (high positive scale ratio), fast scan forward (high positive scale ratio), decrements when in scan
decrements when in scan reverse (high negative scale ratio) and is reverse (high negative scale ratio) and is fixed in pause mode. NPT is
fixed in pause mode. NPT is [logically] equivalent to SMPTE time (logically) equivalent to SMPTE time codes.'' [18]
codes." [18]
npt-range = "npt" "=" npt-time "-" [ npt-time ] npt-range = "npt" "=" npt-time "-" [ npt-time ]
npt-time = 1*DIGIT [ ":" *DIGIT ] npt-time = 1*DIGIT [ ":" *DIGIT ]
Examples: Examples:
npt=123:45-125 npt=123:45-125
3.6 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
Example Example
4 RTSP Message 4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2044). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by
themselves as line terminators.
Text-based protocols make it easier to add optional RTSP is a text-based protocol and uses the ISO 10646 character set
parameters in a self-describing manner. Since the number of in UTF-8 encoding (RFC 2044). Lines are terminated by CRLF, but
parameters and the frequency of commands is low, processing receivers should be prepared to also interpret CR and LF by themselves
efficiency is not a concern. Text-based protocols, if done as line terminators.
carefully, also allow easy implementation of research
prototypes in scripting languages such as Tcl, Visual Basic
and Perl.
The 10646 character set avoids tricky character set switching, but is Text-based protocols make it easier to add optional parameters in a
invisible to the application as long as US-ASCII is being used. This self-describing manner. Since the number of parameters and the
is also the encoding used for RTCP. ISO 8859-1 translates directly frequency of commands is low, processing efficiency is not a
into Unicode, with a high-order octet of zero. ISO 8859-1 characters concern. Text-based protocols, if done carefully, also allow easy
with the most-significant bit set are represented as 1100001x implementation of research prototypes in scripting languages such
10xxxxxx. as Tcl, Visual Basic and Perl.
The 10646 character set avoids tricky character set switching, but
is invisible to the application as long as US-ASCII is being used.
This is also the encoding used for RTCP. ISO 8859-1 translates
directly into Unicode, with a high-order octet of zero. ISO 8859-1
characters with the most-significant bit set are represented as
1100001x 10xxxxxx.
RTSP messages can be carried over any lower-layer transport protocol 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 unless otherwise noted. Methods are also designed to require little or
or no state maintenance at the media server. 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
See [H4.3] See [H4.3]
4.4 Message Length 4.4 Message Length
When a message-body is included with a message, the length of that When a message-body is included with a message, the length of that
body is determined by one of the following (in order of precedence): body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message-body 1.
(such as the 1xx, 204, and 304 responses) is always Any response message which MUST NOT include a message-body
terminated by the first empty line after the header fields, (such as the 1xx, 204, and 304 responses) is always terminated
regardless of the entity-header fields present in the by the first empty line after the header fields, regardless of
message. (Note: An empty line consists of only CRLF.) the entity-header fields present in the message. (Note: An
empty line consists of only CRLF.)
2.
If a Content-Length header field (section 12.15) is present,
its value in bytes represents the length of the message-body.
If this header field is not present, a value of zero is
assumed.
3.
By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a
response.)
2. If a Content-Length header field (section 11.12) is Note that RTSP does not (at present) support the HTTP/1.1 ``chunked''
present, its value in bytes represents the length of the transfer coding(see [H3.6]) and requires the presence of the
message-body. If this header field is not present, a value Content-Length header field.
of zero is assumed.
3. By the server closing the connection. (Closing the Given the moderate length of presentation descriptions returned,
connection cannot be used to indicate the end of a request the server should always be able to determine its length, even if
body, since that would leave no possibility for the server it is generated dynamically, making the chunked transfer encoding
to send back a response.) unnecessary. Even though Content-Length must be present if there is
any entity body, the rules ensure reasonable behavior even if the
length is not given explicitly.
Note that RTSP does not (at present) support the HTTP/1.1 "chunked" 5 General Header Fields
transfer coding and requires the presence of the Content-Length
header field.
Given the moderate length of presentation descriptions See [H4.5], except that Pragma, Transfer-Encoding and Upgrade
returned, the server should always be able to determine its headers are not defined:
length, even if it is generated dynamically, making the
chunked transfer encoding unnecessary. Even though
Content-Length must be present if there is any entity body,
the rules ensure reasonable behavior even if the length is
not given explicitly.
5 Request general-header = Cache-Control ; Section 12.10
| Connection ; Section 12.12
| Date ; Section 12.17
| Via ; Section 12.39
6 Request
A request message from a client to a server or vice versa includes, 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 within the first line of that message, the method to be applied to the
the resource, the identifier of the resource, and the protocol resource, the identifier of the resource, and the protocol version in
version in use. use.
Request = Request-line CRLF Request = Request-Line ; Section 6.1
*request-header *( general-header ; Section 5
| request-header ; Section 6.2
| entity-header ) ; Section 8.1
CRLF CRLF
[ message-body ] [ message-body ] ; Section 4.3
6.1 Request Line
Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no CRLF Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no CRLF
Method = "DESCRIBE" ; Section Method = "DESCRIBE" ; Section 10.2
| "GET_PARAMETER" ; Section | "ANNOUNCE" ; Section 10.3
| "OPTIONS" ; Section | "GET_PARAMETER" ; Section 10.8
| "PAUSE" ; Section | "OPTIONS" ; Section 10.1
| "PLAY" ; Section | "PAUSE" ; Section 10.6
| "RECORD" ; Section | "PLAY" ; Section 10.5
| "REDIRECT" ; Section | "RECORD" ; Section 10.11
| "SETUP" ; Section | "REDIRECT" ; Section 10.10
| "SET_PARAMETER" ; Section | "SETUP" ; Section 10.4
| "TEARDOWN" ; Section | "SET_PARAMETER" ; Section 10.9
| "TEARDOWN" ; Section 10.7
| extension-method | extension-method
extension-method = token extension-method = token
Request-URI = "*" | absolute_URI Request-URI = "*" | absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
seq-no = 1*DIGIT seq-no = 1*DIGIT
6.2 Request Header Fields
request-header = Accept ; Section 12.1
| Accept-Encoding ; Section 12.2
| Accept-Language ; Section 12.3
| Authorization ; Section 12.5
| From ; Section 12.19
| If-Modified-Since ; Section 12.21
| Range ; Section 12.28
| Referer ; Section 12.29
| User-Agent ; Section 12.37
Note that in contrast to HTTP/1.1, RTSP requests always contain the Note that in contrast to HTTP/1.1, RTSP requests always contain the
absolute URL (that is, including the scheme, host and port) rather absolute URL (that is, including the scheme, host and port) rather
than just the absolute path. than just the absolute path.
HTTP/1.1 requires servers to understand the absolute URL, but
clients are supposed to use the Host request header. This is purely
needed for backward-compatibility with HTTP/1.0 servers, a
consideration that does not apply to RTSP.
The asterisk "*" in the Request-URI means that the request does not 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 allowed when the method used does not necessarily apply to a resource.
resource. One example would be One example would be
OPTIONS * RTSP/1.0 OPTIONS * RTSP/1.0
6 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.
define some HTTP codes. The valid response codes and the methods they Also, RTSP defines additional status codes and does not define some
can be used with are defined in the table 1. HTTP codes. The valid response codes and the methods they can be used
with are defined in the table 1.
After receiving and interpreting a request message, the recipient 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 Response = Status-Line ; Section 7.1
*( general-header ; Section *( general-header ; Section 5
| response-header ; Section | response-header ; Section 7.1.2
| entity-header ) ; Section | entity-header ) ; Section 8.1
CRLF CRLF
[ message-body ] ; Section [ message-body ] ; Section 4.3
6.1 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, the of the protocol version followed by a numeric status code, the
sequence number of the corresponding request and the textual phrase sequence number of the corresponding request and the textual phrase
associated with the status code, with each element separated by SP associated with the status code, with each element separated by SP
characters. No CR or LF is allowed except in the final CRLF sequence. characters. No CR or LF is allowed except in the final CRLF sequence.
Note that the addition of a
Status-Line = RTSP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF Status-Line = RTSP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF
6.1.1 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 section10. The Reason-Phrase is intended to give a short defined in section11. 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 Reason- user. The client is not required to examine or display the
Phrase Reason-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:
o 1xx: Informational - Request received, continuing process * 1xx: Informational - Request received, continuing process
* 2xx: Success - The action was successfully received, understood,
o 2xx: Success - The action was successfully received, and accepted
understood, and accepted * 3xx: Redirection - Further action must be taken in order to
o 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
o 4xx: Client Error - The request contains bad syntax or cannot fulfilled
be fulfilled * 5xx: Server Error - The server failed to fulfill an apparently
o 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 below. RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
The reason phrases listed here are only recommended -- they may be presented below. The reason phrases listed here are only recommended -
replaced by local equivalents without affecting the protocol. Note they may be replaced by local equivalents without affecting the
that RTSP adopts most HTTP/1.1 status codes and adds RTSP-specific protocol. Note that RTSP adopts most HTTP/1.1 status codes and adds
status codes in the starting at 450 to avoid conflicts with newly RTSP-specific status codes in the starting at 450 to avoid conflicts
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
| "300" ; Multiple Choices | "300" ; Multiple Choices
| "301" ; Moved Permanently | "301" ; Moved Permanently
| "302" ; Moved Temporarily | "302" ; Moved Temporarily
| "303" ; See Other | "303" ; See Other
| "304" ; Not Modified | "304" ; Not Modified
| "305" ; Use Proxy | "305" ; Use Proxy
skipping to change at page 18, line 39 skipping to change at line 1007
| "414" ; Request-URI Too Large | "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type | "415" ; Unsupported Media Type
| "451" ; Parameter Not Understood | "451" ; Parameter Not Understood
| "452" ; Conference Not Found | "452" ; Conference Not Found
| "453" ; Not Enough Bandwidth | "453" ; Not Enough Bandwidth
| "45x" ; Session Not Found | "45x" ; Session Not Found
| "45x" ; Method Not Valid in This State | "45x" ; Method Not Valid in This State
| "45x" ; Header Field Not Valid for Resource | "45x" ; Header Field Not Valid for Resource
| "45x" ; Invalid Range | "45x" ; Invalid Range
| "45x" ; Parameter Is Read-Only | "45x" ; Parameter Is Read-Only
| "45x" ; Aggregate operation not allowed
| "45x" ; Only aggregate operation allowed
| "500" ; Internal Server Error | "500" ; Internal Server Error
| "501" ; Not Implemented | "501" ; Not Implemented
| "502" ; Bad Gateway | "502" ; Bad Gateway
| "503" ; Service Unavailable | "503" ; Service Unavailable
| "504" ; Gateway Time-out | "504" ; Gateway Time-out
| "505" ; HTTP Version not supported | "505" ; RTSP Version not supported
| extension-code | extension-code
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 x00 status code of that class, with the exception that an unrecognized
unrecognized response MUST NOT be cached. For example, if an response MUST NOT be cached. For example, if an unrecognized status
unrecognized status code of 431 is received by the client, it can code of 431 is received by the client, it can safely assume that there
safely assume that there was something wrong with its request and was something wrong with its request and treat the response as if it
treat the response as if it had received a 400 status code. In such had received a 400 status code. In such cases, user agents SHOULD
cases, user agents SHOULD present to the user the entity returned present to the user the entity returned with the response, since that
with the response, since that entity is likely to include human- entity is likely to include human-readable information which will
readable information which will explain the unusual status. explain the unusual status.
6.1.2 Response Header Fields
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
the Status-Line server and about further access to the resource
identified by the Request-URI
response-header = Location ; Section
| Proxy-Authenticate ; Section
| Public ; Section
| Retry-After ; Section
| Server ; Section
| Vary ; Section
| WWW-Authenticate ; Section
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as
entity-header fields.
7 Entity
Request and Response messages MAY transfer an entity if not otherwise
restricted by the request method or response status code. An entity
consists of entity-header fields and an entity-body, although some
responses will only include the entity-headers.
In this section, both sender and recipient refer to either the client Code Reason
Code reason
_______________________________________________________________
100 Continue all 100 Continue all
_______________________________________________________________
200 OK all 200 OK all
201 Created RECORD 201 Created RECORD
_______________________________________________________________
300 Multiple Choices all 300 Multiple Choices all
301 Moved Permanently all 301 Moved Permanently all
302 Moved Temporarily all 302 Moved Temporarily all
303 See Other all 303 See Other all
305 Use Proxy all 305 Use Proxy all
_______________________________________________________________
400 Bad Request all 400 Bad Request all
401 Unauthorized all 401 Unauthorized all
402 Payment Required all 402 Payment Required all
403 Forbidden all 403 Forbidden all
404 Not Found all 404 Not Found all
405 Method Not Allowed all 405 Method Not Allowed all
406 Not Acceptable all 406 Not Acceptable all
407 Proxy Authentication Required all 407 Proxy Authentication Required all
408 Request Timeout all 408 Request Timeout all
409 Conflict 409 Conflict RECORD
410 Gone all 410 Gone all
411 Length Required SETUP 411 Length Required SETUP
412 Precondition Failed 412 Precondition Failed DESCRIBE, SETUP
413 Request Entity Too Large SETUP 413 Request Entity Too Large SETUP
414 Request-URI Too Long all 414 Request-URI Too Long all
415 Unsupported Media Type SETUP 415 Unsupported Media Type SETUP
45x Only Valid for Stream SETUP 45x Session not found all
45x Invalid parameter SETUP 45x Invalid parameter SETUP
45x Not Enough Bandwidth SETUP 45x Not Enough Bandwidth SETUP
45x Illegal Conference Identifier SETUP 45x Illegal Conference Identifier SETUP
45x Illegal Session Identifier PLAY, RECORD, TEARDOWN 45x Illegal Session Identifier PLAY, RECORD, TEARDOWN
45x Parameter Is Read-Only SET_PARAMETER 45x Parameter Is Read-Only SET_PARAMETER
45x Header Field Not Valid all 45x Header Field Not Valid all
_______________________________________________________________ 45x Method Not Valid In This State all
45x Aggregate operation not allowed all
45x Only aggregate operation allowed all
500 Internal Server Error all 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
!
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
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the server
and about further access to the resource identified by the
Request-URI.
response-header = Location ; Section 12.23
| Proxy-Authenticate ; Section 12.26
| Public ; Section 12.27
| Retry-After ; Section 12.30
| Server ; Section 12.33
| Vary ; Section 12.38
| WWW-Authenticate ; Section 12.40
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of
response-header fields if all parties in the communication recognize
them to be response-header fields. Unrecognized header fields are
treated as entity-header fields.
8 Entity
Request and Response messages MAY transfer an entity if not
otherwise restricted by the request method or response status code. An
entity consists of entity-header fields and an entity-body, although
some responses will only include the entity-headers.
In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity. or the server, depending on who sends and who receives the entity.
7.1 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 14.7 entity-header = Allow ; Section 12.4
| Content-Encoding ; Section 14.12 | Content-Encoding ; Section 12.13
| Content-Language ; Section 14.13 | Content-Language ; Section 12.14
| Content-Length ; Section 14.14 | Content-Length ; Section 12.15
| Content-Type ; Section 14.18 | Content-Type ; Section 12.16
| Expires ; Section 14.21 | Expires ; Section 12.18
| Last-Modified ; Section 14.29 | Last-Modified ; Section 12.22
| extension-header | extension-header
extension-header = message-header extension-header = message-header
The extension-header mechanism allows additional entity-header fields The extension-header mechanism allows additional entity-header fields
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.
7.2 Entity Body 8.2 Entity Body
See [H7.2] See [H7.2]
8 Connections 9 Connections
RTSP requests can be transmitted in several different ways: RTSP requests can be transmitted in several different ways:
o persistent transport connections used for several request- * persistent transport connections used for several request-response
response transactions; transactions;
* one connection per request/response transaction;
o one connection per request/response transaction; * connectionless mode.
o connectionless mode.
The type of transport connection is defined by the RTSP URI (Section The type of transport connection is defined by the RTSP URI
3.2). For the scheme "rtsp", a persistent connection is assumed, (Section 3.2). For the scheme ``rtsp'', a persistent connection is
while the scheme "rtspu" calls for RTSP requests to be send without assumed, while the scheme ``rtspu'' calls for RTSP requests to be send
setting up a connection. without 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.
8.1 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.
8.2 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 multicast group. If there is no acknowledgement, the sender may resend
resend the same message after a timeout of one round-trip time (RTT). the same message after a timeout of one round-trip time (RTT). The
The round-trip time is estimated as in TCP (RFC TBD), with an initial round-trip time is estimated as in TCP (RFC TBD), with an initial
round-trip value of 500 ms. An implementation MAY cache the last RTT round-trip value of 500 ms. An implementation MAY cache the last RTT
measurement as the initial value for future connections. If a measurement as the initial value for future connections. If a reliable
reliable transport protocol is used to carry RTSP, the timeout value transport protocol is used to carry RTSP, the timeout value MAY be set
MAY be set to an arbitrarily large value. to an arbitrarily large value.
This can greatly increase responsiveness for proxies This can greatly increase responsiveness for proxies operating in
operating in local-area networks with small RTTs. The local-area networks with small RTTs. The mechanism is defined such
mechanism is defined such that the client implementation that the client implementation does not have be aware of whether a
does not have be aware of whether a reliable or unreliable reliable or unreliable transport protocol is being used. It is
transport protocol is being used. It is probably a bad idea probably a bad idea to have two reliability mechanisms on top of
to have two reliability mechanisms on top of each other, each other, although the RTSP RTT estimate is likely to be larger
although the RTSP RTT estimate is likely to be larger than than the TCP estimate.
the TCP estimate.
Each request carries a sequence number, which is incremented by one Each request carries a sequence number, which is incremented by one
for each request transmitted. If a request is repeated because of for each request transmitted. If a request is repeated because of lack
lack of acknowledgement, the sequence number is incremented. of acknowledgement, the sequence number is incremented.
This avoids ambiguities when computing round-trip time This avoids ambiguities when computing round-trip time estimates.
estimates. [TBD: An initial sequence number negotiation
needs to be added for UDP; otherwise, a new stream [TBD: An initial sequence number negotiation needs to be added for
connection may see a request be acknowledged by a delayed UDP; otherwise, a new stream connection may see a request be
response from an earlier "connection". This handshake can acknowledged by a delayed response from an earlier ``connection''.
be avoided with a sequence number containing a timestamp of This handshake can be avoided with a sequence number containing a
sufficiently high resolution.] timestamp of sufficiently high resolution.]
The reliability mechanism described here does not protect against The reliability mechanism described here does not protect against
reordering. This may cause problems in some instances. For example, a reordering. This may cause problems in some instances. For example, a
TEARDOWN followed by a PLAY has quite a different effect than the TEARDOWN followed by a PLAY has quite a different effect than the
reverse. Similarly, if a PLAY request arrives before all parameters reverse. Similarly, if a PLAY request arrives before all parameters
are set due to reordering, the media server would have to issue an are set due to reordering, the media server would have to issue an
error indication. Since sequence numbers for retransmissions are error indication. Since sequence numbers for retransmissions are
incremented (to allow easy RTT estimation), the receiver cannot just incremented (to allow easy RTT estimation), the receiver cannot just
ignore out-of-order packets. [TBD: This problem could be fixed by ignore out-of-order packets. [TBD: This problem could be fixed by
including both a sequence number that stays the same for including both a sequence number that stays the same for
retransmissions and a timestamp for RTT estimation.] retransmissions and a timestamp for RTT estimation.]
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. stream. RTSP data MAY be interleaved with RTP and RTCP packets. Unlike
Unlike HTTP, an RTSP method header MUST contain a Content-Length HTTP, an RTSP message MUST contain a Content-Length header whenever
whenever that method contains a payload. Otherwise, an RTSP packet is that message contains a payload. Otherwise, an RTSP packet is
terminated with an empty line immediately following the method terminated with an empty line immediately following the last message
header. header.
9 Method Definitions 10 Method Definitions
The method token indicates the method to be performed on the resource The method token indicates the method to be performed on the
identified by the Request-URI case-sensitive. New methods may be resource identified by the Request-URI. The method is case-sensitive.
defined in the future. Method names may not start with a $ character New methods may be defined in the future. Method names may not start
(decimal 24) and must be a token with a $ character (decimal 24) and must be a token. Methods are
summarized in Table 2.
method direction object requirement method direction object requirement
________________________________________________________ DESCRIBE C->S P,S recommended
DESCRIBE C -> S, S -> C P,S recommended 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 P,S required OPTIONS C -> S P,S required
PAUSE C -> S P,S recommended PAUSE C -> S P,S recommended
PLAY C -> S P,S required PLAY C -> S P,S required
RECORD C -> S P,S optional RECORD C -> S P,S optional
REDIRECT S -> C P,S optional REDIRECT S -> C P,S optional
SETUP C -> S S required SETUP C -> S S required
SET_PARAMETER C -> S, S -> C P,S optional SET_PARAMETER C -> S, S -> C P,S optional
TEARDOWN C -> S P,S required TEARDOWN C -> S P,S required
!
Table 2: Overview of RTSP methods, their direction, and what objects (P:
presentation, S: stream) they operate on
Table 2: Overview of RTSP methods, their direction, and what objects Notes on Table 2: PAUSE is recommended, but not required in that a
(P: presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommend, 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.
9.1 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 request may be issued at any time, e.g., if the client is about to try
try a non-standard request. It does not influence server state. a non-standard request. It does not influence server state.
In addition, if the optional Require header is present, option tags
within the header indicate features needed by the requestor that are
not required at the version level of the protocol.
Example 1:
Example :
C->S: OPTIONS * RTSP/1.0 1 C->S: OPTIONS * RTSP/1.0 1
Require: implicit-play, record-feature PEP: {{map "http://www.iana.org/rtsp/implicit-play"}}
Transport-Require: switch-to-udp-control, gzipped-messages {{map "http://www.iana.org/rtsp/record-feature"}}
C-PEP: {{map "http://www.iana.org/rtsp/udp-control"}}
Note that these are fictional features (though we may want to make {{map "http://www.iana.org/rtsp/gzipped-messages"}}
them real one day).
Example 2 (using RFC2069-style authentication only as an example):
S->C: OPTIONS * RTSP/1.0 1
Authenticate: Digest realm="testrealm@host.com",
nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
opaque="5ccc069c403ebaf9f0171e9517f40e41"
S->C: RTSP/1.0 200 1 OK S->C: RTSP/1.0 200 2 OK
Date: 23 Jan 1997 15:35:06 GMT PEP-Info: {{map "http://www.iana.org/rtsp/implicit-play"}
Nack-Transport-Require: switch-to-udp-control {for "/" *}}
{{map "http://www.iana.org/rtsp/record-feature"}
{for "/" *}}
C-PEP-Info: {{map "http://www.iana.org/rtsp/udp-control"}
{for "/" *}}
{{map "http://www.iana.org/rtsp/gzipped-messages"}
{for "/" *}}
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are fictional features (though we may want to make Note that these are fictional features (though we may want to make
them real one day). them real one day).
Example 2 (using RFC2069-style authentication only as an example): DESCRIBE
C->S: RTSP/1.0 401 1 Unauthorized
Authorization: Digest username="Mufasa",
realm="testrealm@host.com",
nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
uri="/dir/index.html",
response="e966c932a9242554e42c8ee200cec7f6",
opaque="5ccc069c403ebaf9f0171e9517f40e41"
9.2 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. Alternatively, the server may "push" a new description to resource.
the client, for example, if a new stream has become available. If a
new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent
again, rather than just the additional components, so that components
can be deleted.
Example: Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 312 C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 312
Accept: application/sdp, application/rtsl, application/mheg Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 312 OK S->C: RTSP/1.0 200 312 OK
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 376 Content-Length: 376
skipping to change at page 25, line 44 skipping to change at line 1304
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
m=whiteboard 32416 UDP WB m=whiteboard 32416 UDP WB
a=orient:portrait a=orient:portrait
or ANNOUNCE
S->C: RTSP/1.0 200 312 OK The ANNOUNCE method serves two purposes:
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/rtsl
Content-Length: 2782
<2782 octets of data containing stream description> When sent from client to server, ANNOUNCE posts the description of a
presentation or media object identified by the request URL to a
server.
Server to client example: When sent from server to client, ANNOUNCE updates the session
description in real-time.
S->C: DESCRIBE /twister RTSP/1.0 902 If a new media stream is added to a presentation (e.g., during a live
Session: 1234 presentation), the whole presentation description should be sent
Content-Type: application/rtsl again, rather than just the additional components, so that components
can be deleted.
new RTSL presentation description Example:
9.3 SETUP C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0 312
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 332
v=0
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
The SETUP request for a URI specifies the transport mechanism to be S->C: RTSP/1.0 200 312 OK
used for the streamed media. A client can issue a SETUP request for
a stream that is already playing to change transport parameters. For
the benefit of any intervening firewalls, a client must indicate the
transport parameters even if it has no influence over these
parameters, for example, where the server advertises a fixed
multicast address.
This avoids having firewall to parse numerous different SETUP
presentation description formats, for information which is
irrelevant.
If the optional Require header is present, option tags within the The SETUP request for a URI specifies the transport mechanism to be
header indicate features needed by the requestor that are not used for the streamed media. A client can issue a SETUP request for a
required at the version level of the protocol. The Transport-Require stream that is already playing to change transport parameters, which a
header is used to indicate proxy-sensitive features that MUST be server MAY allow(If it does not allow it, it must respond with error
stripped by the proxy to the server if not supported. Furthermore, ``45x Method not valid in this state'' ). For the benefit of any
any Transport-Require header features that are not supported by the intervening firewalls, a client must indicate the transport parameters
proxy MUST be negatively acknowledged by the proxy to the client if even if it has no influence over these parameters, for example, where
not supported. the server advertises a fixed multicast address.
HS: In my opinion, the Require header should be replaced by Segregating content desciption into a DESCRIBE message and
PEP since PEP is standards-track, has more functionality transport information in SETUP avoids having firewall to parse
and somebody already did the work. numerous different presentation description formats for information
which is irrelevant to transport.
The Transport header specifies the transport parameters acceptable The Transport header specifies the transport parameters acceptable to
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 foo/bar/baz.rm RTSP/1.0 302 C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 302
Transport: rtp/udp;port=458 Transport: RTP/AVP;port=4588
S->C: RTSP/1.0 200 302 OK S->C: RTSP/1.0 200 302 OK
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Transport: cush/udp;port=458 Transport: RTP/AVP;port=4588
9.4 PLAY 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. For example, regardless of This allows precise editing.
how closely spaced the two PLAY commands in the example
below arrive, the server will play first second 10 through For example, regardless of how closely spaced the two PLAY commands in
15 and then, immediately following, seconds 20 to 25 and the example below arrive, the server will play first second 10 through
finally seconds 30 through the end. 15 and then, immediately following, seconds 20 to 25 and finally
seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 835 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 835
Range: npt=10-15 Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 836 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 836
Range: npt=20-25 Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 837 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 837
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 stream has been paused via PAUSE, stream delivery resumes at the pause
pause point. If a stream is playing, such a PLAY request causes no point. If a stream is playing, such a PLAY request causes no further
further action and can be used by the client to test server liveness. 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 immediately. The time parameter may be used to aid in synchronisation
synchronisation of streams obtained from different sources. of streams obtained from different sources.
For a on-demand stream, the server replies back with the actual range For a on-demand stream, the server replies back with the actual range
that will be played back. This may differ from the requested range if that will be played back. This may differ from the requested range if
alignment of the requested range to valid frame boundaries is alignment of the requested range to valid frame boundaries is required
required for the media source. If no range is specified in the for the media source. If no range is specified in the request, the
request, the current position is returned in the reply. The unit of current position is returned in the reply. The unit of the range in
the range in the reply is the same as that in the request. 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/1.0 833 C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 833
Range: smpte=0:10:20-;time=19970123T153600Z Range: smpte=0:10:20-;time=19970123T153600Z
skipping to change at page 28, line 41 skipping to change at line 1441
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/1.0 835 C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0 835
Range: clock=19961108T142300Z-19961108T143520Z Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 833 OK S->C: RTSP/1.0 200 833 OK
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 smpte format A media server only supporting playback MUST support the npt format
and MAY support the clock format. and MAY support the clock and smpte formats.
PAUSE
9.5 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 (halted) temporarily. If the request URL names a stream, only playback
playback and recording of that stream is halted. For example, for and recording of that stream is halted. For example, for audio, this
audio, this is equivalent to muting. If the request URL names a is equivalent to muting. If the request URL names a presentation or
presentation or group of streams, delivery of all currently active group of streams, delivery of all currently active streams within the
streams within the presentation or group is halted. After resuming presentation or group is halted. After resuming playback or recording,
playback or recording, synchronization of the tracks MUST be synchronization of the tracks MUST be maintained. Any server resources
maintained. Any server resources are kept. are kept.
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. The header must contain
exactly one value rather than a time range. The normal play time for exactly one value rather than a time range. The normal play time for
the stream is set to that value. The pause request becomes effective the stream is set to that value. The pause request becomes effective
the first time the server is encountering the time point specified. the first time the server is encountering the time point specified. If
If this header is missing, stream delivery is interrupted immediately this header is missing, stream delivery is interrupted immediately on
on receipt of the message. receipt of the message.
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, it stops immediately. If the pause request is for first play request, it stops immediately. If the pause request is for
NPT 16, it stops after completing the first play request and discards NPT 16, it stops after completing the first play request and discards
the second play request. 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
skipping to change at page 29, line 44 skipping to change at line 1486
second, overlapping range. Regardless of when the PAUSE request second, overlapping range. Regardless of when the PAUSE request
arrives, it sets the NPT to 14. 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.
Example: Example:
C->S: PAUSE /fizzle/foo RTSP/1.0 834 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 834
Session: 1234
S->C: RTSP/1.0 200 834 OK S->C: RTSP/1.0 200 834 OK
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
9.6 TEARDOWN TEARDOWN
Stop the stream delivery for the given URI, freeing the resources Stop the stream delivery for the given URI, freeing the resources
associated with it. If the URI is the root node for this associated with it. If the URI is the presentation URI for this
presentation, any RTSP session identifier associated with the session presentation, any RTSP session identifier associated with the session
is no longer valid. Unless all transport parameters are defined by is no longer valid. Unless all transport parameters are defined by the
the session description, a SETUP request has to be issued before the session description, a SETUP request has to be issued before the
session can be played again. session can be played again.
Example: Example:
C->S: TEARDOWN /fizzle/foo RTSP/1.0 892 C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 892
Session: 1234
S->C: RTSP/1.0 200 892 OK S->C: RTSP/1.0 200 892 OK
9.7 GET_PARAMETER GET_PARAMETER
The requests retrieves the value of a parameter of a presentation or The requests retrieves the value of a parameter of a presentation or
stream specified in the URI. Multiple parameters can be requested in stream specified in the URI. Multiple parameters can be requested in
the message body using the content type text/rtsp-parameters Note the message body using the content type text/rtsp-parameters. Note
that parameters include server and client statistics. IANA registers that parameters include server and client statistics. IANA registers
parameter names for statistics and other purposes. GET_PARAMETER with parameter names for statistics and other purposes. GET_PARAMETER with
no entity body may be used to test client or server liveness no entity body may be used to test client or server liveness
("ping"). (``ping'').
Example: Example:
S->C: GET_PARAMETER /fizzle/foo RTSP/1.0 431 S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 431
Content-Type: text/rtsp-parameters Content-Type: text/rtsp-parameters
Session: 1234 Session: 1234
Content-Length: 15 Content-Length: 15
packets_received packets_received
jitter jitter
C->S: RTSP/1.0 200 431 OK C->S: RTSP/1.0 200 431 OK
Content-Length: 46 Content-Length: 46
Content-Type: text/rtsp-parameters Content-Type: text/rtsp-parameters
packets_received: 10 packets_received: 10
jitter: 0.3838 jitter: 0.3838
9.8 SET_PARAMETER 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. A server MUST allow a to determine why a particular request failed. A server MUST allow a
parameter to be set repeatedly to the same value, but it MAY disallow 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 Restricting setting transport parameters to SETUP is for the
the benefit of firewalls. benefit of firewalls.
The parameters are split in a fine-grained fashion so that The parameters are split in a fine-grained fashion so that there
there can be more meaningful error indications. However, it can be more meaningful error indications. However, it may make
may make sense to allow the setting of several parameters sense to allow the setting of several parameters if an atomic
if an atomic setting is desirable. Imagine device control setting is desirable. Imagine device control where the client does
where the client does not want the camera to pan unless it not want the camera to pan unless it can also tilt to the right
can also tilt to the right angle at the same time. angle at the same time.
A SET_PARAMETER request without parameters can be used as a way to A SET_PARAMETER request without parameters can be used as a way to
detect client or server liveness. detect client or server liveness.
Example: Example:
C->S: SET_PARAMETER /fizzle/foo RTSP/1.0 421 C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 421
Content-type: text/rtsp-parameters Content-type: text/rtsp-parameters
fooparam: foostuff
barparam: barstuff barparam: barstuff
S->C: RTSP/1.0 450 421 Invalid Parameter S->C: RTSP/1.0 450 421 Invalid Parameter
Content-Length: 6 Content-Length: 6
barparam barparam
9.9 REDIRECT REDIRECT
A redirect request informs the client that it must connect to another A redirect request informs the client that it must connect to
server location. It contains the mandatory header Location, which another server location. It contains the mandatory header Location,
indicates that the client should issue a DESCRIBE for that URL. It which indicates that the client should issue requests for that URL. It
may contain the parameter Range, which indicates when the may contain the parameter Range, which indicates when the redirection
redirection takes effect. takes effect.
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 /fizzle/foo RTSP/1.0 732 S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0 732
Location: rtsp://bigserver.com:8001 Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z- Range: clock=19960213T143205Z-
9.10 RECORD 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. The Conference header is started, commence recording immediately.
mandatory.
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- request-URI or another URI. If the server does not use the
URI, the response SHOULD be 201 (Created) and contain an entity which request-URI, the response SHOULD be 201 (Created) and contain an
describes the status of the request and refers to the new resource, entity which describes the status of the request and refers to the new
and a Location header. resource, and a Location header.
A media server supporting recording of live presentations MUST A media server supporting recording of live presentations MUST support
support the clock range format; the smpte format does not make sense. 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 /meeting/audio.en RTSP/1.0 954 C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0 954
Session: 1234 Session: 1234
Conference: 128.16.64.19/32492374 Conference: 128.16.64.19/32492374
9.11 Embedded Binary Data 10.12 Embedded (Interleaved) Binary Data
Binary packets such as RTP data are encapsulated by an ASCII dollar Certain firewall designs and other circumstances may force a server
sign (24 decimal), followed by a one-byte session identifier, to interleave RTSP methods and stream data. This interleaving should
followed by the length of the encapsulated binary data as a binary, generally be avoided unless necessary since it complicates client and
two-byte integer in network byte order. The binary data follows server operation and imposes additional overhead. Interleaved binary
immediately afterwards, without a CRLF. data SHOULD only be used if RTSP is carried over TCP.
10 Status Code Definitions Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier, followed
by the length of the encapsulated binary data as a binary, two-byte
integer in network byte order. The stream data follows immediately
afterwards, without a CRLF, but including the upper-layer protocol
headers. Each $ block contains exactly one upper-layer protocol data
unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header 12.35.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 2
Transport: RTP/AVP/TCP;channel=0
S->C: RTSP/1.0 200 2 OK
Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;channel=0
Session: 12345
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0 3
Session: 12345
S->C: RTSP/1.0 200 3 OK
Session: 12345
Date: 05 Jun 1997 18:59:15 GMT
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
11 Status Code Definitions
Where applicable, HTTP status [H10] codes are re-used. Status codes Where applicable, HTTP status [H10] codes are re-used. Status codes
that have the same meaning are not repeated here. See Table 1 for a that have the same meaning are not repeated here. See Table 1 for a
listing of which status codes may be returned by which request. listing of which status codes may be returned by which request.
10.1 Redirection 3xx 11.1 Redirection 3xx
See [H10.3]. See [H10.3].
Within RTSP, redirection may be used for load balancing or Within RTSP, redirection may be used for load balancing or redirecting
redirecting stream requests to a server topologically closer to the stream requests to a server topologically closer to the client.
client. Mechanisms to determine topological proximity are beyond the Mechanisms to determine topological proximity are beyond the scope of
scope of this specification. this specification.
10.2 Client Error 4xx 11.2 Client Error 4xx
10.2.1 451 Parameter Not Understood 11.2.1 405 Method Not Allowed
The method specified in the request is not allowed for the resource
identified by the request URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource.
This status code is also to be used if a request attempts to use a
method not indicated during SETUP, e.g., if a RECORD request is issued
even though the mode parameter in the Transport header only specified
PLAY.
11.2.2 451 Parameter Not Understood
The recipient of the request does not support one or more parameters The recipient of the request does not support one or more parameters
contained in the request. contained in the request.
10.2.2 452 Conference Not Found 11.2.3 452 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.
10.2.3 453 Not Enough Bandwidth 11.2.4 453 Not Enough Bandwidth
The request was refused since there was insufficient bandwidth. This The request was refused since there was insufficient bandwidth. This
may, for example, be the result of a resource reservation failure. may, for example, be the result of a resource reservation failure.
10.2.4 45x Session Not Found 11.2.5 45x Session Not Found
The RTSP session identifier is invalid or has timed out. The RTSP session identifier is invalid or has timed out.
10.2.5 45x Method Not Valid in This State 11.2.6 45x Method Not Valid in This State
The client or server cannot process this request in its current The client or server cannot process this request in its current state.
state.
10.2.6 45x Header Field Not Valid for Resource 11.2.7 45x Header Field Not Valid for Resource
The server could not act on a required request header. For example, The server could not act on a required request header. For example, if
if PLAY contains the Range header field, but the stream does not PLAY contains the Range header field, but the stream does not allow
allow seeking. seeking.
11.2.8 45x Invalid Range
10.2.7 45x 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.
10.2.8 45x Parameter Is Read-Only 11.2.9 45x Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can only be read, but not The parameter to be set by SET_PARAMETER can only be read, but not
modified. modified.
11 Header Field Definitions 11.2.10 45x Aggregate operation not allowed
The requested method may not be applied on the URL in question since
it is an aggregate(presentation) URL. The method may be applied on a
stream URL.
11.2.11 45x Only aggregate operation allowed
The requested method may not be applied on the URL in question since
it is not an aggregate(presentation) URL. The method may be applied on
the presentation URL.
12 Header Field Definitions
HTTP/1.1 or other, non-standard header fields not listed here HTTP/1.1 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.
Tables 3 summarizes the header fields used by RTSP. Type "R" Tables 3 summarizes the header fields used by RTSP. Type ``g''
designates request headers, type "r" response headers. Fields marked designates general request headers, to be found in both requests and
with "req." in the column labeled "support" MUST be implemented by responses, type ``R'' designates request headers, type ``r'' response
the recipient for a particular method, while fields marked "opt." are headers, type ``e'' entity header fields. Fields marked with ``req.''
optional. Note that not all fields marked 'r' will be send in every in the column labeled ``support'' MUST be implemented by the recipient
request of this type; merely, that client (for response headers) and for a particular method, while fields marked ``opt.'' are optional.
server (for request headers) MUST implement them. The last column Note that not all fields marked 'r' will be send in every request of
lists the method for which this header field is meaningful; the this type; merely, that client (for response headers) and server (for
designation "entity" refers to all methods that return a message request headers) MUST implement them. The last column lists the method
body. Within this specification, DESCRIBE and GET_PARAMETER fall for which this header field is meaningful; the designation ``entity''
into this class. refers to all methods that return a message body. Within this
specification, DESCRIBE and GET_PARAMETER fall into this class.
If the field content does not apply to the particular resource, the If the field content does not apply to the particular resource, the
server MUST return status 45x (Header Field Not Valid for Resource). server MUST return status 45x (Header Field Not Valid for Resource).
11.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
The "level" parameter for presentation descriptions is
properly defined as part of the MIME type registration, not
here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl, application/sdp;level=2
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
Authorization R opt. all Authorization R opt. all
Bandwidth R opt. SETUP Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control Rr opt. SETUP Cache-Control g opt. SETUP
Conference R opt. SETUP Conference R opt. SETUP
Connection Rr req. all Connection g req. all
Content-Encoding R req. SET_PARAMETER Content-Encoding e req. SET_PARAMETER
Content-Encoding r req. DESCRIBE Content-Encoding e req. DESCRIBE, ANNOUNCE
Content-Length R req. SET_PARAMETER Content-Language e req. DESCRIBE, ANNOUNCE
Content-Length r req. entity Content-Length e req. SET_PARAMETER, ANNOUNCE
Content-Type R req. SET_PARAMETER Content-Length e req. entity
Content-Type e req. SET_PARAMETER, ANNOUNCE
Content-Type r req. entity Content-Type r req. entity
Date Rr opt. all Date g opt. all
Expires r opt. DESCRIBE Expires e opt. DESCRIBE, ANNOUNCE
From R opt. all
If-Modified-Since R opt. DESCRIBE, SETUP If-Modified-Since R opt. DESCRIBE, SETUP
Last-Modified r opt. entity Last-Modified e opt. entity
Public r opt. all Public r opt. all
Range R opt. PLAY, PAUSE, RECORD Range R opt. PLAY, PAUSE, RECORD
Range r opt. PLAY, PAUSE, RECORD Range r opt. PLAY, PAUSE, RECORD
Referer R opt. all Referer R opt. all
Require R req. all
Retry-After r opt. all Retry-After r opt. all
Scale Rr opt. PLAY, RECORD Scale Rr opt. PLAY, RECORD
Session Rr req. all but SETUP, OPTIONS Session Rr req. all but SETUP, OPTIONS
Server r opt. all Server r opt. all
Speed Rr opt. PLAY Speed Rr opt. PLAY
Transport Rr req. SETUP Transport Rr req. SETUP
Transport-Require R xeq. all Transport-Info r req. PLAY
User-Agent R opt. all User-Agent R opt. all
Via Rr opt. all Via g opt. all
WWW-Authenticate r opt. all WWW-Authenticate r opt. all
!
Table 3: Overview of RTSP header fields Table 3: Overview of RTSP header fields
11.2 Accept-Encoding 12.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
The ``level'' parameter for presentation descriptions is properly
defined as part of the MIME type registration, not here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl, application/sdp;level=2
12.2 Accept-Encoding
See [H14.3] See [H14.3]
11.3 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.
11.4 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
11.5 Authorization 12.5 Authorization
See [H14.8] See [H14.8]
11.6 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. in bits per second.
The bandwidth available to the client may change during an RTSP
session, e.g., due to modem retraining.
Bandwidth = "Bandwidth" ":" 1*DIGIT Bandwidth = "Bandwidth" ":" 1*DIGIT
Example: Example:
Bandwidth: 4000 Bandwidth: 4000
11.7 Blocksize 12.7 Blocksize
This request header field is sent from the client to the media server This request header field is sent from the client to the media
asking the server for a particular media packet size. This packet server asking the server for a particular media packet size. This
size does not include lower-layer headers such as IP, UDP, or RTP. packet size does not include lower-layer headers such as IP, UDP, or
The server is free to use a blocksize which is lower than the one RTP. The server is free to use a blocksize which is lower than the one
requested. The server MAY truncate this packet size to the closest requested. The server MAY truncate this packet size to the closest
multiple of the minimum media-specific block size or overrides it multiple of the minimum media-specific block size or override it with
with the media specific size if necessary. The block size is a the media specific size if necessary. The block size is a strictly
strictly positive decimal number and measured in octets. The server positive decimal number and measured in octets. The server only
only returns an error (416) if the value is syntactically invalid. returns an error (416) if the value is syntactically invalid.
11.8 Cache-Control 12.8 C-PEP
This corresponds to the C-PEP: header in the ``Protocol Extension
Protocol'' defined in RFC XXXX [21]. This field differs from the PEP
field (Section 12.24) only in that it is hop-by-hop rather than
end-to-end as PEP is. Servers and proxies MUST parse this field and
MUST return "420 Bad Extension" when there is a PEP extension of
strength "must". See RFC XXXX for more details on this.
12.9 C-PEP-Info
This corresponds to the C-PEP-Info: header in the ``Protocol
Extension Protocol'' defined in RFC XXXX [21].
12.10 Cache-Control
The Cache-Control general header field is used to specify directives 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 response. Note: Cache-Control does not govern the caching of responses
responses as for HTTP, but rather of the stream identified by the as for HTTP, but rather of the stream identified by the SETUP request.
SETUP request. Responses to RTSP requests are not cacheable. Responses to RTSP requests are not cacheable, except for responses to
DESCRIBE.
[HS: Should there be an exception for 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 = cache-request-directive =
"no-cache" "no-cache"
| "max-stale" | "max-stale"
| "min-fresh" | "min-fresh"
skipping to change at page 37, line 48 skipping to change at line 1907
| "private" | "private"
| "no-cache" | "no-cache"
| "no-transform" | "no-transform"
| "must-revalidate" | "must-revalidate"
| "proxy-revalidate" | "proxy-revalidate"
| "max-age" "=" delta-seconds | "max-age" "=" delta-seconds
| cache-extension | cache-extension
cache-extension = token [ "=" ( token | quoted-string ) ] cache-extension = token [ "=" ( token | quoted-string ) ]
no-cache: Indicates that the media stream MUST NOT be cached no-cache:
anywhere. This allows an origin server to prevent caching even Indicates that the media stream MUST NOT be cached anywhere.
by caches that have been configured to return stale responses to This allows an origin server to prevent caching even by caches
client requests. that have been configured to return stale responses to client
requests.
public: Indicates that the media stream is cachable by any cache. public:
Indicates that the media stream is cachable by any cache.
private: Indicates that the media stream is intended for a single private:
user and MUST NOT be cached by a shared cache. A private (non- Indicates that the media stream is intended for a single user
shared) cache may cache the media stream. and MUST NOT be cached by a shared cache. A private
(non-shared) cache may cache the media stream.
no-transform: An intermediate cache (proxy) may find it useful to no-transform:
convert the media type of certain stream. A proxy might, for An intermediate cache (proxy) may find it useful to convert the
example, convert between video formats to save cache space or to media type of certain stream. A proxy might, for example,
reduce the amount of traffic on a slow link. Serious operational convert between video formats to save cache space or to reduce
the amount of traffic on a slow link. Serious operational
problems may occur, however, when these transformations have problems may occur, however, when these transformations have
been applied to streams intended for certain kinds of been applied to streams intended for certain kinds of
applications. For example, applications for medical imaging, applications. For example, applications for medical imaging,
scientific data analysis and those using end-to-end scientific data analysis and those using end-to-end
authentication, all depend on receiving a stream that is bit for authentication, all depend on receiving a stream that is bit
bit identical to the original entity-body. Therefore, if a for bit identical to the original entity-body. Therefore, if a
response includes the no-transform directive, an intermediate response includes the no-transform directive, an intermediate
cache or proxy MUST NOT change the encoding of the stream. cache or proxy MUST NOT change the encoding of the stream.
Unlike HTTP, RTSP does not provide for partial transformation at Unlike HTTP, RTSP does not provide for partial transformation
this point, e.g., allowing translation into a different at this point, e.g., allowing translation into a different
language. language.
only-if-cached: In some cases, such as times of extremely poor only-if-cached:
network connectivity, a client may want a cache to return only In some cases, such as times of extremely poor network
those media streams that it currently has stored, and not to connectivity, a client may want a cache to return only those
receive these from the origin server. To do this, the client may media streams that it currently has stored, and not to receive
these from the origin server. To do this, the client may
include the only-if-cached directive in a request. If it 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 operated Timeout) status. However, if a group of caches is being
as a unified system with good internal connectivity, such a operated as a unified system with good internal connectivity,
request MAY be forwarded within that group of caches. such a request MAY be forwarded within that group of caches.
max-stale: Indicates that the client is willing to accept a media max-stale:
stream that has exceeded its expiration time. If max-stale is Indicates that the client is willing to accept a media stream
assigned a value, then the client is willing to accept a that has exceeded its expiration time. If max-stale is assigned
response that has exceeded its expiration time by no more than a value, then the client is willing to accept a response that
the specified number of seconds. If no value is assigned to has exceeded its expiration time by no more than the specified
max-stale, then the client is willing to accept a stale response number of seconds. If no value is assigned to max-stale, then
of any age. the client is willing to accept a stale response of any age.
min-fresh: Indicates that the client is willing to accept a media min-fresh:
stream whose freshness lifetime is no less than its current age Indicates that the client is willing to accept a media stream
plus the specified time in seconds. That is, the client wants a whose freshness lifetime is no less than its current age plus
the specified time in seconds. That is, the client wants a
response that will still be fresh for at least the specified response that will still be fresh for at least the specified
number of seconds. number of seconds.
must-revalidate: When the must-revalidate directive is present in a must-revalidate:
SETUP response received by a cache, that cache MUST NOT use the When the must-revalidate directive is present in a SETUP
entry after it becomes stale to respond to a subsequent request response received by a cache, that cache MUST NOT use the entry
without first revalidating it with the origin server. (I.e., the after it becomes stale to respond to a subsequent request
cache must do an end-to-end revalidation every time, if, based without first revalidating it with the origin server. (I.e.,
solely on the origin server's Expires, the cached response is the cache must do an end-to-end revalidation every time, if,
stale.) based solely on the origin server's Expires, the cached
response is stale.)
11.9 Conference 12.11 Conference
This request header field establishes a logical connection between a This request header field establishes a logical connection between a
conference, established using non-RTSP means, and an RTSP stream. The conference, established using non-RTSP means, and an RTSP stream. The
conference-id must not be changed for the same RTSP session. conference-id must 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 12.12 Connection
11.10 Connection
See [H14.10]. See [H14.10].
11.11 Content-Encoding 12.13 Content-Encoding
See [H14.12] See [H14.12]
11.12 Content-Length 12.14 Content-Language
See [H14.13]
12.15 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 after the double CRLF following the last header). Unlike HTTP, it MUST
MUST be included in all messages that carry content beyond the header be included in all messages that carry content beyond the header
portion of the message. It is interpreted according to [H14.14]. portion of the message. It is interpreted according to [H14.14].
11.13 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.
11.14 Date 12.17 Date
See [H14.19]. See [H14.19].
11.15 Expires 12.18 Expires
The Expires entity-header field gives the date/time after which the The Expires entity-header field gives the date/time after which the
media-stream should be considered stale. A stale cache entry may not media-stream should be considered stale. A stale cache entry may not
normally be returned by a cache (either a proxy cache or an user normally be returned by a cache (either a proxy cache or an user agent
agent cache) unless it is first validated with the origin server (or cache) unless it is first validated with the origin server (or with an
with an intermediate cache that has a fresh copy of the entity). See intermediate cache that has a fresh copy of the entity). See section
section 13.2 for further discussion of the expiration model. 13.2 for further discussion of the 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 resource will change or cease to exist at, before, or after that time.
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
skipping to change at page 40, line 43 skipping to change at line 2048
expired"). 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. an Expires date that is equal to the Date header value.
To mark a response as "never expires," an origin server should use an To mark a response as "never expires," an origin server should use an
Expires date approximately one year from the time the response is Expires date approximately one year from the time the response is
sent. RTSP/1.0 servers should not send Expires dates more than one sent. RTSP/1.0 servers should not send Expires dates more than one
year in the future. year in the future.
The presence of an Expires header field with a date value of some The presence of an Expires header field with a date value of some time
time in the future on a media stream that otherwise would by default in the future on a media stream that otherwise would by default be
be non-cacheable indicates that the media stream is cachable, unless non-cacheable indicates that the media stream is cachable, unless
indicated otherwise by a Cache-Control header field (Section 11.8. indicated otherwise by a Cache-Control header field (Section 12.10).
11.16 If-Modified-Since 12.19 From
See [H14.22].
12.20 Host
This HTTP request header field is not needed for RTSP. It should be
silently ignored if sent.
12.21 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE 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 setup ( SETUP); instead, a 304 (not modified) stream will not be setup ( 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
11.17 Last-modified 12.22 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 variant was last modified. See which the origin server believes the variant was last modified. See
[H14.29]. If the request URI refers to an aggregate, the field [H14.29]. If the request URI refers to an aggregate, the field
indicates the last modification time across all leave nodes of that indicates the last modification time across all leave nodes of that
aggregate. aggregate.
11.18 Location 12.23 Location
See [H14.30]. See [H14.30].
11.19 Nack-Transport-Require 12.24 PEP
Negative acknowledgement of features not supported by the server. If
there is a proxy on the path between the client and the server, the
proxy MUST insert a message reply with an error message 506 (Feature
not supported).
HS: Same caveat as for Require applies.
11.20 Range This corresponds to the PEP: header in the ``Protocol Extension
Protocol'' defined in RFC XXXX. Servers MUST parse this field and MUST
return ``420 Bad Extension'' when there is a PEP extension of strength
``must'' (see RFC XXXX).
This request header field specifies a range of time. The range can be 12.25 PEP-Info
specified in a number of units. This specification defines the smpte
(see Section 3.4) and clock (see Section 3.6) range units. Within
RTSP, byte ranges [H14.36.1] are not meaningful and MUST NOT be used.
The header may also contain a time parameter in UTC, specifying the
time at which the operation is to be made effective.
Range = "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ] This corresponds to the PEP-Info: header in the ``Protocol Extension
Protocol'' defined in RFC XXXX.
ranges-specifier = npt-range | utc-range | smpte-range 12.26 Proxy-Authenticate
Example: See [H14.33].
Range: clock=19960213T143205Z-;Time=19970123T143720Z 12.27 Public
The notation is similar to that used for the HTTP/1.1 See [H14.35].
header. It allows to select a clip from the media object,
to play from a given point to the end and from the current
location to a given point.
11.21 Require 12.28 Range
The Require header is used by clients to query the server about This request header field specifies a range of time. The range can
features that it may or may not support. The server MUST respond to be specified in a number of units. This specification defines the
this header by negatively acknowledging those features which are NOT smpte (see Section 3.5) and clock (see Section 3.7) range units.
supported in the Unsupported header. Within RTSP, byte ranges [H14.36.1] are not meaningful and MUST NOT be
used. The header may also contain a time parameter in UTC, specifying
the time at which the operation is to be made effective. Servers
supporting the Range header MUST understand the NPT and SMPTE range
formats.
HS: Naming of features -- yet another name space. I believe Range = "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ]
this header field to be redundant. PEP should be used
instead.
For example ranges-specifier = npt-range | utc-range | smpte-range
C->S: SETUP /foo/bar/baz.rm RTSP/1.0 302 Example:
Require: funky-feature Range: clock=19960213T143205Z-;time=19970123T143720Z
Funky-Parameter: funkystuff
S->C: RTSP/1.0 200 506 Option not supported The notation is similar to that used for the HTTP/1.1 header. It
Unsupported: funky-feature allows to select a clip from the media object, to play from a given
point to the end and from the current location to a given point.
The start of playback can be scheduled for at any time in the
future, although a server may refuse to keep server resources for
extended idle periods.
C->S: SETUP /foo/bar/baz.rm RTSP/1.0 303 12.29 Referer
S->C: RTSP/1.0 200 303 OK See [H14.37]. The URL refers to that of the presentation
This is to make sure that the client-server interaction will proceed description, typically retrieved via HTTP.
optimally when all options are understood by both sides, and only
slow down if options aren't understood (as in the case above). For a
well-matched client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation mechanisms. In
addition, it also removes state ambiguity when the client requires
features that the server doesn't understand.
11.22 Retry-After 12.30 Retry-After
See [H14.38]. See [H14.38].
11.23 Scale 12.31 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 forward viewing rate. If not 1, the value corresponds to the rate with
with respect to normal viewing rate. For example, a ratio of 2 respect to normal viewing rate. For example, a ratio of 2 indicates
indicates twice the normal viewing rate ("fast forward") and a ratio twice the normal viewing rate (``fast forward'') and a ratio of 0.5
of 0.5 indicates half the normal viewing rate. In other words, a indicates half the normal viewing rate. In other words, a ratio of 2
ratio of 2 has normal play time increase at twice the wallclock rate. has normal play time increase at twice the wallclock rate. For every
For every second of elapsed (wallclock) time, 2 seconds of content second of elapsed (wallclock) time, 2 seconds of content will be
will be delivered. A negative value indicates reverse direction. 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 the audio while preserving pitch or, less desirably, deliver fragments
fragments of audio. 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 restrict the range of scale values that it supports. The response MUST
MUST contain the actual scale value chosen by the server. 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
11.24 Speed 12.32 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. A negative value indicates that the stream speed of zero is invalid. If the request contains a Range parameter,
is to be played back in reverse direction. the new speed value will take effect at that time.
HS: With 'Scale', the negative value is redundant and
should probably be removed since it only leads to possible
conflicts when Scale is positive and Speed negative.
If the request contains a Range parameter, 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
11.25 Server Use of this field changes the bandwidth used for data delivery. It is
meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. Implementors
should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss
rates.
12.33 Server
See [H14.39] See [H14.39]
11.26 Session 12.34 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 and has the same syntax as a conference identifier. the media server (see Section 3.4). Once a client receives a Session
Once a client receives a Session identifier, it MUST return it for identifier, it MUST return it for any request related to that session.
any request related to that session.
HS: This may be redundant with the standards-track HTTP Session = "Session" ":" session-id
state maintenance mechanism [2]. The equivalent way of
doing this would be for the server to send Set-Cookie:
Session="123"; Version=1; Path = "/twister" and for the
client to return later Cookie: Session = "123"; $Version=1;
$Path = "/twister" response to the TEARDOWN message, the
server would simply send Set-Cookie: Session="123";
Version=1; Max-Age=0 to get rid of the cookie on the client
side. Cookies also have a time-out, so that a server may
limit the lifetime of a session at will. Unlike a web
browser, a client would not store these states on disk. To
avoid privacy issues, we should prohibit the Host
parameter.
11.27 Transport Note that a session identifier identifies a RTSP session across
transport sessions or connections. Control messages for more than one
RTSP URL may be sent within a single RTSP session. Hence, it is
possible that clients use the same session for controlling many
streams comprising a presentation, as long as all the streams come
from the same server. (See example in Section 14). However, multiple
``user'' sessions for the same URL from the same client MUST use
different session identifiers.
This request header indicates which transport protocol is to be used The session identifier is needed to distinguish several delivery
and configures its parameters such as multicast, compression, requests for the same URL coming from the same client.
multicast time-to-live and destination port for a single stream. It
sets those values not already determined by a presentation
description. In some cases, the presentation description contains all
necessary information. In those cases, a Transport header field
(and the SETUP request containing it) are not needed.
in whatever protocol is being used by the control stream. Currently, 12.35 Transport
the next-layer protocols RTP is defined. Parameters may be added to
each protocol, separated by a semicolon. For RTP, the boolean
parameter compressed is defined, indicating compressed RTP according
to RFC XXXX. For multicast UDP, the integer parameter ttl defines
the time-to-live value to be used. The client may specify the
multicast address with the multicast parameter. A server SHOULD
authenticate the client before allowing the client to direct a media
stream to a multicast address not chosen by the server to avoid
becoming the unwitting perpetrator of a denial-of-service attack. For
UDP and TCP, the parameter port defines the port data is to be sent
to.
The SSRC parameter indicates the RTP SSRC value that should be This request header indicates which transport protocol is to be used
(request) or will be (response) used by the media server. This and configures its parameters such as destination address,
parameter is only valid for unicast transmission. It identifies the compression, multicast time-to-live and destination port for a single
synchronization source to be associated with the media stream. stream. It sets those values not already determined by a presentation
description.
Transports are comma separated, listed in order of preference.
Parameters may be added to each tranpsort, separated by a semicolon.
The Transport header MAY also be used to change certain transport 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 options acceptable to the client. In that case, the server MUST return
return a single option which was actually chosen. The Transport a single option which was actually chosen.
header field makes sense only for an individual media stream, not a
presentation. The syntax for the transport specifier is
transport/profile/lower-transport. Defaults for "lower-transport" are
specific to the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport:
General parameters:
destination:
The address to which a stream will be sent. The client may
specify the multicast address with the destination parameter. A
server SHOULD authenticate the client and SHOULD log such
attempts before allowing the client to direct a media stream to
an address not chosen by the server to avoid becoming the
unwitting perpetrator of a remote-controlled denial-of-service
attack. This is particularly important if RTSP commands are
issued via UDP, but TCP cannot be relied upon as reliable means
of client identification by itself. A server SHOULD not allow a
client to direct media streams to an address that differs from
the address commands are coming from.
mode:
The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY. For RECORD, the append flag
indicates that the media data should be appended to the
existing resource rather than overwriting it. If appending is
requested and the server does not support this, it MUST refuse
the request rather than overwrite the resouce identified by the
URI. The append parameter is ignored if the mode parameter does
not contain RECORD.
interleaved:
The interleaved parameter implies mixing the media stream with
the control stream, in whatever protocol is being used by the
control stream. Currently, the next-layer protocols RTP is
defined. The `channel' parameter defines the channel number to
be used in the $ statement (see section 10.12).
Multicast specific:
ttl:
multicast time-to-live
RTP Specific:
compressed:
Boolean parameter indicating compressed RTP according to RFC
XXXX.
port:
RTP/RTCP destination ports on client. The client receives RTCP
reports on the value of port plus one, as is standard RTP
convention.
cport:
the control port that the data server wishes the client to send
its RTCP reports to.
ssrc:
Indicates the RTP SSRC [19, Sec. 3] value that should be
(request) or will be (response) used by the media server. This
parameter is only valid for unicast transmission. It identifies
the synchronization source to be associated with the media
stream.
Transport = "Transport" ":" Transport = "Transport" ":"
1#transport-protocol/upper-layer *parameter 1#transport-protocol/profile[/lower-transport] *parameter
transport-protocol = "UDP" | "TCP" transport-protocol = "RTP"
upper-layer = "RTP" profile = "AVP"
parameters = ";" "multicast" [ "=" mca ] lower-transport = "TCP" | "UDP"
parameter = ";" "destination" [ "=" address ]
| ";" "compressed" | ";" "compressed"
| ";" "interleaved" | ";" "channel" "=" channel
| ";" "append"
| ";" "ttl" "=" ttl | ";" "ttl" "=" ttl
| ";" "port" "=" port | ";" "port" "=" port
| ";" "cport" "=" port
| ";" "ssrc" "=" ssrc | ";" "ssrc" "=" ssrc
| ";" "mode" = <"> 1#mode <">
ttl = 1*3(DIGIT) ttl = 1*3(DIGIT)
port = 1*5(DIGIT) port = 1*5(DIGIT)
ssrc = 8*8(HEX) ssrc = 8*8(HEX)
mca = host channel = 1*3(DIGIT)
address = host
mode = "PLAY" | "RECORD" *parameter
Example: Example:
Transport: RTP/AVP;compressed;ttl=127;port=3456;
mode="PLAY,RECORD;append"
Transport: udp/rtp;compressed;ttl=127;port=3456 The Transport header is restricted to describing a single RTP
stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of
firewalls.
11.28 Transport-Require 12.36 Transport-Info
The Transport-Require header is used to indicate proxy-sensitive This field is used to set Transport specific parameters in the PLAY
features that MUST be stripped by the proxy to the server if not response.
supported. Furthermore, any Transport-Require header features that
are not supported by the proxy MUST be negatively acknowledged by the
proxy to the client if not supported.
See Section 11.21 for more details on the mechanics of this message seq:
and a usage example. Indicates the sequence number of the first packet of the
stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that
originated after the seek.
HS: Same caveat as for Require applies. Transport-Info = "Transport-Info" ":"
1#transport-protocol/profile[/lower-transport] ";"
streamid
*parameter
transport-protocol = "RTP"
profile = "AVP"
lower-transport = "TCP" | "UDP"
stream-id = "streamid" "=" streamid
parameter = ";" "seq" "=" sequence number
sequence-number = 1*16(DIGIT)
11.29 Unsupported Example:
Transport-Info: RTP/AVP;streamid=0;seq=43754027,
RTP/AVP;streamid=1;seq=34834738
See Section 11.21 for a usage example. 12.37 User-Agent
HS: same caveat as for Require applies. See [H14.42]
11.30 User-Agent 12.38 Vary
See [H14.42] See [H14.43]
11.31 Via 12.39 Via
See [H14.44]. See [H14.44].
11.32 WWW-Authenticate 12.40 WWW-Authenticate
See [H14.46]. See [H14.46].
12 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 cachable, with the significantly in that respect. Responses are not cachable, with the
exception of the stream description returned by DESCRIBE. (Since the exception of the stream description returned by DESCRIBE. (Since the
responses for anything but DESCRIBE and GET_PARAMETER do not return responses for anything but DESCRIBE and GET_PARAMETER do not return
any data, caching is not really an issue for these requests.) any data, caching is not really an issue for these requests.) However,
However, it is desirable for the continuous media data, typically it is desirable for the continuous media data, typically delivered
delivered out-of-band with respect to RTSP, to be cached. out-of-band with respect to RTSP, to be cached.
On receiving a SETUP or PLAY request, the proxy would ascertain as On receiving a SETUP or PLAY request, the proxy would ascertain as to
to whether it has an up-to-date copy of the continuous media content. whether it has an up-to-date copy of the continuous media content. If
If not, it would modify the SETUP transport parameters as not, it would modify the SETUP transport parameters as appropriate and
appropriate and forward the request to the origin server. Subsequent forward the request to the origin server. Subsequent control commands
control commands such as PLAY or PAUSE would pass the proxy such as PLAY or PAUSE would pass the proxy unmodified. The proxy would
unmodified. The proxy would then pass the continuous media data to then pass the continuous media data to the client, while possibly
the client, while possibly making a local copy for later re-use. The making a local copy for later re-use. The exact behavior allowed to
exact behavior allowed to the cache is given by the cache-response the cache is given by the cache-response directives described in
directives described in Section 11.8. A cache MUST answer any Section 12.10. A cache MUST answer any DESCRIBE requests if it is
DESCRIBE requests if it is currently serving the stream to the currently serving the stream to the requestor, as it is possible that
requestor, as it is possible that low-level details of the stream low-level details of the stream description may have changed on the
description may have changed on the origin-server. origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the "cut- Note that an RTSP cache, unlike the HTTP cache, is of the
through" variety. Rather than retrieving the whole resource from the ``cut-through'' variety. Rather than retrieving the whole resource
origin server, the cache simply copies the streaming data as it from the origin server, the cache simply copies the streaming data as
passes by on its way to the client, thus, it does not introduce it passes by on its way to the client, thus, it does not introduce
additional latency. additional latency.
To the client, an RTSP proxy cache would appear like a regular media To the client, an RTSP proxy cache would appear like a regular media
server, to the media origin server like a client. Just like an HTTP server, to the media origin server like a client. Just like an HTTP
cache has to store the content type, content language, etc. for the cache has to store the content type, content language, etc. for the
objects it caches, a media cache has to store the presentation objects it caches, a media cache has to store the presentation
description. Typically, a cache would eliminate all transport- description. Typically, a cache would eliminate all
references (that is, multicast information) from the presentation transport-references (that is, multicast information) from the
description, since these are independent of the data delivery from presentation description, since these are independent of the data
the cache to the client. Information on the encodings remains the delivery from the cache to the client. Information on the encodings
same. If the cache is able to translate the cached media data, it remains the same. If the cache is able to translate the cached media
would create a new presentation description with all the encoding data, it would create a new presentation description with all the
possibilities it can offer. encoding possibilities it can offer.
13 Examples 14 Examples
The following examples reference stream description formats that are The following examples reference stream description formats that are
not finalized, such as RTSL and SDP. Please do not use these examples not finalized, such as RTSL and SDP. Please do not use these examples
as a reference for those formats. as a reference for those formats.
13.1 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 time line of the movie. give an indication about the time line of the movie.
In our example, the client is only interested in the last part of the In our example, the client is only interested in the last part of the
movie. The server requires authentication for this movie. The audio movie. The server requires authentication for this movie.
track can be dynamically switched between between two sets of
encodings. The URL with scheme rtpsu indicates the media servers
want to use UDP for exchanging RTSP messages.
C->W: DESCRIBE /twister HTTP/1.1 C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com Host: www.example.com
Accept: application/rtsl; application/sdp Accept: application/sdp
W->C: 200 OK W->C: HTTP/1.0 200 OK
Content-Type: application/rtsl Content-Type: application/sdp
<session> v=0
<group language=en lipsync> o=- 2890844526 2890842807 IN IP4 192.16.24.202
<switch> s=RTSP Session
<track type=audio m=audio 0 RTP/AVP 0
e="PCMU/8000/1" a=murl:rtsp://audio.example.com/twister/audio.en
src="rtsp://audio.example.com/twister/audio.en/lofi"> m=video 0 RTP/AVP 31
<track type=audio a=murl:rtsp://audio.example.com/twister/video
e="DVI4/16000/2" pt="90 DVI4/8000/1"
src="rtsp://audio.example.com/twister/audio.en/hifi">
</switch>
<track type="video/jpeg"
src="rtspu://video.example.com/twister/video">
</group>
</session>
C->A: SETUP rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 1 C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 1
Transport: rtp/udp;compression;port=3056 Transport: rtp/udp;port=3056
A->C: RTSP/1.0 200 1 OK A->C: RTSP/1.0 200 1 OK
Session: 1234 Session: 1234
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 1 C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 1
Transport: rtp/udp;compression;port=3058 Transport: rtp/udp;port=3058
V->C: RTSP/1.0 200 1 OK V->C: RTSP/1.0 200 1 OK
Session: 1235 Session: 1235
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 2 C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 2
Session: 1235 Session: 1235
Range: smpte=0:10:00- Range: smpte=0:10:00-
V->C: RTSP/1.0 200 2 OK V->C: RTSP/1.0 200 2 OK
C->A: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 2 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 2
Session: 1234 Session: 1234
Range: smpte=0:10:00- Range: smpte=0:10:00-
A->C: 200 2 OK A->C: RTSP/1.0 200 2 OK
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 3 C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 3
Session: 1234 Session: 1234
A->C: 200 3 OK A->C: RTSP/1.0 200 3 OK
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 3 C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 3
Session: 1235 Session: 1235
V->C: 200 3 OK V->C: RTSP/1.0 200 3 OK
Even though the audio and video track are on two different servers, Even though the audio and video track are on two different servers,
may start at slightly different times and may drift with respect to and may start at slightly different times and may drift with respect
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.
13.2 Live Media Presentation Using Multicast 14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in
which multiple continuous media types pertaining to the same end-user
presentation are present. In effect, the container file represents a
RTSP presentation, with each of its components being RTSP streams.
Container files are a widely used means to store such presentations.
While the components are essentially transported as independant
streams, it is desirable to maintain a common context for those
streams at the server end.
This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case of
any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent
selective retreival of the streams by client in order to preserve the
artistic effect of the combined media presentation. Similarly, in such
a tightly bound presentation, it is desirable to be able to control
all the streams via a single control message using an aggregate URL.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M . The movie is
stored in a container file. The client has obtained a RTSP URL to the
container file.
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0 1
M->C: RTSP/1.0 200 1 OK
Content-Type: application/sdp
Content-Length: 64
s= sample rtsp presentation
r = rtsp://foo/twister /* aggregate URL*/
m= audio 0 RTP/AVP 0
r = rtsp://foo/twister/audio
m=video 0 RTP/AVP 26
r = rtsp://foo/twister/video
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0 2
Transport: RTP/AVP;port=8000
M->C: RTSP/1.0 200 2 OK
Session: 1234
C->M: SETUP rtsp://foo/twister/video RTSP/1.0 3
Transport: RTP/AVP;port=8002
Session: 1234
M->C: RTSP/1.0 200 3 OK
Session: 1234
C->M: PLAY rtsp://foo/twister RTSP/1.0 4
Range: npt=0-
Session: 1234
M->C: RTSP/1.0 200 4 OK
Session: 1234
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0 5
Session: 1234
M->C: RTSP/1.0 4xx 5 Only aggregate operation allowed
C->M: PAUSE rtsp://foo/twister RTSP/1.0 6
Session: 1234
M->C: RTSP/1.0 200 6 OK
Session: 1234
C->M: SETUP rtsp://foo/twister RTSP/1.0 7
Transport: RTP/AVP;port=10000
M->C: RTSP/1.0 4xx 7 Aggregate operation not allowed
In the first instance of failure, the client tries to pause one
stream(in this case video) of the presentation which is disallowed for
that presentation by the server. In the second instance, the aggregate
URL may not be used for SETUP and one control message is required per
stream to setup transport parameters.
This keeps the syntax of the Transport header simple, and allows
easy parsing of transport information by firewalls.
14.3 Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, we The media server M chooses the multicast address and port. Here, we
assume that the web server only contains a pointer to the full assume that the web server only contains a pointer to the full
description, while the media server M maintains the full description. description, while the media server M maintains the full description.
During the RTSP session, a new subtitling stream is added.
C->W: GET /concert HTTP/1.1 C->W: GET /concert.sdp HTTP/1.1
Host: www.example.com Host: www.example.com
W->C: HTTP/1.1 200 OK W->C: HTTP/1.1 200 OK
Content-Type: application/rtsl Content-Type: application/rtsl
<session> <session>
<track id=17 src="rtsp://live.example.com/concert/audio"> <track src="rtsp://live.example.com/concert/audio">
</session> </session>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0 1 C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0 1
M->C: RTSP/1.0 200 1 OK M->C: RTSP/1.0 200 1 OK
Content-Type: application/rtsl Content-Type: application/sdp
<track id=17 type=audio address=224.2.0.1 port=3456 ttl=16> v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 3456 RTP/AVP 0
c=IN IP4 224.2.0.1/16
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 2 C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 2
Transport: multicast=224.2.0.1; port=3456; ttl=16 Transport: multicast=224.2.0.1; port=3456; ttl=16
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 3 C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 3
Range: smpte 1:12:0
M->C: RTSP/1.0 405 3 No positioning possible M->C: RTSP/1.0 200 3 OK
M->C: DESCRIBE concert RTSP/1.0
Content-Type: application/rtsl
<session>
<track id=17
media=audio/g.728 src="rtsp://live.example.com/concert/audio">
<track id=18
media=text/html src="rtsp://live.example.com/concert/lyrics">
</session>
C->M: PLAY rtsp://live.example.com/concert/lyrics RTSP/1.0
The attempt to position the stream fails since this is a live The attempt to position the stream fails since this is a live
presentation. presentation.
13.3 Playing media into an existing session 14.4 Playing media into an existing session
A conference participant C wants to have the media server M play back A conference participant C wants to have the media server M play back
a demo tape into an existing conference. When retrieving the a demo tape into an existing conference. When retrieving the
presentation description, C indicates to the media server that the presentation description, C indicates to the media server that the
network addresses and encryption keys are already given by the network addresses and encryption keys are already given by the
conference, so they should not be chosen by the server. The example conference, so they should not be chosen by the server. The example
omits the simple ACK responses. omits the simple ACK responses.
C->M: GET /demo HTTP/1.1 C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0 1
Host: www.example.com Accept: application/sdp
Accept: application/rtsl, application/sdp
M->C: HTTP/1.1 200 1 OK M->C: RTSP/1.0 200 1 OK
Content-type: application/rtsl Content-type: application/rtsl
<session> v=0
<track type=audio/g.723.1 o=- 2890844526 2890842807 IN IP4 192.16.24.202
src="rtsp://server.example.com/demo/548/sound"> s=RTSP Session
</session> m=audio 0 RTP/AVP 0
C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0 2 C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0 2
Conference: 218kadjk Conference: 218kadjk
13.4 Recording 14.5 Recording
The conference participant C asks the media server M to record a The conference participant C asks the media server M to record a
meeting. If the presentation description contains any alternatives, meeting. If the presentation description contains any alternatives,
the server records them all. the server records them all.
C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 89 C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 90
Content-Type: application/sdp Content-Type: application/sdp
v=0 v=0
s=Mbone Audio s=Mbone Audio
i=Discussion of Mbone Engineering Issues i=Discussion of Mbone Engineering Issues
M->C: 415 89 Unsupported Media Type M->C: RTSP/1.0 200 90 OK
Accept: application/rtsl
C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 90 C->S: SETUP rtsp://server.example.com/meeting RTSP/1.0 91
Content-Type: application/rtsl Transport: RTP/AVP;mode=record
M->C: 200 90 OK S->C: RTSP/1.0 200 91 OK
Transport: RTP/AVP;port=3244;mode=record
Session: 508876
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 91 C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 92
Session: 508876
Range: clock 19961110T1925-19961110T2015 Range: clock 19961110T1925-19961110T2015
14 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 (HTTP/1.1). as used in RFC 2068 (HTTP/1.1).
14.1 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
LWS = [CRLF] 1*( SP | HT ) LWS = [CRLF] 1*( SP | HT )
TEXT = <any OCTET except CTLs> TEXT = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@" tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | " | "," | ";" | ":" | "\" | <">
| "/" | "[" | "]" | "?" | "=" | "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT | "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials> token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> ) quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">> qdtext = <any TEXT except <">>
quoted-pair = " quoted-pair = "\" CHAR
message-header = field-name ":" [ field-value ] CRLF message-header = field-name ":" [ field-value ] CRLF
field-name = token field-name = token
field-value = *( field-content | LWS ) field-value = *( field-content | LWS )
field-content = <the OCTETs making up the field-value and consisting field-content = <the OCTETs making up the field-value and consisting
of either *TEXT or combinations of token, tspecials, of either *TEXT or combinations of token, tspecials,
and quoted-string> and quoted-string>
15 Security Considerations 16 Security Considerations
The protocol offers the opportunity for a remote-control denial-of-
service attack. The attacker, using a forged source IP address, can
ask for a stream to be played back to that forged IP address.
Since there is no relation between a transport layer connection and The protocol offers the opportunity for a remote-controlled
an RTSP session, it is possible for a malicious client to issue denial-of-service attack.
requests with random session identifiers which would affect
unsuspecting clients. This does not require spoofing of network The attacker, using a forged source IP address, can ask for a stream
packet addresses. The server SHOULD use a large random session to be played back to that forged IP address. Thus, an RTSP server
identifier to make this attack more difficult. SHOULD only allow client-specified destinations for RTSP-initiated
traffic flows if the server has verified the client's identity, e.g.,
using the RTSP authentication mechanisms.
Since there is no relation between a transport layer connection and an
RTSP session, it is possible for a malicious client to issue requests
with random session identifiers which would affect unsuspecting
clients. This does not require spoofing of network packet addresses.
The server SHOULD use a large random session identifier to make this
attack more difficult.
Both problems can be be prevented by appropriate authentication. Both problems can be be prevented by appropriate authentication.
Servers SHOULD implement both basic and digest [8] authentication.
In addition, the security considerations outlined in [H15] apply. In addition, the security considerations outlined in [H15] apply.
A RTSP Protocol State Machines 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.
[TBD: should we allow for the trivial case of a server that only
implements the PLAY message, with no control.]
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. (A identified by the stream URL and the RTSP session identifier. Any
server may choose to generate dynamic presentation descriptions where request/reply using aggregate URLs denoting RTSP presentations
the URL is unique for a particular RTSP session and thus may not need comprised of multiple streams will have an effect on the individual
an explicit RTSP session identifier in the request header.) Any states of all the streams. For example, if the presentation /movie
request/reply using URLs denoting an RTSP session comprised of contains two streams /movie/audio and /movie/video, then the following
multiple streams will have an effect on the individual states of all command:
the substreams. For example, if the stream /movie contains two
substreams /movie/audio and /movie/video, then the following command:
PLAY /movie RTSP/1.0 559 PLAY rtsp://foo.com/movie RTSP/1.0 559
Session: 12345 Session: 12345
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 This example does not imply a standard way to represent streams in
substreams in URLs or a relation to the filesystem. See URLs or a relation to the filesystem. See Section 3.2.
Section 3.2.
The requests OPTIONS, DESCRIBE, GET_PARAMETER, SET_PARAMETER do
not have any effect on client or server state and are therefore not
listed in the state tables.
Client and servers MUST disregard messages with a sequence number The requests OPTIONS, DESCRIBE, GET_PARAMETER, SET_PARAMETER do not
less than the last one. If no message has been received, the first have any effect on client or server state and are therefore not listed
received message's sequence number will be the starting point. 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: SETUP has been sent, waiting for reply. Init:
SETUP has been sent, waiting for reply.
Ready: SETUP reply received OR after playing, PAUSE reply received. Ready:
SETUP reply received OR after playing, PAUSE reply received.
Playing: PLAY reply received Playing:
PLAY reply received
Recording: RECORD reply received Recording:
RECORD reply received
The client changes state on receipt of replies to requests. If no In general, the client changes state on receipt of replies to
requests. Note that some requests are effective at a future time or
position(such as a PAUSE), and state also changes accordingly. If no
explicit SETUP is required for the object (for example, it is explicit SETUP is required for the object (for example, it is
available via a multicast group), state begins at READY. In this available via a multicast group), state begins at READY. In this case,
case, there are only two states, READY and PLAYING. there are only two states, READY and PLAYING.
The "next state" column indicates the state assumed after receiving a The client also changes state from Playing/Recording to Ready when the
success response (2xx). If a request yields a status code greater or end of the requested range is reached.
equal to 300, the client state becomes Init, with the exception of
status codes 401 (Unauthorized) and 402 (Payment Required), where the
state remains unchanged and the request should be re-issued with the
appropriate authentication or payment information. Messages not
listed for each state MUST NOT be issued by the client in that state,
with the exception of messages not affecting state, as listed above.
Receiving a REDIRECT from the server is equivalent to receiving a 3xx
redirect status from the server.
HS: Depends on allowing PLAY without SETUP. After 4xx or The ``next state'' column indicates the state assumed after receiving
5xx error, do we go back to Init? a success response (2xx). If a request yields a status code of 3xx,
the state becomes Init, and a status code of 4xx yields no change in
state. Messages not listed for each state MUST NOT be issued by the
client in that state, with the exception of messages not affecting
state, as listed above. Receiving a REDIRECT from the server is
equivalent to receiving a 3xx redirect status from the server.
state message next state state message next state
_______________________________________________________
Init SETUP Ready Init SETUP Ready
TEARDOWN Init TEARDOWN Init
Ready PLAY Playing Ready PLAY Playing
RECORD Recording RECORD Recording
TEARDOWN Init TEARDOWN Init
Playing PAUSE Ready Playing PAUSE Ready
TEARDOWN Init TEARDOWN Init
PLAY Playing PLAY Playing
RECORD Recording
SETUP Playing (changed transport) SETUP Playing (changed transport)
Recording PAUSE Init Recording PAUSE Ready
TEARDOWN Init TEARDOWN Init
PLAY Playing
RECORD Recording RECORD Recording
SETUP Recording (changed transport) SETUP Recording (changed transport)
A.2 Server State Machine A.2 Server State Machine
The server can assume the following states: The server can assume the following states:
Init: The initial state, no valid SETUP received. Init:
The initial state, no valid SETUP received.
Ready: Last SETUP received was successful, reply sent or after Ready:
Last SETUP received was successful, reply sent or after
playing, last PAUSE received was successful, reply sent. playing, last PAUSE received was successful, reply sent.
Playing: Last PLAY received was successful, reply sent. Data is Playing:
being sent. Last PLAY received was successful, reply sent. Data is being
sent.
Recording: The server is recording media data. Recording:
The server is recording media data.
The server changes state on receiving requests. If the server is in In general,the server changes state on receiving requests. If the
state Playing or Recording and in unicast mode, it MAY revert to Init server is in state Playing or Recording and in unicast mode, it MAY
and tear down the RTSP session if it has not received "wellness" revert to Init and tear down the RTSP session if it has not received
information, such as RTCP reports, from the client for a defined ``wellness'' information, such as RTCP reports, from the client for a
interval, with a default of one minute. If the server is in state defined interval, with a default of one minute. If the server is in
Ready, it MAY revert to Init if it does not receive an RTSP request state Ready, it MAY revert to Init if it does not receive an RTSP
for an interval of more than one minute. request for an interval of more than one minute. Note that some
requests(such as PAUSE) may be effective at a future time or position,
and server state transitions at the appropriate time. The server
reverts from state Playing or Recording to state Ready at the end of
the range requested by the client.
The REDIRECT message, when sent, is effective immediately. If a The REDIRECT message, when sent, is effective immediately unless it
similar change of location occurs at a certain time in the future, has a Range: header specifying when the redirect is effective. In such
this is assumed to be indicated by the presentation description. a case, server state will also change at the appropriate time.
SETUP is valid in states Init and Ready only. An error message should If no explicit SETUP is required for the object, state starts at
be returned in other cases. If no explicit SETUP is required for the READY, there are only two states READY and PLAYING.
object, state starts at READY, there are only two states READY and
PLAYING. The ``next state'' column indicates the state assumed after sending a
success response (2xx). If a request results in a status code of 3xx,
the state becomes Init. A status code of 4xx results in no change.
state message next state state message next state
___________________________________
Init SETUP Ready Init SETUP Ready
TEARDOWN Init TEARDOWN Init
Ready PLAY Playing Ready PLAY Playing
SETUP Ready SETUP Ready
TEARDOWN Ready TEARDOWN Init
RECORD Recording
Playing PLAY Playing Playing PLAY Playing
PAUSE Ready PAUSE Ready
TEARDOWN Ready TEARDOWN Init
RECORD Recording
SETUP Playing SETUP Playing
Recording RECORD Recording Recording RECORD Recording
PAUSE Ready PAUSE Ready
TEARDOWN Ready TEARDOWN Init
PLAY Playing
SETUP Recording SETUP Recording
B Open Issues B Open Issues
o Define text/rtsp-parameter MIME type. 1.
Define text/rtsp-parameter MIME type.
o HS believes that RTSP should only control individual media 2.
Reverse: Scale: -1, with reversed start times, or both?
3.
HS believes that RTSP should only control individual media
objects rather than aggregates. This avoids disconnects between objects rather than aggregates. This avoids disconnects between
presentation descriptions and streams and avoids having to deal presentation descriptions and streams and avoids having to deal
separately with single-host and multi-host case. Cost: several separately with single-host and multi-host case. Cost: several
PLAY/PAUSE/RECORD in one packet, one for each stream. PLAY/PAUSE/RECORD in one packet, one for each stream.
4.
o Allow changing of transport for a stream that's playing? May Allow changing of transport for a stream that's playing? May
not be a great idea since the same can be accomplished by tear not be a great idea since the same can be accomplished by tear
down and re-setup. down and re-setup. Exception: near-video-on-demand, where the
server changes the address in a PLAY response. Servers may not
o Allow fragment (#) identifiers for controlling substreams in be able to reliably send TEARDOWN to clients and the client
Quicktime, AVI and ASF files? wouldn't know why this happened in any event.
5.
o How does the server get back to the client unless a persistent How does the server get back to the client unless a persistent
connection is used? Probably cannot, in general. connection is used? Probably cannot, in general.
6.
o Cache and proxy behavior? Server issues TEARDOWN and other 'event' notifications to
client? This raises the problem discussed in the previous open
o Session: or Set-Cookie: ? issue, but is useful for the client if the data stream contains
no end indication.
o When do relative RTSP URLs make sense?
o Nack-require, etc. are dubious. This is getting awfully close
to the HTTP extension mechanisms [19] in complexity, but is
different.
o Use HTTP absolute path + Host field or do the right thing and
carry full URL, including host in request?
C Changes C Changes
Since the February 1997 version, the following changes were made: Since the March 1997 version, the following changes were made:
o Various editorial changes and clarifications.
o Removed references to SDF and replaced by RTSL.
o Added Scale general header.
o Clarify behavior of PLAY.
o Rename GET to DESCRIBE.
o Removed SESSION since it is just DESCRIBE in the other
direction.
o Rename CLOSE to TEARDOWN, in symmetry with SETUP.
o Terminology adjusted to "presentation" and "stream".
o Redundant syntax BNF in appendix removed since it just
duplicates HTTP spec.
o Beginnings of cache control. * Allowing the Transport header to direct media streams to unicast
and multicast addresses, with an appropriate warning about
denial-of-service attacks.
* Add mode parameter to Transport header to allow RECORD or PLAY.
* The Embedded binary data section was modified to clearly indicate
the stream the data corresponds to, and a reference to the
Transport header was added.
* The Transport header format has been changed to use a more general
means to specify data channel and application level protocol. It
also conveys the port to be used at the server for RTCP messages,
and the start sequence number that will be used in the RTP
packets.
* The use of the Session: header has been enhanced. Requests for
multiple URLs may be sent in a single session.
* There is a distinction between aggregate(presentation) URLs and
stream URLs. Error codes have been added to reflect the fact that
some methods may be allowed only on a particular type of URL.
* Example showing the use of aggregate/presentation control using a
single RTSP session has been added.
* Support for the PEP(Protocol Extension Protocol) headers has been
added.
* Server-Client DESCRIBE messages have been renamed to ANNOUNCE for
better clarity and differentiation.
Changes are marked by changebars in the margins of the PostScript Note that this list does not reflect minor changes in wording or
version. correction of typographical errors.
D Author Addresses D Author 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 electronic mail: schulzrinne@cs.columbia.edu
skipping to change at page 57, line 39 skipping to change at line 2938
D Author Addresses D Author 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 electronic mail: schulzrinne@cs.columbia.edu
Anup Rao Anup Rao
Netscape Communications Corp. Netscape Communications Corp.
501 E. Middlefield Road
Mountain View, CA 94043
USA USA
electronic mail: anup@netscape.com electronic mail: anup@netscape.com
Robert Lanphier Robert Lanphier
Progressive Networks Progressive Networks
1111 Third Avenue Suite 2900 1111 Third Avenue Suite 2900
Seattle, WA 98101 Seattle, WA 98101
USA USA
electronic mail: robla@prognet.com electronic mail: robla@prognet.com
E Acknowledgements E Acknowledgements
This draft is based on the functionality of the RTSP draft. It also
borrows format and descriptions from HTTP/1.1. This draft is based on the functionality of the original RTSP draft
submitted in October 96. It also borrows format and descriptions from
HTTP/1.1.
This document has benefited greatly from the comments of all those 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 Eduardo F. Llach Rahul Agarwal Eduardo F. Llach
Bruce Butterfield Rob McCool Bruce Butterfield Rob McCool
Steve Casner David Oran
Martin Dunsmuir Sujal Patel Martin Dunsmuir Sujal Patel
Eric Fleischman Eric Fleischman
Mark Handley Igor Plotnikov Mark Handley Igor Plotnikov
Peter Haight Pinaki Shah Peter Haight Pinaki Shah
Brad Hefta-Gaub Jeff Smith Brad Hefta-Gaub Jeff Smith
John K. Ho Alexander Sokolsky John K. Ho Alexander Sokolsky
Ruth Lang Dale Stammen Ruth Lang Dale Stammen
Stephanie Leif John Francis Stracke Stephanie Leif John Francis Stracke
F Bibliography References
[1] H. Schulzrinne, "RTP profile for audio and video conferences with
minimal control," RFC 1890, Internet Engineering Task Force, Jan.
1996.
[2] D. Kristol and L. Montulli, "HTTP state management mechanism,"
RFC 2109, Internet Engineering Task Force, Feb. 1997.
[3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
"Internationalization of the hypertext markup language," RFC 2070,
Internet Engineering Task Force, Jan. 1997.
[4] S. Bradner, "Key words for use in RFCs to indicate requirement 1
levels," Internet Draft, Internet Engineering Task Force, Jan. 1997. H. Schulzrinne, ``RTP profile for audio and video conferences
with minimal control,'' RFC 1890, Internet Engineering Task
Force, Jan. 1996.
2
D. Kristol and L. Montulli, ``HTTP state management
mechanism,'' RFC 2109, Internet Engineering Task Force, Feb.
1997.
3
F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
``Internationalization of the hypertext markup language,'' RFC
2070, Internet Engineering Task Force, Jan. 1997.
4
S. Bradner, ``Key words for use in RFCs to indicate requirement
levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997.
5
R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T.
Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC
2068, Internet Engineering Task Force, Jan. 1997.
6
M. Handley, ``SDP: Session description protocol,'' Internet
Draft, Internet Engineering Task Force, Nov. 1996.
Work in progress. Work in progress.
7
[5] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-Lee, A. Freier, P. Karlton, and P. Kocher, ``The TLS protocol,''
"Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet Internet Draft, Internet Engineering Task Force, Dec. 1996.
Engineering Task Force, Jan. 1997. Work in progress.
8
[6] M. Handley, "SDP: Session description protocol," Internet Draft, J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and
Internet Engineering Task Force, Nov. 1996. Work in progress. E. L. Stewart, ``An extension to HTTP: digest access
authentication,'' RFC 2069, Internet Engineering Task Force,
[7] A. Freier, P. Karlton, and P. Kocher, "The TLS protocol," Jan. 1997.
Internet Draft, Internet Engineering Task Force, Dec. 1996. Work in 9
progress. J. Postel, ``User datagram protocol,'' STD 6, RFC 768, Internet
[8] J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and E.
L. Stewart, "An extension to HTTP: digest access authentication,"
RFC 2069, Internet Engineering Task Force, Jan. 1997.
[9] J. Postel, "User datagram protocol," STD 6, RFC 768, Internet
Engineering Task Force, Aug. 1980. Engineering Task Force, Aug. 1980.
10
[10] R. Hinden and C. Partridge, "Version 2 of the reliable data R. Hinden and C. Partridge, ``Version 2 of the reliable data
protocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr. protocol (RDP),'' RFC 1151, Internet Engineering Task Force,
1990. Apr. 1990.
11
[11] J. Postel, "Transmission control protocol," STD 7, RFC 793, J. Postel, ``Transmission control protocol,'' STD 7, RFC 793,
Internet Engineering Task Force, Sept. 1981. Internet Engineering Task Force, Sept. 1981.
12
[12] M. Handley, H. Schulzrinne, and E. Schooler, "SIP: Session M. Handley, H. Schulzrinne, and E. Schooler, ``SIP: Session
initiation protocol," Internet Draft, Internet Engineering Task initiation protocol,'' Internet Draft, Internet Engineering
Force, Dec. 1996. Work in progress. Task Force, Dec. 1996.
Work in progress.
[13] P. McMahon, "GSS-API authentication method for SOCKS version 5," 13
RFC 1961, Internet Engineering Task Force, June 1996. P. McMahon, ``GSS-API authentication method for SOCKS version
5,'' RFC 1961, Internet Engineering Task Force, June 1996.
[14] D. Crocker, "Augmented BNF for syntax specifications: ABNF," 14
Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in D. Crocker, ``Augmented BNF for syntax specifications: ABNF,''
progress. Internet Draft, Internet Engineering Task Force, Oct. 1996.
Work in progress.
[15] R. Elz, "A compact representation of IPv6 addresses," RFC 1924, 15
Internet Engineering Task Force, Apr. 1996. R. Elz, ``A compact representation of IPv6 addresses,'' RFC
1924, Internet Engineering Task Force, Apr. 1996.
[16] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource 16
locators (URL)," RFC 1738, Internet Engineering Task Force, Dec. T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform
1994. resource locators (URL),'' RFC 1738, Internet Engineering Task
Force, Dec. 1994.
[17] International Telecommunication Union, "Visual telephone systems 17
and equipment for local area networks which provide a non-guaranteed International Telecommunication Union, ``Visual telephone
quality of service," Recommendation H.323, Telecommunication systems and equipment for local area networks which provide a
Standardization Sector of ITU, Geneva, Switzerland, May 1996. non-guaranteed quality of service,'' Recommendation H.323,
Telecommunication Standardization Sector of ITU, Geneva,
[18] ISO/IEC, "Information technology -- generic coding of moving Switzerland, May 1996.
pictures and associated audio informaiton -- part 6: extension for 18
digital storage media and control," Draft International Standard ISO ISO/IEC, ``Information technology - generic coding of moving
13818-6, International Organization for Standardization ISO/IEC pictures and associated audio informaiton - part 6: extension
JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995. for digital storage media and control,'' Draft International
Standard ISO 13818-6, International Organization for
[19] D. Connolly, "PEP: an extension mechanism for http," Internet Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
Draft, Internet Engineering Task Force, Jan. 1997. Work in progress. Nov. 1995.
19
Table of Contents H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
``RTP: a transport protocol for real-time applications,'' RFC
1 Introduction ........................................ 1 1889, Internet Engineering Task Force, Jan. 1996.
1.1 Purpose ............................................. 1 20
1.2 Requirements ........................................ 3 J. Miller, P. Resnick, and D. Singer, ``Rating Services and
1.3 Terminology ......................................... 3 Rating Systems(and Their Machine Readable Descriptions), ''
1.4 Protocol Properties ................................. 5 REC-PICS-services-961031, Worldwide Web Consortium, Oct. 1996.
1.5 Extending RTSP ...................................... 6 21
1.6 Overall Operation ................................... 7 D. Connolly, R. Khare, H.F. Nielsen, ``PEP - an Extension
1.7 RTSP States ......................................... 8 Mechanism for HTTP", Internet draft, work-in-progress. W3C
1.8 Relationship with Other Protocols ................... 9 Draft WD-http-pep-970714
2 Notational Conventions .............................. 10 http://www.w3.org/TR/WD-http-pep-970714, July, 1996.
3 Protocol Parameters ................................. 10
3.1 H3.1 ................................................ 10
3.2 RTSP URL ............................................ 10
3.3 Conference Identifiers .............................. 11
3.4 SMPTE Relative Timestamps ........................... 12
3.5 Normal Play Time .................................... 13
3.6 Absolute Time ....................................... 13
4 RTSP Message ........................................ 13
4.1 Message Types ....................................... 14
4.2 Message Headers ..................................... 14
4.3 Message Body ........................................ 14
4.4 Message Length ...................................... 14
5 Request ............................................. 15
6 Response ............................................ 16
6.1 Status-Line ......................................... 17
6.1.1 Status Code and Reason Phrase ....................... 17
6.1.2 Response Header Fields .............................. 19
7 Entity .............................................. 19
7.1 Entity Header Fields ................................ 21
7.2 Entity Body ......................................... 21
8 Connections ......................................... 21
8.1 Pipelining .......................................... 22
8.2 Reliability and Acknowledgements .................... 22
9 Method Definitions .................................. 23
9.1 OPTIONS ............................................. 24
9.2 DESCRIBE ........................................... 25
9.3 SETUP .............................................. 26
9.4 PLAY ............................................... 27
9.5 PAUSE .............................................. 28
9.6 TEARDOWN ........................................... 30
9.7 GET_PARAMETER ...................................... 30
9.8 SET_PARAMETER ...................................... 31
9.9 REDIRECT ........................................... 31
9.10 RECORD ............................................. 32
9.11 Embedded Binary Data ................................ 32
10 Status Code Definitions ............................. 33
10.1 Redirection 3xx ..................................... 33
10.2 Client Error 4xx .................................... 33
10.2.1 451 Parameter Not Understood ........................ 33
10.2.2 452 Conference Not Found ............................ 33
10.2.3 453 Not Enough Bandwidth ............................ 33
10.2.4 45x Session Not Found ............................... 33
10.2.5 45x Method Not Valid in This State .................. 33
10.2.6 45x Header Field Not Valid for Resource ............. 33
10.2.7 45x Invalid Range ................................... 33
10.2.8 45x Parameter Is Read-Only .......................... 34
11 Header Field Definitions ............................ 34
11.1 Accept .............................................. 34
11.2 Accept-Encoding ..................................... 35
11.3 Accept-Language ..................................... 35
11.4 Allow ............................................... 36
11.5 Authorization ....................................... 36
11.6 Bandwidth ........................................... 36
11.7 Blocksize ........................................... 36
11.8 Cache-Control ....................................... 37
11.9 Conference .......................................... 39
11.10 Connection .......................................... 39
11.11 Content-Encoding .................................... 39
11.12 Content-Length ...................................... 39
11.13 Content-Type ........................................ 39
11.14 Date ................................................ 40
11.15 Expires ............................................. 40
11.16 If-Modified-Since ................................... 41
11.17 Last-modified ....................................... 41
11.18 Location ............................................ 41
11.19 Nack-Transport-Require .............................. 41
11.20 Range ............................................... 41
11.21 Require ............................................. 42
11.22 Retry-After ......................................... 43
11.23 Scale ............................................... 43
11.24 Speed ............................................... 44
11.25 Server .............................................. 44
11.26 Session ............................................. 44
11.27 Transport ........................................... 45
11.28 Transport-Require ................................... 46
11.29 Unsupported ......................................... 46
11.30 User-Agent .......................................... 47
11.31 Via ................................................. 47
11.32 WWW-Authenticate .................................... 47
12 Caching ............................................. 47
13 Examples ............................................ 48
13.1 Media on Demand (Unicast) ........................... 48
13.2 Live Media Presentation Using Multicast ............. 49
13.3 Playing media into an existing session .............. 50
13.4 Recording ........................................... 51
14 Syntax .............................................. 52
14.1 Base Syntax ......................................... 52
15 Security Considerations ............................. 52
A RTSP Protocol State Machines ........................ 53
A.1 Client State Machine ................................ 54
A.2 Server State Machine ................................ 55
B Open Issues ......................................... 56
C Changes ............................................. 56
D Author Addresses .................................... 57
E Acknowledgements .................................... 57
F Bibliography ........................................ 58
 End of changes. 

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