draft-ietf-mmusic-rtsp-03.txt   draft-ietf-mmusic-rtsp-04.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
draft-ietf-mmusic-rtsp-03.txt Columbia U./Netscape/Progressive Networks draft-ietf-mmusic-rtsp-04.txt Columbia U./Netscape/Progressive Networks
July 30, 1997 Expires: January 30, 1998 September 17, 1997 Expires: March 17, 1998
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 line 40 skipping to change at line 40
The Real Time Streaming Protocol, or RTSP, is an application-level The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP, sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC
1889). 1889).
This is a snapshot of the current draft which will become the next
version of the ``official'' Internet Draft.
H. Schulzrinne, A. Rao, R. Lanphier Page 1 H. Schulzrinne, A. Rao, R. Lanphier Page 1
Contents Contents
* Contents
* 1 Introduction * 1 Introduction
+ 1.1 Purpose + 1.1 Purpose
+ 1.2 Requirements + 1.2 Requirements
+ 1.3 Terminology + 1.3 Terminology
+ 1.4 Protocol Properties + 1.4 Protocol Properties
+ 1.5 Extending RTSP + 1.5 Extending RTSP
+ 1.6 Overall Operation + 1.6 Overall Operation
+ 1.7 RTSP States + 1.7 RTSP States
+ 1.8 Relationship with Other Protocols + 1.8 Relationship with Other Protocols
* 2 Notational Conventions * 2 Notational Conventions
skipping to change at line 87 skipping to change at line 91
+ 8.2 Entity Body + 8.2 Entity Body
* 9 Connections * 9 Connections
+ 9.1 Pipelining + 9.1 Pipelining
+ 9.2 Reliability and Acknowledgements + 9.2 Reliability and Acknowledgements
* 10 Method Definitions * 10 Method Definitions
+ 10.1 OPTIONS + 10.1 OPTIONS
+ 10.2 DESCRIBE + 10.2 DESCRIBE
+ 10.3 ANNOUNCE + 10.3 ANNOUNCE
+ 10.4 SETUP + 10.4 SETUP
+ 10.5 PLAY + 10.5 PLAY
H. Schulzrinne, A. Rao, R. Lanphier Page 2
+ 10.6 PAUSE + 10.6 PAUSE
+ 10.7 TEARDOWN + 10.7 TEARDOWN
+ 10.8 GET_PARAMETER + 10.8 GET_PARAMETER
+ 10.9 SET_PARAMETER + 10.9 SET_PARAMETER
+ 10.10 REDIRECT + 10.10 REDIRECT
H. Schulzrinne, A. Rao, R. Lanphier Page 2
+ 10.11 RECORD + 10.11 RECORD
+ 10.12 Embedded (Interleaved) Binary Data + 10.12 Embedded (Interleaved) Binary Data
* 11 Status Code Definitions * 11 Status Code Definitions
+ 11.1 Redirection 3xx + 11.1 Redirection 3xx
+ 11.2 Client Error 4xx + 11.2 Client Error 4xx
o 11.2.1 405 Method Not Allowed o 11.2.1 405 Method Not Allowed
o 11.2.2 451 Parameter Not Understood o 11.2.2 451 Parameter Not Understood
o 11.2.3 452 Conference Not Found o 11.2.3 452 Conference Not Found
o 11.2.4 453 Not Enough Bandwidth o 11.2.4 453 Not Enough Bandwidth
o 11.2.5 45x Session Not Found o 11.2.5 454 Session Not Found
o 11.2.6 45x Method Not Valid in This State o 11.2.6 455 Method Not Valid in This State
o 11.2.7 45x Header Field Not Valid for Resource o 11.2.7 456 Header Field Not Valid for Resource
o 11.2.8 45x Invalid Range o 11.2.8 457 Invalid Range
o 11.2.9 45x Parameter Is Read-Only o 11.2.9 458 Parameter Is Read-Only
o 11.2.10 45x Aggregate operation not allowed o 11.2.10 459 Aggregate operation not allowed
o 11.2.11 45x Only aggregate operation allowed o 11.2.11 460 Only aggregate operation allowed
* 12 Header Field Definitions * 12 Header Field Definitions
+ 12.1 Accept + 12.1 Accept
+ 12.2 Accept-Encoding + 12.2 Accept-Encoding
+ 12.3 Accept-Language + 12.3 Accept-Language
+ 12.4 Allow + 12.4 Allow
+ 12.5 Authorization + 12.5 Authorization
+ 12.6 Bandwidth + 12.6 Bandwidth
+ 12.7 Blocksize + 12.7 Blocksize
+ 12.8 C-PEP + 12.8 Cache-Control
+ 12.9 C-PEP-Info + 12.9 Conference
+ 12.10 Cache-Control + 12.10 Connection
+ 12.11 Conference + 12.11 Content-Base
+ 12.12 Connection + 12.12 Content-Encoding
+ 12.13 Content-Encoding + 12.13 Content-Language
+ 12.14 Content-Language + 12.14 Content-Length
+ 12.15 Content-Length + 12.15 Content-Location
+ 12.16 Content-Type + 12.16 Content-Type
+ 12.17 Date + 12.17 CSeq
+ 12.18 Expires + 12.18 Date
+ 12.19 From + 12.19 Expires
+ 12.20 Host + 12.20 From
+ 12.21 If-Modified-Since + 12.21 Host
+ 12.22 Last-Modified + 12.22 If-Match
+ 12.23 Location + 12.23 If-Modified-Since
+ 12.24 PEP + 12.24 Last-Modified
+ 12.25 PEP-Info
+ 12.26 Proxy-Authenticate
+ 12.27 Public
+ 12.28 Range
+ 12.29 Referer
+ 12.30 Retry-After
+ 12.31 Scale
+ 12.32 Speed
+ 12.33 Server
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+ 12.34 Session + 12.25 Location
+ 12.35 Transport + 12.26 Proxy-Authenticate
+ 12.36 Transport-Info + 12.27 Proxy-Require
+ 12.37 User-Agent + 12.28 Public
+ 12.38 Vary + 12.29 Range
+ 12.39 Via + 12.30 Referer
+ 12.40 WWW-Authenticate + 12.31 Retry-After
+ 12.32 Require
+ 12.33 RTP-Info
+ 12.34 Scale
+ 12.35 Speed
+ 12.36 Server
+ 12.37 Session
+ 12.38 Timestamp
+ 12.39 Transport
+ 12.40 Unsupported
+ 12.41 User-Agent
+ 12.42 Vary
+ 12.43 Via
+ 12.44 WWW-Authenticate
* 13 Caching * 13 Caching
* 14 Examples * 14 Examples
+ 14.1 Media on Demand (Unicast) + 14.1 Media on Demand (Unicast)
+ 14.2 Streaming of a Container file + 14.2 Streaming of a Container file
+ 14.3 Live Media Presentation Using Multicast + 14.3 Live Media Presentation Using Multicast
+ 14.4 Playing media into an existing session + 14.4 Playing media into an existing session
+ 14.5 Recording + 14.5 Recording
* 15 Syntax * 15 Syntax
+ 15.1 Base Syntax + 15.1 Base Syntax
* 16 Security Considerations * 16 Security Considerations
* A RTSP Protocol State Machines * A RTSP Protocol State Machines
+ A.1 Client State Machine + A.1 Client State Machine
+ A.2 Server State Machine + A.2 Server State Machine
* B Open Issues * B Interaction with RTP
* C Changes * C Use of SDP for RTSP Session Descriptions
* D Author Addresses + C.1 Specification
* E Acknowledgements o C.1.1 Control URL
o C.1.2 Media streams
o C.1.3 Payload type(s)
o C.1.4 Format specific parameters
o C.1.5 Length of presentation
o C.1.6 Time of availability
o C.1.7 Connection Information
o C.1.8 Entity Tag
+ C.2 Scenario A
+ C.3 Scenario B
H. Schulzrinne, A. Rao, R. Lanphier Page 4
* D Minimal RTSP implementation
+ D.1 Client
o D.1.1 Basic Playback
o D.1.2 Authentication-enabled
+ D.2 Server
o D.2.1 Basic Playback
o D.2.2 Authentication-enabled
* E Open Issues
* F Changes
* G Author Addresses
* H Acknowledgements
* References * References
1 Introduction 1 Introduction
1.1 Purpose 1.1 Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls The Real-Time Streaming Protocol (RTSP) establishes and controls
either a single or several time-synchronized streams of continuous either a single or several time-synchronized streams of continuous
media such as audio and video. It does not typically deliver the media such as audio and video. It does not typically deliver the
continuous streams itself, although interleaving of the continuous continuous streams itself, although interleaving of the continuous
media stream with the control stream is possible (see Section 10.12). media stream with the control stream is possible (see Section 10.12).
In other words, RTSP acts as a ``network remote control'' for In other words, RTSP acts as a ``network remote control'' for
multimedia servers. multimedia servers.
The set of streams to be controlled is defined by a presentation The set of streams to be controlled is defined by a presentation
description. This memorandum does not define a format for a description. This memorandum does not define a format for a
presentation description. presentation description.
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:
H. Schulzrinne, A. Rao, R. Lanphier Page 5
* RTSP introduces a number of new methods and has a different * RTSP introduces a number of new methods and has a different
protocol identifier. protocol identifier.
* An RTSP server needs to maintain state by default in almost all * An RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP. cases, as opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests. * Both an RTSP server and client can issue requests.
* Data is carried out-of-band, by a different protocol. (There is an * Data is carried out-of-band, by a different protocol. (There is an
exception to this.) exception to this.)
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts [3]. consistent with current HTML internationalization efforts [3].
* The Request-URI always contains the absolute URI. Because of * The Request-URI always contains the absolute URI. Because of
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The protocol supports the following operations: The protocol supports the following operations:
Retrieval of media from media server: Retrieval of media from media server:
The client can request a presentation description via HTTP or The client can request a presentation description via HTTP or
some other method. If the presentation is being multicast, the some other method. If the presentation is being multicast, the
presentation description contains the multicast addresses and presentation description contains the multicast addresses and
ports to be used for the continuous media. If the presentation ports to be used for the continuous media. If the presentation
is to be sent only to the client via unicast, the client is to be sent only to the client via unicast, the client
provides the destination for security reasons. provides the destination for security reasons.
H. Schulzrinne, A. Rao, R. Lanphier Page 5
Invitation of a media server to a conference: Invitation of a media server to a conference:
A media server can be ``invited'' to join an existing A media server can be ``invited'' to join an existing
conference, either to play back media into the presentation or conference, either to play back media into the presentation or
to record all or a subset of the media in a presentation. This to record all or a subset of the media in a presentation. This
mode is useful for distributed teaching applications. Several mode is useful for distributed teaching applications. Several
parties in the conference may take turns ``pushing the remote parties in the conference may take turns ``pushing the remote
control buttons''. control buttons''.
Addition of media to an existing presentation: Addition of media to an existing presentation:
Particularly for live presentations, it is useful if the server Particularly for live presentations, it is useful if the server
can tell the client 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.
H. Schulzrinne, A. Rao, R. Lanphier Page 6
1.2 Requirements 1.2 Requirements
The key words ``MUST'', ``MUST NOT'', ``REQUIRED'', ``SHALL'', ``SHALL The key words ``MUST'', ``MUST NOT'', ``REQUIRED'', ``SHALL'', ``SHALL
NOT'', ``SHOULD'', ``SHOULD NOT'', ``RECOMMENDED'', ``MAY'', and NOT'', ``SHOULD'', ``SHOULD NOT'', ``RECOMMENDED'', ``MAY'', and
``OPTIONAL'' in this document are to be interpreted as described in ``OPTIONAL'' in this document are to be interpreted as described in
RFC 2119 [4]. RFC 2119 [4].
1.3 Terminology 1.3 Terminology
Some of the terminology has been adopted from HTTP/1.1 [5]. Terms not Some of the terminology has been adopted from HTTP/1.1 [5]. Terms not
listed here are defined as in HTTP/1.1. listed here are defined as in HTTP/1.1.
Aggregate control:
The control of the multiple streams using a single timeline by
the server. For audio/video feeds, this means that the client
may issue a single play or pause message to control both the
audio and video feeds.
Conference: Conference:
a multiparty, multimedia presentation, where ``multi'' implies a multiparty, multimedia presentation, where ``multi'' implies
greater than or equal to one. greater than or equal to one.
Client: Client:
The client requests continuous media data from the media The client requests continuous media data from the media
server. server.
Connection: Connection:
A transport layer virtual circuit established between two A transport layer virtual circuit established between two
programs for the purpose of communication. programs for the purpose of communication.
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Continuous media: Continuous media:
Data where there is a timing relationship between source and Data where there is a timing relationship between source and
sink, that is, the sink must reproduce the timing relationshop sink, that is, the sink must reproduce the timing relationshop
that existed at the source. The most common examples of that existed at the source. The most common examples of
continuous media are audio and motion video. Continuous media continuous media are audio and motion video. Continuous media
can be realtime (interactive), where there is a ``tight'' can be realtime (interactive), where there is a ``tight''
timing relationship between source and sink, or streaming timing relationship between source and sink, or streaming
(playback), where the relationship is less strict. (playback), where the relationship is less strict.
Participant: Media initialization:
Participants are members of conferences. A participant may be a Datatype/codec specific initialization. This includes such
machine, e.g., a media record or playback server. things as clockrates, color tables, etc. Any
transport-independent information which is required by a client
for playback of a media stream occurs in the media
initialization phase of stream setup.
H. Schulzrinne, A. Rao, R. Lanphier Page 7
Media parameter:
Parameter specific to a media type that may be changed while
the stream is being played or prior to it.
Media server: Media server:
The network entity providing playback or recording services for The network entity providing playback or recording services for
one or more media streams. Different media streams within a one or more media streams. Different media streams within a
presentation may originate from different media servers. A presentation may originate from different media servers. A
media server may reside on the same or a different host as the media server may reside on the same or a different host as the
web server the presentation is invoked from. web server the presentation is invoked from.
Media parameter: Media server indirection:
Parameter specific to a media type that may be changed while Redirection of a media client to a different media server.
the stream is being played or prior to it.
(Media) stream: (Media) stream:
A single media instance, e.g., an audio stream or a video A single media instance, e.g., an audio stream or a video
stream as well as a single whiteboard or shared application stream as well as a single whiteboard or shared application
group. When using RTP, a stream consists of all RTP and RTCP group. When using RTP, a stream consists of all RTP and RTCP
packets created by a source within an RTP session. This is packets created by a source within an RTP session. This is
equivalent to the definition of a DSM-CC stream([18]). equivalent to the definition of a DSM-CC stream([19]).
Message: Message:
The basic unit of RTSP communication, consisting of a The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in structured sequence of octets matching the syntax defined in
Section 15 and transmitted via a connection or a connectionless Section 15 and transmitted via a connection or a connectionless
protocol. protocol.
Participant:
Participants are members of conferences. A participant may be a
machine, e.g., a media record or playback server.
Presentation: Presentation:
A set of one or more streams which the server allows the client A set of one or more streams presented to the client as a
to manipulate together. A presentation has a single time axis complete media feed, using a presentation description as
for all streams belonging to it. Presentations are defined by defined below. In most cases in the RTSP context, this implies
presentation descriptions (see below). A presentation aggregate control of those streams, but doesn't have to.
description contains RTSP URIs that define which streams can be
controlled individually and an RTSP URI to control the whole
presentation. A movie or live concert consisting of one or more
audio and video streams is an example of a presentation.
H. Schulzrinne, A. Rao, R. Lanphier Page 7
Presentation description: Presentation description:
A presentation description contains information about one or A presentation description contains information about one or
more media streams within a presentation, such as the set of more media streams within a presentation, such as the set of
encodings, network addresses and information about the content. encodings, network addresses and information about the content.
Other IETF protocols such as SDP [6] use the term ``session'' Other IETF protocols such as SDP [6] use the term ``session''
for a live presentation. The presentation description may take for a live presentation. The presentation description may take
several different formats, including but not limited to the several different formats, including but not limited to the
session description format SDP. session description format SDP.
H. Schulzrinne, A. Rao, R. Lanphier Page 8
Response: Response:
An RTSP response. If an HTTP response is meant, that is An RTSP response. If an HTTP response is meant, that is
indicated explicitly. indicated explicitly.
Request: Request:
An RTSP request. If an HTTP request is meant, that is indicated An RTSP request. If an HTTP request is meant, that is indicated
explicitly. explicitly.
RTSP session: RTSP session:
A complete RTSP ``transaction'', e.g., the viewing of a movie. A complete RTSP ``transaction'', e.g., the viewing of a movie.
A session typically consists of a client setting up a transport A session typically consists of a client setting up a transport
mechanism for the continuous media stream (SETUP), starting the mechanism for the continuous media stream (SETUP), starting the
stream with PLAY or RECORD and closing the stream with stream with PLAY or RECORD and closing the stream with
TEARDOWN. TEARDOWN.
Transport initialization:
The negotiation of transport information (i.e. port numbers,
transport protocols, etc) between the client and the server.
1.4 Protocol Properties 1.4 Protocol Properties
RTSP has the following properties: RTSP has the following properties:
Extendable: Extendable:
New methods and parameters can be easily added to RTSP. New methods and parameters can be easily added to RTSP.
Easy to parse: Easy to parse:
RTSP can be parsed by standard HTTP or MIME parsers. RTSP can be parsed by standard HTTP or MIME parsers.
skipping to change at line 368 skipping to change at line 415
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, Section 11.1] and authentication mechanisms such as basic [5, Section 11.1] and
digest authentication [8] are directly applicable. digest authentication [8] are directly applicable.
Transport-independent: Transport-independent:
RTSP may use either an unreliable datagram protocol (UDP) [9], RTSP may use either an unreliable datagram protocol (UDP) [9],
a reliable datagram protocol (RDP, not widely used [10]) or a a reliable datagram protocol (RDP, not widely used [10]) or a
reliable stream protocol such as TCP [11] as it implements reliable stream protocol such as TCP [11] as it implements
application-level reliability. application-level reliability.
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Multi-server capable: Multi-server capable:
Each media stream within a presentation can reside on a Each media stream within a presentation can reside on a
different server. The client automatically establishes several different server. The client automatically establishes several
concurrent control sessions with the different media servers. concurrent control sessions with the different media servers.
Media synchronization is performed at the transport level. Media synchronization is performed at the transport level.
H. Schulzrinne, A. Rao, R. Lanphier Page 9
Control of recording devices: Control of recording devices:
The protocol can control both recording and playback devices, The protocol can control both recording and playback devices,
as well as devices that can alternate between the two modes as well as devices that can alternate between the two modes
(``VCR''). (``VCR'').
Separation of stream control and conference initiation: Separation of stream control and conference initiation:
Stream control is divorced from inviting a media server to a Stream control is divorced from inviting a media server to a
conference. The only requirement is that the conference conference. The only requirement is that the conference
initiation protocol either provides or can be used to create a initiation protocol either provides or can be used to create a
unique conference identifier. In particular, SIP [12] or H.323 unique conference identifier. In particular, SIP [12] or H.323
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Proxy and firewall friendly: Proxy and firewall friendly:
The protocol should be readily handled by both application and The protocol should be readily handled by both application and
transport-layer (SOCKS [13]) firewalls. A firewall may need to transport-layer (SOCKS [13]) firewalls. A firewall may need to
understand the SETUP method to open a ``hole'' for the UDP understand the SETUP method to open a ``hole'' for the UDP
media stream. media stream.
HTTP-friendly: HTTP-friendly:
Where sensible, RTSP re-uses HTTP concepts, so that the Where sensible, RTSP re-uses HTTP concepts, so that the
existing infrastructure can be re-used. This infrastructure existing infrastructure can be re-used. This infrastructure
includes PICS (Platform for Internet Content Selection [20]) includes PICS (Platform for Internet Content Selection [21])
for associating labels with content. However, RTSP does not for associating labels with content. However, RTSP does not
just add methods to HTTP, since the controlling continuous just add methods to HTTP, since the controlling continuous
media requires server state in most cases. media requires server state in most cases.
Appropriate server control: Appropriate server control:
If a client can start a stream, it must be able to stop a If a client can start a stream, it must be able to stop a
stream. Servers should not start streaming to clients in such a stream. Servers should not start streaming to clients in such a
way that clients cannot stop the stream. way that clients cannot stop the stream.
H. Schulzrinne, A. Rao, R. Lanphier Page 9
Transport negotiation: Transport negotiation:
The client can negotiate the transport method prior to actually The client can negotiate the transport method prior to actually
needing to process a continuous media stream. needing to process a continuous media stream.
Capability negotiation: Capability negotiation:
If basic features are disabled, there must be some clean If basic features are disabled, there must be some clean
mechanism for the client to determine which methods are not mechanism for the client to determine which methods are not
going to be implemented. This allows clients to present the going to be implemented. This allows clients to present the
appropriate user interface. For example, if seeking is not appropriate user interface. For example, if seeking is not
allowed, the user interface must be able to disallow moving a allowed, the user interface must be able to disallow moving a
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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 across the methods described in [H19.6] are not likely to be supported 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:
* Existing methods can be extended with new parameters, as long as * Existing methods can be extended with new parameters, as long as
these parameters can be safely ignored by the recipient. (This is these parameters can be safely ignored by the recipient. (This is
equivalent to adding new parameters to an HTML tag.) equivalent to adding new parameters to an HTML tag.) If the client
needs negative acknowledgement when a method extension is not
supported, a tag corresponding to the extension may be added in
the Require: field (see Section 12.32).
* New methods can be added. If the recipient of the message does not * New methods can be added. If the recipient of the message does not
understand the request, it responds with error code 501 (Not understand the request, it responds with error code 501 (Not
implemented) and the sender should not attempt to use this method implemented) and the sender should not attempt to use this method
again. A client may also use the OPTIONS method to inquire about again. A client may also use the OPTIONS method to inquire about
methods supported by the server. The server SHOULD list the methods supported by the server. The server SHOULD list the
methods it supports using the Public response header. methods it supports using the Public response header.
* A new version of the protocol can be defined, allowing almost all * A new version of the protocol can be defined, allowing almost all
aspects (except the position of the protocol version number) to aspects (except the position of the protocol version number) to
change. change.
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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]. URLs may refer to a stream or an aggregate of streams ie. a URLs [17]. URLs may refer to a stream or an aggregate of streams ie. a
presentation. Accordingly, requests described in Section 10 can apply presentation. Accordingly, requests described in Section 10 can apply
to either the whole presentation or an individual stream within the to either the whole presentation or an individual stream within the
presentation. Note that some request methods can only be applied to presentation. Note that some request methods can only be applied to
streams, not presentations and vice versa. streams, not presentations and vice versa.
For example, the RTSP URL For example, the RTSP URL
rtsp://media.example.com:554/twister/audiotrack rtsp://media.example.com:554/twister/audiotrack
identifies the audio stream within the presentation ``twister'', which identifies the audio stream within the presentation ``twister'', which
can be controlled via RTSP requests issued over a TCP connection to can be controlled via RTSP requests issued over a TCP connection to
skipping to change at line 688 skipping to change at line 737
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 sessions to Conference identifiers are used to allow to allow RTSP sessions to
obtain parameters from multimedia conferences the media server is obtain parameters from multimedia conferences the media server is
participating in. These conferences are created by protocols participating in. These conferences are created by protocols
outside the scope of this specification, e.g., H.323 [17] or SIP outside the scope of this specification, e.g., H.323 [18] or SIP
[12]. Instead of the RTSP client explicitly providing transport [12]. Instead of the RTSP client explicitly providing transport
information, for example, it asks the media server to use the information, for example, it asks the media server to use the
values in the conference description instead. If the conference values in the 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 party, the conference identifier which is unique for that inviting party, the
media server could add an internal identifier for that party, e.g., media server could add an internal identifier for that party, e.g.,
its Internet address. However, this would prevent that the its Internet address. However, this would prevent that the
conference participant and the initiator of the RTSP commands are conference participant and the initiator of the RTSP commands are
two different entities. two different entities.
skipping to change at line 714 skipping to change at line 763
more difficult. (See Section 16). more difficult. (See Section 16).
session-id = 1*OCTET ; LWS must be URL-escaped session-id = 1*OCTET ; LWS must be URL-escaped
3.5 SMPTE Relative Timestamps 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.subframes, with the origin at the start 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 of the clip. RTSP uses the ``SMPTE 30 drop'' format. The frame rate is
is handled by dropping the first two frame indices (values 00 and 01) 29.97 frames per second. The ``frames'' field in the time value can
of every minute, except every tenth minute. If the frame value is assume the values 0 through 29. The difference between 30 and 29.97
zero, it may be omitted. Subframes are measured in one-hundredth of a frames per second is handled by dropping the first two frame indices
frame. (values 00 and 01) of every minute, except every tenth minute. If the
frame value is zero, it may be omitted. Subframes are measured in
one-hundredth of a frame.
smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ] smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ]
smpte-time = 2DIGIT ":" 2DIGIT ":" 2DIGIT [ ":" 2DIGIT ] [ "." 2DIGIT] smpte-time = 2DIGIT ":" 2DIGIT ":" 2DIGIT [ ":" 2DIGIT ] [ "." 2DIGIT]
Examples: Examples:
smpte=10:12:33:20- smpte=10:12:33:20-
smpte=10:07:33- smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01 smpte=10:07:00-10:07:33:05.01
3.6 Normal Play Time 3.6 Normal Play Time
Normal play time (NPT) indicates the stream absolute position relative Normal play time (NPT) indicates the stream absolute position relative
to the beginning of the presentation, measured in seconds and to the beginning of the presentation. The timestamp consists is a
microseconds. The beginning of a presentation corresponds to 0 seconds decimal fraction. The part left of the decimal may be expressed in
and 0 microseconds. Negative values are not defined. The microsecond either seconds or hours, minutes and seconds. The part right of the
field is always less than 1,000,000. NPT is defined as in DSM-CC [18]: decimal point measures fractions of a second.
``Intuitively, NPT is the clock the viewer associates with a program.
It is often digitally displayed on a VCR. NPT advances normally when
in normal play mode (scale = 1), advances at a faster rate when in
fast scan forward (high positive scale ratio), decrements when in scan
reverse (high negative scale ratio) and is fixed in pause mode. NPT is
(logically) equivalent to SMPTE time codes.'' [18]
npt-range = "npt" "=" npt-time "-" [ npt-time ] The beginning of a presentation corresponds to 0.0 seconds. Negative
npt-time = 1*DIGIT [ ":" *DIGIT ] values are not defined. The special constant now is defined as the
current instant of a live event. It may be used only for live events.
NPT is defined as in DSM-CC: ``Intuitively, NPT is the clock the
viewer associates with a program. It is often digitally displayed on a
VCR. NPT advances normally when in normal play mode (scale = 1),
advances at a faster rate when in fast scan forward (high positive
scale ratio), decrements when in scan reverse (high negative scale
ratio) and is fixed in pause mode. NPT is (logically) equivalent to
SMPTE time codes.'' [19]
npt-time = "now" | npt-sec | npt-hhmmss
npt-sec = 1*DIGIT [ "." *DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT]
npt-hh = 1*DIGIT ; any positive number
npt-mm = 2DIGIT ; 00-59
npt-ss = 2DIGIT ; 00-59
Examples: Examples:
npt=123:45-125 npt=123.45-125
npt=12:05:35.3
npt=now
The syntax conforms to ISO 8601. The npt-sec notation is optimized
for automatic generation, the ntp-hhmmss notation for consumption
by human readers. The ``now'' constant allows clients to request to
receive the live feed rather than the stored or time-delayed
version. This is needed since neither absolute time, nor zero time
are appropriate for this case.
3.7 Absolute Time 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
4 RTSP Message 4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set RTSP is a text-based protocol and uses the ISO 10646 character set
in UTF-8 encoding (RFC 2044). Lines are terminated by CRLF, but in UTF-8 encoding (RFC 2044). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by themselves receivers should be prepared to also interpret CR and LF by themselves
as line terminators. as line terminators.
Text-based protocols make it easier to add optional parameters in a Text-based protocols make it easier to add optional parameters in a
self-describing manner. Since the number of parameters and the self-describing manner. Since the number of parameters and the
frequency of commands is low, processing efficiency is not a frequency of commands is low, processing efficiency is not a
skipping to change at line 811 skipping to change at line 879
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. 1. Any response message which MUST NOT include a message-body
Any response message which MUST NOT include a message-body
(such as the 1xx, 204, and 304 responses) is always terminated (such as the 1xx, 204, and 304 responses) is always terminated
by the first empty line after the header fields, regardless of by the first empty line after the header fields, regardless of
the entity-header fields present in the message. (Note: An the entity-header fields present in the message. (Note: An
empty line consists of only CRLF.) empty line consists of only CRLF.)
2. 2. If a Content-Length header field (section 12.14) is present,
If a Content-Length header field (section 12.15) is present,
its value in bytes represents the length of the message-body. its value in bytes represents the length of the message-body.
If this header field is not present, a value of zero is If this header field is not present, a value of zero is
assumed. assumed.
3. 3. By the server closing the connection. (Closing the connection
By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a that would leave no possibility for the server to send back a
response.) response.)
Note that RTSP does not (at present) support the HTTP/1.1 ``chunked'' Note that RTSP does not (at present) support the HTTP/1.1 ``chunked''
transfer coding(see [H3.6]) and requires the presence of the transfer coding(see [H3.6]) and requires the presence of the
Content-Length header field. Content-Length header field.
Given the moderate length of presentation descriptions returned, Given the moderate length of presentation descriptions returned,
the server should always be able to determine its length, even if the server should always be able to determine its length, even if
it is generated dynamically, making the chunked transfer encoding it is generated dynamically, making the chunked transfer encoding
unnecessary. Even though Content-Length must be present if there is unnecessary. Even though Content-Length must be present if there is
any entity body, the rules ensure reasonable behavior even if the any entity body, the rules ensure reasonable behavior even if the
length is not given explicitly. length is not given explicitly.
5 General Header Fields 5 General Header Fields
See [H4.5], except that Pragma, Transfer-Encoding and Upgrade See [H4.5], except that Pragma, Transfer-Encoding and Upgrade
headers are not defined: headers are not defined:
general-header = Cache-Control ; Section 12.10 general-header = Cache-Control ; Section 12.8
| Connection ; Section 12.12 | Connection ; Section 12.10
| Date ; Section 12.17 | Date ; Section 12.18
| Via ; Section 12.39 | Via ; Section 12.43
6 Request 6 Request
A request message from a client to a server or vice versa includes, A request message from a client to a server or vice versa includes,
within the first line of that message, the method to be applied to the within the first line of that message, the method to be applied to the
resource, the identifier of the resource, and the protocol version in resource, the identifier of the resource, and the protocol version in
use. use.
Request = Request-Line ; Section 6.1 Request = Request-Line ; Section 6.1
*( general-header ; Section 5 *( general-header ; Section 5
| request-header ; Section 6.2 | request-header ; Section 6.2
| entity-header ) ; Section 8.1 | entity-header ) ; Section 8.1
CRLF CRLF
[ message-body ] ; Section 4.3 [ message-body ] ; Section 4.3
6.1 Request Line 6.1 Request Line
Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no CRLF Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Method = "DESCRIBE" ; Section 10.2 Method = "DESCRIBE" ; Section 10.2
| "ANNOUNCE" ; Section 10.3 | "ANNOUNCE" ; Section 10.3
| "GET_PARAMETER" ; Section 10.8 | "GET_PARAMETER" ; Section 10.8
| "OPTIONS" ; Section 10.1 | "OPTIONS" ; Section 10.1
| "PAUSE" ; Section 10.6 | "PAUSE" ; Section 10.6
| "PLAY" ; Section 10.5 | "PLAY" ; Section 10.5
| "RECORD" ; Section 10.11 | "RECORD" ; Section 10.11
| "REDIRECT" ; Section 10.10 | "REDIRECT" ; Section 10.10
| "SETUP" ; Section 10.4 | "SETUP" ; Section 10.4
skipping to change at line 879 skipping to change at line 944
| "GET_PARAMETER" ; Section 10.8 | "GET_PARAMETER" ; Section 10.8
| "OPTIONS" ; Section 10.1 | "OPTIONS" ; Section 10.1
| "PAUSE" ; Section 10.6 | "PAUSE" ; Section 10.6
| "PLAY" ; Section 10.5 | "PLAY" ; Section 10.5
| "RECORD" ; Section 10.11 | "RECORD" ; Section 10.11
| "REDIRECT" ; Section 10.10 | "REDIRECT" ; Section 10.10
| "SETUP" ; Section 10.4 | "SETUP" ; Section 10.4
| "SET_PARAMETER" ; Section 10.9 | "SET_PARAMETER" ; Section 10.9
| "TEARDOWN" ; Section 10.7 | "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
6.2 Request Header Fields 6.2 Request Header Fields
request-header = Accept ; Section 12.1 request-header = Accept ; Section 12.1
| Accept-Encoding ; Section 12.2 | Accept-Encoding ; Section 12.2
| Accept-Language ; Section 12.3 | Accept-Language ; Section 12.3
| Authorization ; Section 12.5 | Authorization ; Section 12.5
| From ; Section 12.19 | From ; Section 12.20
| If-Modified-Since ; Section 12.21 | If-Modified-Since ; Section 12.23
| Range ; Section 12.28 | Range ; Section 12.29
| Referer ; Section 12.29 | Referer ; Section 12.30
| User-Agent ; Section 12.37 | User-Agent ; Section 12.41
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 HTTP/1.1 requires servers to understand the absolute URL, but
clients are supposed to use the Host request header. This is purely clients are supposed to use the Host request header. This is purely
needed for backward-compatibility with HTTP/1.0 servers, a needed for backward-compatibility with HTTP/1.0 servers, a
consideration that does not apply to RTSP. consideration that does not apply to RTSP.
The asterisk "*" in the Request-URI means that the request does not The asterisk "*" in the Request-URI means that the request does not
skipping to change at line 934 skipping to change at line 997
Response = Status-Line ; Section 7.1 Response = Status-Line ; Section 7.1
*( general-header ; Section 5 *( general-header ; Section 5
| response-header ; Section 7.1.2 | response-header ; Section 7.1.2
| entity-header ) ; Section 8.1 | entity-header ) ; Section 8.1
CRLF CRLF
[ message-body ] ; Section 4.3 [ message-body ] ; Section 4.3
7.1 Status-Line 7.1 Status-Line
The first line of a Response message is the Status-Line, consisting The first line of a Response message is the Status-Line, consisting of
of the protocol version followed by a numeric status code, the the protocol version followed by a numeric status code, and the
sequence number of the corresponding request and the textual phrase textual phrase associated with the status code, with each element
associated with the status code, with each element separated by SP separated by SP characters. No CR or LF is allowed except in the final
characters. No CR or LF is allowed except in the final CRLF sequence. CRLF sequence.
Status-Line = RTSP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
7.1.1 Status Code and Reason Phrase 7.1.1 Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully attempt to understand and satisfy the request. These codes are fully
defined in section11. 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 user. The client is not required to examine or display the
Reason-Phrase. Reason-Phrase.
skipping to change at line 1002 skipping to change at line 1064
| "409" ; Conflict | "409" ; Conflict
| "410" ; Gone | "410" ; Gone
| "411" ; Length Required | "411" ; Length Required
| "412" ; Precondition Failed | "412" ; Precondition Failed
| "413" ; Request Entity Too Large | "413" ; Request Entity Too Large
| "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 | "454" ; Session Not Found
| "45x" ; Method Not Valid in This State | "455" ; Method Not Valid in This State
| "45x" ; Header Field Not Valid for Resource | "456" ; Header Field Not Valid for Resource
| "45x" ; Invalid Range | "457" ; Invalid Range
| "45x" ; Parameter Is Read-Only | "458" ; Parameter Is Read-Only
| "45x" ; Aggregate operation not allowed | "459" ; Aggregate operation not allowed
| "45x" ; Only aggregate operation allowed | "460" ; 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" ; RTSP 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>
skipping to change at line 1034 skipping to change at line 1096
digit, and treat any unrecognized response as being equivalent to the digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an unrecognized x00 status code of that class, with the exception that an unrecognized
response MUST NOT be cached. For example, if an unrecognized status response MUST NOT be cached. For example, if an unrecognized status
code of 431 is received by the client, it can safely assume that there code of 431 is received by the client, it can safely assume that there
was something wrong with its request and treat the response as if it was something wrong with its request and treat the response as if it
had received a 400 status code. In such cases, user agents SHOULD had received a 400 status code. In such cases, user agents SHOULD
present to the user the entity returned with the response, since that present to the user the entity returned with the response, since that
entity is likely to include human-readable information which will entity is likely to include human-readable information which will
explain the unusual status. explain the unusual status.
Code Reason Code reason
100 Continue all 100 Continue all
200 OK all 200 OK all
201 Created RECORD 201 Created RECORD
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
skipping to change at line 1059 skipping to change at line 1124
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 RECORD 409 Conflict RECORD
410 Gone all 410 Gone all
411 Length Required SETUP 411 Length Required SETUP
412 Precondition Failed DESCRIBE, SETUP 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 Session not found all 451 Invalid parameter SETUP
45x Invalid parameter SETUP 452 Illegal Conference Identifier SETUP
45x Not Enough Bandwidth SETUP 453 Not Enough Bandwidth SETUP
45x Illegal Conference Identifier SETUP 454 Session not found all
45x Illegal Session Identifier PLAY, RECORD, TEARDOWN 455 Method Not Valid In This State all
45x Parameter Is Read-Only SET_PARAMETER 456 Header Field Not Valid all
45x Header Field Not Valid all 457 Invalid Range PLAY
45x Method Not Valid In This State all 458 Parameter Is Read-Only SET_PARAMETER
45x Aggregate operation not allowed all 459 Aggregate operation not allowed all
45x Only aggregate operation allowed all 460 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 Status codes and their usage with RTSP methods
7.1.2 Response Header Fields 7.1.2 Response Header Fields
The response-header fields allow the request recipient to pass The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the server the Status-Line. These header fields give information about the server
and about further access to the resource identified by the and about further access to the resource identified by the
Request-URI. Request-URI.
response-header = Location ; Section 12.23 response-header = Location ; Section 12.25
| Proxy-Authenticate ; Section 12.26 | Proxy-Authenticate ; Section 12.26
| Public ; Section 12.27 | Public ; Section 12.28
| Retry-After ; Section 12.30 | Retry-After ; Section 12.31
| Server ; Section 12.33 | Server ; Section 12.36
| Vary ; Section 12.38 | Vary ; Section 12.42
| WWW-Authenticate ; Section 12.40 | WWW-Authenticate ; Section 12.44
Response-header field names can be extended reliably only in Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of experimental header fields MAY be given the semantics of
response-header fields if all parties in the communication recognize response-header fields if all parties in the communication recognize
them to be response-header fields. Unrecognized header fields are them to be response-header fields. Unrecognized header fields are
treated as entity-header fields. treated as entity-header fields.
8 Entity 8 Entity
skipping to change at line 1117 skipping to change at line 1184
In this section, both sender and recipient refer to either the client In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity. or the server, depending on who sends and who receives the entity.
8.1 Entity Header Fields 8.1 Entity Header Fields
Entity-header fields define optional metainformation about the Entity-header fields define optional metainformation about the
entity-body or, if no body is present, about the resource identified entity-body or, if no body is present, about the resource identified
by the request. by the request.
entity-header = Allow ; Section 12.4 entity-header = Allow ; Section 12.4
| Content-Encoding ; Section 12.13 | Content-Base ; Section 12.11
| Content-Language ; Section 12.14 | Content-Encoding ; Section 12.12
| Content-Length ; Section 12.15 | Content-Language ; Section 12.13
| Content-Length ; Section 12.14
| Content-Location ; Section 12.15
| Content-Type ; Section 12.16 | Content-Type ; Section 12.16
| Expires ; Section 12.18 | Expires ; Section 12.19
| Last-Modified ; Section 12.22 | Last-Modified ; Section 12.24
| 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.
8.2 Entity Body 8.2 Entity Body
See [H7.2] See [H7.2]
9 Connections 9 Connections
skipping to change at line 1187 skipping to change at line 1255
probably a bad idea to have two reliability mechanisms on top of probably a bad idea to have two reliability mechanisms on top of
each other, although the RTSP RTT estimate is likely to be larger each other, although the RTSP RTT estimate is likely to be larger
than the TCP estimate. than 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 lack for each request transmitted. If a request is repeated because of lack
of acknowledgement, the sequence number is incremented. of acknowledgement, the sequence number is incremented.
This avoids ambiguities when computing round-trip time estimates. This avoids ambiguities when computing round-trip time estimates.
[TBD: An initial sequence number negotiation needs to be added for
UDP; otherwise, a new stream connection may see a request be
acknowledged by a delayed response from an earlier ``connection''.
This handshake can be avoided with a sequence number containing a
timestamp of sufficiently high resolution.]
The reliability mechanism described here does not protect against 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.]
skipping to change at line 1236 skipping to change at line 1298
ANNOUNCE C->S, S->C P,S optional ANNOUNCE C->S, S->C P,S optional
GET_PARAMETER C->S, S->C P,S optional GET_PARAMETER C->S, S->C P,S optional
OPTIONS C->S 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: Overview of RTSP methods, their direction, and what objects (P:
presentation, S: stream) they operate on presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommended, but not required in that a Notes on Table 2: PAUSE is recommended, but not required in that a
fully functional server can be built that does not support this fully functional server can be built that does not support this
method, for example, for live feeds. If a server does not support a method, for example, for live feeds. If a server does not support a
particular method, it MUST return "501 Not Implemented" and a client particular method, it MUST return "501 Not Implemented" and a client
SHOULD not try this method again for this server. SHOULD not try this method again for this server.
10.1 OPTIONS 10.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An OPTIONS The behavior is equivalent to that described in [H9.2]. An OPTIONS
request may be issued at any time, e.g., if the client is about to try request may be issued at any time, e.g., if the client is about to try
a non-standard request. It does not influence server state. a non-standard request. It does not influence server state.
Example : Example :
C->S: OPTIONS * RTSP/1.0 1 C->S: OPTIONS * RTSP/1.0
PEP: {{map "http://www.iana.org/rtsp/implicit-play"}} CSeq: 1
{{map "http://www.iana.org/rtsp/record-feature"}} Require: implicit-play
C-PEP: {{map "http://www.iana.org/rtsp/udp-control"}} Proxy-Require: gzipped-messages
{{map "http://www.iana.org/rtsp/gzipped-messages"}}
S->C: RTSP/1.0 200 2 OK S->C: RTSP/1.0 200 OK
PEP-Info: {{map "http://www.iana.org/rtsp/implicit-play"} CSeq: 1
{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 Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are necessarily fictional features (one would hope
Note that these are fictional features (though we may want to make that we would not purposefully overlook a truly useful feature just so
them real one day). that we could have a strong example in this section).
DESCRIBE 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. resource.
The DESCRIBE reply-response pair constitutes the media initialization
phase of RTSP.
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
CSeq: 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 OK
CSeq: 312
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
v=0 v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4 o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar s=SDP Seminar
i=A Seminar on the session description protocol i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley) e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127 c=IN IP4 224.2.17.12/127
t=2873397496 2873404696 t=2873397496 2873404696
a=recvonly a=recvonly
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
skipping to change at line 1303 skipping to change at line 1363
i=A Seminar on the session description protocol i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley) e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127 c=IN IP4 224.2.17.12/127
t=2873397496 2873404696 t=2873397496 2873404696
a=recvonly a=recvonly
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31 m=video 2232 RTP/AVP 31
m=whiteboard 32416 UDP WB m=whiteboard 32416 UDP WB
a=orient:portrait a=orient:portrait
The DESCRIBE response MUST contain all media initialization
information for the resource(s) that it describes. If a media client
obtains a presentation description from a source other than DESCRIBE
and that description contains a complete set of media initialization
parameters, the client SHOULD use those parameters and not then
request a description for the same media via RTSP.
Additionally, servers SHOULD NOT use the DESCRIBE response as a means
of media indirection.
Clear ground rules need to be established so that clients have an
unambiguous means of knowing when to request media initialization
information via DESCRIBE, and when not to. By forcing a DESCRIBE
response to contain all media initialization for the set of streams
that it describes, and discouraging use of DESCRIBE for media
indirection, we avoid looping problems that might result from other
approaches.
ANNOUNCE ANNOUNCE
The ANNOUNCE method serves two purposes: The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a When sent from client to server, ANNOUNCE posts the description of a
presentation or media object identified by the request URL to a presentation or media object identified by the request URL to a
server. server. When sent from server to client, ANNOUNCE updates the session
When sent from server to client, ANNOUNCE updates the session
description in real-time. description in real-time.
If a new media stream is added to a presentation (e.g., during a live If a new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent presentation), the whole presentation description should be sent
again, rather than just the additional components, so that components again, rather than just the additional components, so that components
can be deleted. can be deleted.
Example: Example:
C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0 312 C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Session: 4711
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 332 Content-Length: 332
v=0 v=0
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4 o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
s=SDP Seminar s=SDP Seminar
i=A Seminar on the session description protocol i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley) e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127 c=IN IP4 224.2.17.12/127
t=2873397496 2873404696 t=2873397496 2873404696
a=recvonly a=recvonly
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
skipping to change at line 1338 skipping to change at line 1416
s=SDP Seminar s=SDP Seminar
i=A Seminar on the session description protocol i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley) e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127 c=IN IP4 224.2.17.12/127
t=2873397496 2873404696 t=2873397496 2873404696
a=recvonly a=recvonly
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31 m=video 2232 RTP/AVP 31
S->C: RTSP/1.0 200 312 OK S->C: RTSP/1.0 200 OK
CSeq: 312
SETUP SETUP
The SETUP request for a URI specifies the transport mechanism to be The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. A client can issue a SETUP request for a used for the streamed media. A client can issue a SETUP request for a
stream that is already playing to change transport parameters, which a stream that is already playing to change transport parameters, which a
server MAY allow(If it does not allow it, it must respond with error server MAY allow(If it does not allow it, it must respond with error
``45x Method not valid in this state'' ). For the benefit of any ``455 Method not valid in this state'' ). For the benefit of any
intervening firewalls, a client must indicate the transport parameters intervening firewalls, a client must indicate the transport parameters
even if it has no influence over these parameters, for example, where even if it has no influence over these parameters, for example, where
the server advertises a fixed multicast address. the server advertises a fixed multicast address.
Segregating content desciption into a DESCRIBE message and Segregating content desciption into a DESCRIBE message and
transport information in SETUP avoids having firewall to parse transport information in SETUP avoids having firewall to parse
numerous different presentation description formats for information numerous different presentation description formats for information
which is irrelevant to transport. which is irrelevant to transport.
The Transport header specifies the transport parameters acceptable to The Transport header specifies the transport parameters acceptable to
the client for data transmission; the response will contain the the client for data transmission; the response will contain the
transport parameters selected by the server. transport parameters selected by the server.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 302 C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Transport: RTP/AVP;port=4588 Transport: RTP/AVP;port=4588
S->C: RTSP/1.0 200 302 OK S->C: RTSP/1.0 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Transport: RTP/AVP;port=4588 Transport: RTP/AVP;port=4588
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.
skipping to change at line 1388 skipping to change at line 1469
request arriving while a previous PLAY request is still active is request arriving while a previous PLAY request is still active is
delayed until the first has been completed. delayed until the first has been completed.
This allows precise editing. This allows precise editing.
For example, regardless of how closely spaced the two PLAY commands in For example, regardless of how closely spaced the two PLAY commands in
the example below arrive, the server will play first second 10 through the example below arrive, the server will play first second 10 through
15 and then, immediately following, seconds 20 to 25 and finally 15 and then, immediately following, seconds 20 to 25 and finally
seconds 30 through the end. 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
CSeq: 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
CSeq: 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
CSeq: 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 pause stream has been paused via PAUSE, stream delivery resumes at the pause
point. If a stream is playing, such a PLAY request causes no further point. If a stream is playing, such a PLAY request causes no further
action and can be used by the client to test server liveness. 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
skipping to change at line 1425 skipping to change at line 1508
current position is returned in the reply. The unit of the range in current position is returned in the reply. The unit of 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
CSeq: 833
Range: smpte=0:10:20-;time=19970123T153600Z Range: smpte=0:10:20-;time=19970123T153600Z
S->C: RTSP/1.0 200 833 OK S->C: RTSP/1.0 200 OK
CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Range: smpte=0:10:22-;time=19970123T153600Z Range: smpte=0:10:22-;time=19970123T153600Z
For playing back a recording of a live presentation, it may be For playing back a recording of a live presentation, it may be
desirable to use clock units: desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0 835 C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
CSeq: 835
Range: clock=19961108T142300Z-19961108T143520Z Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 833 OK S->C: RTSP/1.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the npt format A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats. and MAY support the clock and smpte formats.
PAUSE PAUSE
The PAUSE request causes the stream delivery to be interrupted The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only playback (halted) temporarily. If the request URL names a stream, only playback
and recording of that stream is halted. For example, for audio, this and recording of that stream is halted. For example, for audio, this
skipping to change at line 1486 skipping to change at line 1573
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 rtsp://example.com/fizzle/foo RTSP/1.0 834 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834
Session: 1234 Session: 1234
S->C: RTSP/1.0 200 834 OK S->C: RTSP/1.0 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
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 presentation URI 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 the is no longer valid. Unless all transport parameters are defined by 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 rtsp://example.com/fizzle/foo RTSP/1.0 892 C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 892
Session: 1234 Session: 1234
S->C: RTSP/1.0 200 892 OK S->C: RTSP/1.0 200 OK
CSeq: 892
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 rtsp://example.com/fizzle/foo RTSP/1.0 431 S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 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 OK
CSeq: 431
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
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.
skipping to change at line 1563 skipping to change at line 1656
sense to allow the setting of several parameters if an atomic sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client does setting is desirable. Imagine device control where the client does
not want the camera to pan unless it can also tilt to the right not want the camera to pan unless it can also tilt to the right
angle at the same time. angle at the same time.
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 rtsp://example.com/fizzle/foo RTSP/1.0 421 C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 421
Content-type: text/rtsp-parameters Content-type: text/rtsp-parameters
barparam: barstuff barparam: barstuff
S->C: RTSP/1.0 450 421 Invalid Parameter S->C: RTSP/1.0 450 Invalid Parameter
CSeq: 421
Content-Length: 6 Content-Length: 6
barparam barparam
REDIRECT REDIRECT
A redirect request informs the client that it must connect to A redirect request informs the client that it must connect to
another server location. It contains the mandatory header Location, another server location. It contains the mandatory header Location,
which indicates that the client should issue requests 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 redirection may contain the parameter Range, which indicates when the 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 rtsp://example.com/fizzle/foo RTSP/1.0 732 S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 732
Location: rtsp://bigserver.com:8001 Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z- Range: clock=19960213T143205Z-
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. started, commence recording immediately.
skipping to change at line 1608 skipping to change at line 1704
request-URI, the response SHOULD be 201 (Created) and contain an request-URI, the response SHOULD be 201 (Created) and contain an
entity which describes the status of the request and refers to the new entity which describes the status of the request and refers to the new
resource, and a Location header. resource, and a Location header.
A media server supporting recording of live presentations MUST support A media server supporting recording of live presentations MUST 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 rtsp://example.com/meeting/audio.en RTSP/1.0 954 C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
CSeq: 954
Session: 1234 Session: 1234
Conference: 128.16.64.19/32492374 Conference: 128.16.64.19/32492374
10.12 Embedded (Interleaved) Binary Data 10.12 Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server Certain firewall designs and other circumstances may force a server
to interleave RTSP methods and stream data. This interleaving should to interleave RTSP methods and stream data. This interleaving should
generally be avoided unless necessary since it complicates client and generally be avoided unless necessary since it complicates client and
server operation and imposes additional overhead. Interleaved binary server operation and imposes additional overhead. Interleaved binary
data SHOULD only be used if RTSP is carried over TCP. data SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier, followed sign (24 decimal), followed by a one-byte channel identifier, followed
by the length of the encapsulated binary data as a binary, two-byte by the length of the encapsulated binary data as a binary, two-byte
integer in network byte order. The stream data follows immediately integer in network byte order. The stream data follows immediately
afterwards, without a CRLF, but including the upper-layer protocol afterwards, without a CRLF, but including the upper-layer protocol
headers. Each $ block contains exactly one upper-layer protocol data headers. Each $ block contains exactly one upper-layer protocol data
unit, e.g., one RTP packet. unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header 12.35. The channel identifier is defined in the Transport header with the
interleaved parameter 12.39.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 2 C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
Transport: RTP/AVP/TCP;channel=0 CSeq: 2
S->C: RTSP/1.0 200 2 OK Transport: RTP/AVP/TCP;interleaved=0
S->C: RTSP/1.0 200 OK
CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;channel=0 Transport: RTP/AVP/TCP;interleaved=0
Session: 12345 Session: 12345
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0 3 C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
CSeq: 3
Session: 12345 Session: 12345
S->C: RTSP/1.0 200 3 OK S->C: RTSP/1.0 200 OK
CSeq: 3
Session: 12345 Session: 12345
Date: 05 Jun 1997 18:59:15 GMT 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}
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 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
skipping to change at line 1689 skipping to change at line 1792
11.2.3 452 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.
11.2.4 453 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.
11.2.5 45x Session Not Found 11.2.5 454 Session Not Found
The RTSP session identifier is invalid or has timed out. The RTSP session identifier is invalid or has timed out.
11.2.6 45x Method Not Valid in This State 11.2.6 455 Method Not Valid in This State
The client or server cannot process this request in its current state. The client or server cannot process this request in its current state.
11.2.7 45x Header Field Not Valid for Resource 11.2.7 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example, if The server could not act on a required request header. For example, if
PLAY contains the Range header field, but the stream does not allow PLAY contains the Range header field, but the stream does not allow
seeking. seeking.
11.2.8 45x Invalid Range 11.2.8 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the The Range value given is out of bounds, e.g., beyond the end of the
presentation. presentation.
11.2.9 45x Parameter Is Read-Only 11.2.9 458 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.
11.2.10 45x Aggregate operation not allowed 11.2.10 459 Aggregate operation not allowed
The requested method may not be applied on the URL in question since The requested method may not be applied on the URL in question since
it is an aggregate(presentation) URL. The method may be applied on a it is an aggregate(presentation) URL. The method may be applied on a
stream URL. stream URL.
11.2.11 45x Only aggregate operation allowed 11.2.11 460 Only aggregate operation allowed
The requested method may not be applied on the URL in question since The requested method may not be applied on the URL in question since
it is not an aggregate(presentation) URL. The method may be applied on it is not an aggregate(presentation) URL. The method may be applied on
the presentation URL. the presentation URL.
12 Header Field Definitions 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.
skipping to change at line 1745 skipping to change at line 1848
in the column labeled ``support'' MUST be implemented by the recipient in the column labeled ``support'' MUST be implemented by the recipient
for a particular method, while fields marked ``opt.'' are optional. for a particular method, while fields marked ``opt.'' are optional.
Note that not all fields marked 'r' will be send in every request of Note that not all fields marked 'r' will be send in every request of
this type; merely, that client (for response headers) and server (for this type; merely, that client (for response headers) and server (for
request headers) MUST implement them. The last column lists the method request headers) MUST implement them. The last column lists the method
for which this header field is meaningful; the designation ``entity'' for which this header field is meaningful; the designation ``entity''
refers to all methods that return a message body. Within this refers to all methods that return a message body. Within this
specification, DESCRIBE and GET_PARAMETER fall into this class. 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 456 (Header Field Not Valid for Resource).
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. all Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control g opt. SETUP Cache-Control g opt. SETUP
Conference R opt. SETUP Conference R opt. SETUP
Connection g req. all Connection g req. all
Content-Base e opt. entity
Content-Encoding e req. SET_PARAMETER Content-Encoding e req. SET_PARAMETER
Content-Encoding e req. DESCRIBE, ANNOUNCE Content-Encoding e req. DESCRIBE, ANNOUNCE
Content-Language e req. DESCRIBE, ANNOUNCE Content-Language e req. DESCRIBE, ANNOUNCE
Content-Length e req. SET_PARAMETER, ANNOUNCE Content-Length e req. SET_PARAMETER, ANNOUNCE
Content-Length e req. entity Content-Length e req. entity
Content-Location e opt. entity
Content-Type e req. SET_PARAMETER, ANNOUNCE Content-Type e req. SET_PARAMETER, ANNOUNCE
Content-Type r req. entity Content-Type r req. entity
CSeq g req. all
Date g opt. all Date g opt. all
Expires e opt. DESCRIBE, ANNOUNCE Expires e opt. DESCRIBE, ANNOUNCE
From R opt. all From R opt. all
If-Modified-Since R opt. DESCRIBE, SETUP If-Modified-Since R opt. DESCRIBE, SETUP
Last-Modified e opt. entity Last-Modified e opt. entity
Proxy-Authenticate
Proxy-Require R req. all
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
RTP-Info r req. PLAY
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-Info r req. PLAY Unsupported r req. all
User-Agent R opt. all User-Agent R opt. all
Via g opt. all Via g opt. all
WWW-Authenticate r opt. all WWW-Authenticate r opt. all
! Overview of RTSP header fields
Table 3: Overview of RTSP header fields
12.1 Accept 12.1 Accept
The Accept request-header field can be used to specify certain The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the presentation description content types which are acceptable for the
response. response.
The ``level'' parameter for presentation descriptions is properly The ``level'' parameter for presentation descriptions is properly
defined as part of the MIME type registration, not here. defined as part of the MIME type registration, not here.
skipping to change at line 1829 skipping to change at line 1938
Allow: SETUP, PLAY, RECORD, SET_PARAMETER Allow: SETUP, PLAY, RECORD, SET_PARAMETER
12.5 Authorization 12.5 Authorization
See [H14.8] See [H14.8]
12.6 Bandwidth 12.6 Bandwidth
The Bandwidth request header field describes the estimated bandwidth The Bandwidth request header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured available to the client, expressed as a positive integer and measured
in bits per second. in bits per second. The bandwidth available to the client may change
during an RTSP session, e.g., due to modem retraining.
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
12.7 Blocksize 12.7 Blocksize
This request header field is sent from the client to the media This request header field is sent from the client to the media
server asking the server for a particular media packet size. This server asking the server for a particular media packet size. This
packet size does not include lower-layer headers such as IP, UDP, or packet size does not include lower-layer headers such as IP, UDP, or
RTP. The server is free to use a blocksize which is lower than the one 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 override it with multiple of the minimum media-specific block size or override it with
the media specific size if necessary. The block size is a strictly the media specific size if necessary. The block size is a strictly
positive decimal number and measured in octets. The server only positive decimal number and measured in octets. The server only
returns an error (416) if the value is syntactically invalid. returns an error (416) if the value is syntactically invalid.
12.8 C-PEP 12.8 Cache-Control
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.
skipping to change at line 1975 skipping to change at line 2068
must-revalidate: must-revalidate:
When the must-revalidate directive is present in a SETUP When the must-revalidate directive is present in a SETUP
response received by a cache, that cache MUST NOT use the entry response received by a cache, that cache MUST NOT use the entry
after it becomes stale to respond to a subsequent request after it becomes stale to respond to a subsequent request
without first revalidating it with the origin server. (I.e., without first revalidating it with the origin server. (I.e.,
the cache must do an end-to-end revalidation every time, if, the cache must do an end-to-end revalidation every time, if,
based solely on the origin server's Expires, the cached based solely on the origin server's Expires, the cached
response is stale.) response is stale.)
12.11 Conference 12.9 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%20Starr
12.12 Connection A response code of 452 (452 Conference Not Found) is returned if the
conference-id is not valid.
12.10 Connection
See [H14.10]. See [H14.10].
12.13 Content-Encoding TBD: Connection: timeout=<secs>
12.11 Content-Base
See [H14.11]
12.12 Content-Encoding
See [H14.12] See [H14.12]
12.14 Content-Language 12.13 Content-Language
See [H14.13] See [H14.13]
12.15 Content-Length 12.14 Content-Length
This field contains the length of the content of the method (i.e. This field contains the length of the content of the method (i.e.
after the double CRLF following the last header). Unlike HTTP, it MUST after the double CRLF following the last header). Unlike HTTP, it 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].
12.15 Content-Location
See [H14.15]
12.16 Content-Type 12.16 Content-Type
See [H14.18]. Note that the content types suitable for RTSP are See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and likely to be restricted in practice to presentation descriptions and
parameter-value types. parameter-value types.
12.17 Date 12.17 CSeq
This field is a mandatory field that specifies the sequence number
for an RTSP request-response pair. For every RTSP request containing
the given sequence number, there will be a corresponding response
having the same number.
12.18 Date
See [H14.19]. See [H14.19].
12.18 Expires 12.19 Expires
The Expires entity-header field gives the date/time after which the The Expires entity-header field gives a date and time after which
media-stream should be considered stale. A stale cache entry may not the description or media-stream should be considered stale.
normally be returned by a cache (either a proxy cache or an user agent
cache) unless it is first validated with the origin server (or with an The interpretation depends on the method:
intermediate cache that has a fresh copy of the entity). See section
13.2 for further discussion of the expiration model. DESCRIBE response:
The Expires header indicates a date and time after which the
description should be considered stale.
A stale cache entry may not normally be returned by a cache (either a
proxy cache or an user agent cache) unless it is first validated with
the origin server (or with an intermediate cache that has a fresh copy
of the entity). See section 13 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 time. resource will change or cease to exist at, before, or after that time.
The format is an absolute date and time as defined by HTTP-date in The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format: [H3.3]; it MUST be in RFC1123-date format:
Expires = "Expires" ":" HTTP-date Expires = "Expires" ":" HTTP-date
An example of its use is An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats, RTSP/1.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as in the past (i.e., "already especially including the value "0", as in the past (i.e., ``already
expired"). expired'').
To mark a response as "already expired," an origin server should use
an Expires date that is equal to the Date header value.
To mark a response as "never expires," an origin server should use an To mark a response as ``already expired,'' an origin server should use
Expires date approximately one year from the time the response is an Expires date that is equal to the Date header value. To mark a
sent. RTSP/1.0 servers should not send Expires dates more than one response as ``never expires,'' an origin server should use an Expires
year in the future. date approximately one year from the time the response is sent.
RTSP/1.0 servers should not send Expires dates more than one year in
the future.
The presence of an Expires header field with a date value of some time The presence of an Expires header field with a date value of some time
in the future on a media stream that otherwise would by default be in the future on a media stream that otherwise would by default 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 12.10). indicated otherwise by a Cache-Control header field (Section 12.8).
12.19 From 12.20 From
See [H14.22]. See [H14.22].
12.20 Host 12.21 Host
This HTTP request header field is not needed for RTSP. It should be This HTTP request header field is not needed for RTSP. It should be
silently ignored if sent. silently ignored if sent.
12.21 If-Modified-Since 12.22 If-Match
See [H14.25].
This field is especially useful for ensuring the integrity of the
presentation description, in both the case where it is fetched via
means external to RTSP (such as HTTP), or in the case where the server
implementation is guaranteeing the integrety of the description
between the time of the DESCRIBE message and the SETUP message.
The identifier is an opaque identifier, and thus is not specific to
any particular session description language.
12.23 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE 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
12.22 Last-Modified 12.24 Last-Modified
The Last-Modified entity-header field indicates the date and time at The Last-Modified entity-header field indicates the date and time at
which the origin server believes the variant was last modified. See which the origin server believes the entity (presentation description
[H14.29]. If the request URI refers to an aggregate, the field or media stream) was last modified. See [H14.29]. For the methods
indicates the last modification time across all leave nodes of that DESCRIBE or ANNOUNCE, the header field indicates the last modification
aggregate. date and time of the description, for SETUP that of the media stream.
12.23 Location 12.25 Location
See [H14.30]. See [H14.30].
12.24 PEP 12.26 Proxy-Authenticate
This corresponds to the PEP: header in the ``Protocol Extension See [H14.33].
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).
12.25 PEP-Info 12.27 Proxy-Require
This corresponds to the PEP-Info: header in the ``Protocol Extension The Proxy-Require header is used to indicate proxy-sensitive
Protocol'' defined in RFC XXXX. features that MUST be stripped by the proxy to the server if not
supported. Furthermore, any Proxy-Require header features that are not
supported by the proxy MUST be negatively acknowledged by the proxy to
the client if not supported.
12.26 Proxy-Authenticate See Section 12.32 for more details on the mechanics of this message
and a usage example.
See [H14.33]. We explored using the W3C's PEP proposal [22] for this
functionality. However, we determined that such a device was too
complex for our needs.
12.27 Public This field roughly corresponds to the C-PEP field in the PEP draft.
12.28 Public
See [H14.35]. See [H14.35].
12.28 Range 12.29 Range
This request header field specifies a range of time. The range can This request and response header field specifies a range of time.
be specified in a number of units. This specification defines the The range can be specified in a number of units. This specification
smpte (see Section 3.5) and clock (see Section 3.7) range units. defines the smpte (see Section 3.5) and clock (see Section 3.7) range
Within RTSP, byte ranges [H14.36.1] are not meaningful and MUST NOT be units. Within RTSP, byte ranges [H14.36.1] are not meaningful and MUST
used. The header may also contain a time parameter in UTC, specifying NOT be used. The header may also contain a time parameter in UTC,
the time at which the operation is to be made effective. Servers specifying the time at which the operation is to be made effective.
supporting the Range header MUST understand the NPT and SMPTE range Servers supporting the Range header MUST understand the NPT range
formats. format and SHOULD understand the SMPTE range format. The Range
response header indicates what range of time is actually being played
or recorded.
Range = "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ] Range = "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ]
ranges-specifier = npt-range | utc-range | smpte-range ranges-specifier = npt-range | utc-range | smpte-range
Example: Example:
Range: clock=19960213T143205Z-;time=19970123T143720Z Range: clock=19960213T143205Z-;time=19970123T143720Z
The notation is similar to that used for the HTTP/1.1 header. It The notation is similar to that used for the HTTP/1.1 header. It
allows to select a clip from the media object, to play from a given 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. 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 The start of playback can be scheduled for at any time in the
future, although a server may refuse to keep server resources for future, although a server may refuse to keep server resources for
extended idle periods. extended idle periods.
12.29 Referer 12.30 Referer
See [H14.37]. The URL refers to that of the presentation See [H14.37]. The URL refers to that of the presentation
description, typically retrieved via HTTP. description, typically retrieved via HTTP.
12.30 Retry-After 12.31 Retry-After
See [H14.38]. See [H14.38].
12.31 Scale 12.32 Require
The Require header is used by clients to query the server about
features that it may or may not support. The server MUST respond to
this header by negatively acknowledging those features which are NOT
supported in the Unsupported header.
For example
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/1.0 200 Option not supported
CSeq: 302
Unsupported: funky-feature
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 303
S->C: RTSP/1.0 200 OK
CSeq: 303
This is to make sure that the client-server interaction will proceed
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.
We explored using the W3C's PEP proposal [22] for this
functionality. However, we determined that such a device was too
complex for our needs.
This field roughly corresponds to the PEP field in the PEP draft.
Proxies and other intermediary devices SHOULD ignore features that are
not understood in this field. If a particular extension requires that
intermediate devices support it, the extension should be tagged in the
Proxy-Require field instead (see Section 3.4).
12.33 RTP-Info
This field is used to set RTP-specific parameters in the PLAY
response.
url:
Indicates the stream URL which for which the following RTP
parameters correspond.
seq:
Indicates the sequence number of the first packet of the
stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that
originated after the seek.
rtptime:
Indicates the RTP timestamp of the first packet of the stream.
The client uses this value to calculate the mapping of RTP time
to NPT.
This information is also available in RTCP timestamps. However, in
order to ensure that this information is available at the necessary
time (immediately at startup or after a seek), and that it is
delivered reliably, it is placed in the RTSP control channel as
well.
RTP-Info = "RTP-Info" ":"
1#stream-url ";"
*parameter
stream-url = "url" "=" url
parameter = ";" "seq" "=" sequence-number
sequence-number = 1*16(DIGIT)
Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=43754027,
url=rtsp://foo.com/bar.avi/streamid=1;seq=34834738
12.34 Scale
A scale value of 1 indicates normal play or record at the normal A scale value of 1 indicates normal play or record at the normal
forward viewing rate. If not 1, the value corresponds to the rate with forward viewing rate. If not 1, the value corresponds to the rate with
respect to normal viewing rate. For example, a ratio of 2 indicates respect to normal viewing rate. For example, a ratio of 2 indicates
twice the normal viewing rate (``fast forward'') and a ratio of 0.5 twice the normal viewing rate (``fast forward'') and a ratio of 0.5
indicates half the normal viewing rate. In other words, a ratio of 2 indicates half the normal viewing rate. In other words, a ratio of 2
has normal play time increase at twice the wallclock rate. For every has normal play time increase at twice the wallclock rate. For every
second of elapsed (wallclock) time, 2 seconds of content will be second of elapsed (wallclock) time, 2 seconds of content will be
delivered. A negative value indicates reverse direction. delivered. A negative value indicates reverse direction.
skipping to change at line 2174 skipping to change at line 2394
If the request contains a Range parameter, the new scale value will If the request contains a Range parameter, the new scale value will
take effect at that time. take effect at that time.
Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ] Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
Example of playing in reverse at 3.5 times normal rate: Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5 Scale: -3.5
12.32 Speed 12.35 Speed
This request header fields parameter requests the server to deliver This request header fields parameter requests the server to deliver
data to the client at a particular speed, contingent on the server's data to the client at a particular speed, contingent on the server's
ability and desire to serve the media stream at the given speed. ability and desire to serve the media stream at the given speed.
Implementation by the server is OPTIONAL. The default is the bit rate Implementation by the server is OPTIONAL. The default is the bit rate
of the stream. of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of The parameter value is expressed as a decimal ratio, e.g., a value of
2.0 indicates that data is to be delivered twice as fast as normal. A 2.0 indicates that data is to be delivered twice as fast as normal. A
speed of zero is invalid. If the request contains a Range parameter, speed of zero is invalid. If the request contains a Range parameter,
skipping to change at line 2201 skipping to change at line 2421
Use of this field changes the bandwidth used for data delivery. It is Use of this field changes the bandwidth used for data delivery. It is
meant for use in specific circumstances where preview of the meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. Implementors presentation at a higher or lower rate is necessary. Implementors
should keep in mind that bandwidth for the session may be negotiated should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. When data is delivered over UDP, it is highly may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss recommended that means such as RTCP be used to track packet loss
rates. rates.
12.33 Server 12.36 Server
See [H14.39] See [H14.39]
12.34 Session 12.37 Session
This request and response header field identifies an RTSP session, This request and response header field identifies an RTSP session,
started by the media server in a SETUP response and concluded by started by the media server in a SETUP response and concluded by
TEARDOWN on the presentation URL. The session identifier is chosen by TEARDOWN on the presentation URL. The session identifier is chosen by
the media server (see Section 3.4). Once a client receives a Session the media server (see Section 3.4). Once a client receives a Session
identifier, it MUST return it for any request related to that session. identifier, it MUST return it for any request related to that session.
Session = "Session" ":" session-id A server does not have to set up a session identifier if it has other
means of identifying a session, such as dynamically generated URLs.
Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
The timeout parameter is only allowed in a response header. The server
uses it to indicate to the client how long the server is prepared to
wait between RTSP commands before closing the session due to lack of
activity (see Section A). The timeout is measured in seconds, with a
default of 60 seconds (1 minute).
Note that a session identifier identifies a RTSP session across Note that a session identifier identifies a RTSP session across
transport sessions or connections. Control messages for more than one transport sessions or connections. Control messages for more than one
RTSP URL may be sent within a single RTSP session. Hence, it is RTSP URL may be sent within a single RTSP session. Hence, it is
possible that clients use the same session for controlling many possible that clients use the same session for controlling many
streams comprising a presentation, as long as all the streams come streams comprising a presentation, as long as all the streams come
from the same server. (See example in Section 14). However, multiple from the same server. (See example in Section 14). However, multiple
``user'' sessions for the same URL from the same client MUST use ``user'' sessions for the same URL from the same client MUST use
different session identifiers. different session identifiers.
The session identifier is needed to distinguish several delivery The session identifier is needed to distinguish several delivery
requests for the same URL coming from the same client. requests for the same URL coming from the same client.
12.35 Transport The response 454 (Session Not Found) is returned if the session
identifier is invalid.
12.38 Timestamp
The timestamp general header describes when the client sent the
request to the server. The value of the timestamp is of significance
only to the client and may use any timescale. The server MUST echo the
exact same value and MAY, if it has accurate information about this,
add a floating point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp is used by
the client to compute the round-trip time to the server so that it can
adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ]
[ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ]
12.39 Transport
This request header indicates which transport protocol is to be used This request header indicates which transport protocol is to be used
and configures its parameters such as destination address, and configures its parameters such as destination address,
compression, multicast time-to-live and destination port for a single compression, multicast time-to-live and destination port for a single
stream. It sets those values not already determined by a presentation stream. It sets those values not already determined by a presentation
description. description.
Transports are comma separated, listed in order of preference. Transports are comma separated, listed in order of preference.
Parameters may be added to each tranpsort, separated by a semicolon. Parameters may be added to each tranpsort, separated by a semicolon.
skipping to change at line 2250 skipping to change at line 2497
stream. stream.
The server MAY return a Transport response header in the response to The server MAY return a Transport response header in the response to
indicate the values actually chosen. indicate the values actually chosen.
A Transport request header field may contain a list of transport A Transport request header field may contain a list of transport
options acceptable to the client. In that case, the server MUST return options acceptable to the client. In that case, the server MUST return
a single option which was actually chosen. a single option which was actually chosen.
The syntax for the transport specifier is The syntax for the transport specifier is
transport/profile/lower-transport. Defaults for "lower-transport" are
specific to the profile. For RTP/AVP, the default is UDP. transport/profile/lower-transport.
The default value for the ``lower-transport'' parameters is specific
to the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport: Below are the configuration parameters associated with transport:
General parameters: General parameters:
unicast | multicast:
Mutually exclusive indication of whether unicast or multicast
delivery will be attempted. Default value is multicast. Clients
that are capable of handling both unicast and multicast
transmission MUST indicate such capability by including two
full transport-specs with separate parameters for each.
destination: destination:
The address to which a stream will be sent. The client may The address to which a stream will be sent. The client may
specify the multicast address with the destination parameter. A specify the multicast address with the destination parameter. A
server SHOULD authenticate the client and SHOULD log such server SHOULD authenticate the client and SHOULD log such
attempts before allowing the client to direct a media stream to attempts before allowing the client to direct a media stream to
an address not chosen by the server to avoid becoming the an address not chosen by the server to avoid becoming the
unwitting perpetrator of a remote-controlled denial-of-service unwitting perpetrator of a remote-controlled denial-of-service
attack. This is particularly important if RTSP commands are attack. This is particularly important if RTSP commands are
issued via UDP, but TCP cannot be relied upon as reliable means issued via UDP, but TCP cannot be relied upon as reliable means
of client identification by itself. A server SHOULD not allow a of client identification by itself. A server SHOULD not allow a
client to direct media streams to an address that differs from client to direct media streams to an address that differs from
the address commands are coming from. the address commands are coming from.
source:
Unicast only. If the source address for the stream is different
than can be derived from the RTSP endpoint address (the server
in playback or the client in recording), the source MAY be
specified.
This information may also be available through SDP, however, since
this is more a feature of transport than media initialization, the
authoritative source for this information should be in the SETUP
response.
layers:
The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses starting
at the destination address.
mode: mode:
The mode parameter indicates the methods to be supported for The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY. For RECORD, the append flag provided, the default is PLAY. For RECORD, the append flag
indicates that the media data should be appended to the indicates that the media data should be appended to the
existing resource rather than overwriting it. If appending is existing resource rather than overwriting it. If appending is
requested and the server does not support this, it MUST refuse requested and the server does not support this, it MUST refuse
the request rather than overwrite the resouce identified by the the request rather than overwrite the resouce identified by the
URI. The append parameter is ignored if the mode parameter does URI. The append parameter is ignored if the mode parameter does
not contain RECORD. not contain RECORD.
interleaved: interleaved:
The interleaved parameter implies mixing the media stream with The interleaved parameter implies mixing the media stream with
the control stream, in whatever protocol is being used by the the control stream, in whatever protocol is being used by the
control stream. Currently, the next-layer protocols RTP is control stream, using the mechanism defined in Section 10.12.
defined. The `channel' parameter defines the channel number to The argument provides the the channel number to be used in the
be used in the $ statement (see section 10.12). $ statement.
Multicast specific: Multicast specific:
ttl: ttl:
multicast time-to-live multicast time-to-live
RTP Specific: RTP Specific:
compressed: compressed:
Boolean parameter indicating compressed RTP according to RFC Boolean parameter indicating compressed RTP according to RFC
XXXX. XXXX.
port: port:
RTP/RTCP destination ports on client. The client receives RTCP the RTP/RTCP port pair for a multicast session. Specified as a
reports on the value of port plus one, as is standard RTP range (e.g. port=3456-3457).
convention.
cport: client_port:
the control port that the data server wishes the client to send the RTP/RTCP port pair on the server in the unicast model.
its RTCP reports to. Specified as a range (e.g. port=3456-3457).
server_port:
the RTP/RTCP port pair on the server in the unicast model.
Specified as a range (e.g. port=3456-3457).
ssrc: ssrc:
Indicates the RTP SSRC [19, Sec. 3] value that should be Indicates the RTP SSRC [20, Sec. 3] value that should be
(request) or will be (response) used by the media server. This (request) or will be (response) used by the media server. This
parameter is only valid for unicast transmission. It identifies parameter is only valid for unicast transmission. It identifies
the synchronization source to be associated with the media the synchronization source to be associated with the media
stream. stream.
Transport = "Transport" ":" Transport = "Transport" ":"
1#transport-protocol/profile[/lower-transport] *parameter 1\#transport-spec
transport-spec = transport-protocol/profile[/lower-transport]
*parameter
transport-protocol = "RTP" transport-protocol = "RTP"
profile = "AVP" profile = "AVP"
lower-transport = "TCP" | "UDP" lower-transport = "TCP" | "UDP"
parameter = ";" "destination" [ "=" address ] parameter = ( "unicast" | "multicast" )
| ";" "destination" [ "=" address ]
| ";" "compressed" | ";" "compressed"
| ";" "channel" "=" channel | ";" "interleaved" "=" channel
| ";" "append" | ";" "append"
| ";" "ttl" "=" ttl | ";" "ttl" "=" ttl
| ";" "port" "=" port | ";" "layers" "=" 1*DIGIT
| ";" "cport" "=" port | ";" "port" "=" port [ "-" port ]
| ";" "client_port" "=" port [ "-" port ]
| ";" "server_port" "=" port [ "-" port ]
| ";" "ssrc" "=" ssrc | ";" "ssrc" "=" ssrc
| ";" "mode" = <"> 1#mode <"> | ";" "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)
channel = 1*3(DIGIT) channel = 1*3(DIGIT)
address = host address = host
mode = "PLAY" | "RECORD" *parameter mode = "PLAY" | "RECORD" *parameter
Example: Example:
Transport: RTP/AVP;compressed;ttl=127;port=3456; Transport: RTP/AVP;multicast;compressed;ttl=127;mode="PLAY",
mode="PLAY,RECORD;append" RTP/AVP;unicast;compressed;client_port=3456-3457;mode="PLAY"
The Transport header is restricted to describing a single RTP The Transport header is restricted to describing a single RTP
stream. (RTSP can also control multiple streams as a single stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of of session description formats greatly simplifies designs of
firewalls. firewalls.
12.36 Transport-Info 12.40 Unsupported
This field is used to set Transport specific parameters in the PLAY Negative acknowledgement of features not supported by the server. In
response. the case where the feature was specified via the Proxy-Require: field
(Section 12.32), if there is a proxy on the path between the client
and the server, the proxy MUST insert a message reply with an error
message 506 (Feature not supported).
seq: We explored using the W3C's PEP proposal [22] for this
Indicates the sequence number of the first packet of the functionality. However, we determined that such a device was too
stream. This allows clients to gracefully deal with packets complex for our needs.
when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that
originated after the seek.
Transport-Info = "Transport-Info" ":" This field roughly corresponds to the PEP-Info and C-PEP-Info in
1#transport-protocol/profile[/lower-transport] ";" the PEP draft.
streamid
*parameter
transport-protocol = "RTP"
profile = "AVP"
lower-transport = "TCP" | "UDP"
stream-id = "streamid" "=" streamid
parameter = ";" "seq" "=" sequence number
sequence-number = 1*16(DIGIT)
Example: See Section 12.32 for a usage example.
Transport-Info: RTP/AVP;streamid=0;seq=43754027,
RTP/AVP;streamid=1;seq=34834738
12.37 User-Agent 12.41 User-Agent
See [H14.42] See [H14.42]
12.38 Vary 12.42 Vary
See [H14.43] See [H14.43]
12.39 Via 12.43 Via
See [H14.44]. See [H14.44].
12.40 WWW-Authenticate 12.44 WWW-Authenticate
See [H14.46]. See [H14.46].
13 Caching 13 Caching
In HTTP, response-request pairs are cached. RTSP differs In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not 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.) However, any data, caching is not really an issue for these requests.) However,
skipping to change at line 2407 skipping to change at line 2679
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 to On receiving a SETUP or PLAY request, the proxy would ascertain as to
whether it has an up-to-date copy of the continuous media content. If whether it has an up-to-date copy of the continuous media content. If
not, it would modify the SETUP transport parameters as appropriate and not, it would modify the SETUP transport parameters as appropriate and
forward the request to the origin server. Subsequent control commands forward the request to the origin server. Subsequent control commands
such as PLAY or PAUSE would pass the proxy unmodified. The proxy would such as PLAY or PAUSE would pass the proxy unmodified. The proxy would
then pass the continuous media data to the client, while possibly then pass the continuous media data to the client, while possibly
making a local copy for later re-use. The exact behavior allowed to making a local copy for later re-use. The exact behavior allowed to
the cache is given by the cache-response directives described in the cache is given by the cache-response directives described in
Section 12.10. A cache MUST answer any DESCRIBE requests if it is Section 12.8. A cache MUST answer any DESCRIBE requests if it is
currently serving the stream to the requestor, as it is possible that currently serving the stream to the requestor, as it is possible that
low-level details of the stream description may have changed on the low-level details of the stream description may have changed on the
origin-server. origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the Note that an RTSP cache, unlike the HTTP cache, is of the
``cut-through'' variety. Rather than retrieving the whole resource ``cut-through'' variety. Rather than retrieving the whole resource
from the origin server, the cache simply copies the streaming data as from the origin server, the cache simply copies the streaming data as
it passes by on its way to the client, thus, it does not introduce it passes by on its way to the client, thus, it does not introduce
additional latency. additional latency.
skipping to change at line 2464 skipping to change at line 2736
Content-Type: application/sdp Content-Type: application/sdp
v=0 v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202 o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session s=RTSP Session
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=murl:rtsp://audio.example.com/twister/audio.en a=murl:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31 m=video 0 RTP/AVP 31
a=murl:rtsp://audio.example.com/twister/video a=murl:rtsp://audio.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 1 C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1
Transport: rtp/udp;port=3056 Transport: rtp/udp;port=3056
A->C: RTSP/1.0 200 1 OK A->C: RTSP/1.0 200 OK
CSeq: 1
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
CSeq: 1
Transport: rtp/udp;port=3058 Transport: rtp/udp;port=3058
V->C: RTSP/1.0 200 1 OK V->C: RTSP/1.0 200 OK
CSeq: 1
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
CSeq: 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 OK
CSeq: 2
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 2 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 2
Session: 1234 Session: 1234
Range: smpte=0:10:00- Range: smpte=0:10:00-
A->C: RTSP/1.0 200 OK
CSeq: 2
A->C: RTSP/1.0 200 2 OK C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 3
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 3
Session: 1234 Session: 1234
A->C: RTSP/1.0 200 3 OK A->C: RTSP/1.0 200 OK
CSeq: 3
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
CSeq: 3
Session: 1235 Session: 1235
V->C: RTSP/1.0 200 3 OK V->C: RTSP/1.0 200 OK
CSeq: 3
Even though the audio and video track are on two different servers, Even though the audio and video track are on two different servers,
and may start at slightly different times and may drift with respect and may start at slightly different times and may drift with respect
to each other, the client can synchronize the two using standard RTP to each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender methods, in particular the time scale contained in the RTCP sender
reports. reports.
14.2 Streaming of a Container file 14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in For purposes of this example, a container file is a storage entity in
skipping to change at line 2520 skipping to change at line 2803
Container files are a widely used means to store such presentations. Container files are a widely used means to store such presentations.
While the components are essentially transported as independant While the components are essentially transported as independant
streams, it is desirable to maintain a common context for those streams, it is desirable to maintain a common context for those
streams at the server end. streams at the server end.
This enables the server to keep a single storage handle open This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case of easily. It also allows treating all the streams equally in case of
any prioritization of streams by the server. any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent It is also possible that the presentation author may wish to prevent
selective retreival of the streams by client in order to preserve the selective retrieval of the streams by client in order to preserve the
artistic effect of the combined media presentation. Similarly, in such artistic effect of the combined media presentation. Similarly, in such
a tightly bound presentation, it is desirable to be able to control a tightly bound presentation, it is desirable to be able to control
all the streams via a single control message using an aggregate URL. 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 The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URLs. multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M . The movie is Client C requests a presentation from media server M . The movie is
stored in a container file. The client has obtained a RTSP URL to the stored in a container file. The client has obtained a RTSP URL to the
container file. container file.
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0 1 C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 1 OK M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 64 Content-Length: 164
s= sample rtsp presentation
r = rtsp://foo/twister /* aggregate URL*/ v=0
o=- 2890844256 2890842807 IN IP4 172.16.2.93
s=RTSP Session
i=An Example of RTSP Session Usage
a=control:rtsp://foo/twister # aggregate URL
t=0 0
m= audio 0 RTP/AVP 0 m= audio 0 RTP/AVP 0
r = rtsp://foo/twister/audio a=control:rtsp://foo/twister/audio
m=video 0 RTP/AVP 26 m=video 0 RTP/AVP 26
r = rtsp://foo/twister/video a=control:rtsp://foo/twister/video
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0 2 C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
CSeq: 2
Transport: RTP/AVP;port=8000 Transport: RTP/AVP;port=8000
M->C: RTSP/1.0 200 2 OK M->C: RTSP/1.0 200 OK
CSeq: 2
Session: 1234 Session: 1234
C->M: SETUP rtsp://foo/twister/video RTSP/1.0 3 C->M: SETUP rtsp://foo/twister/video RTSP/1.0
CSeq: 3
Transport: RTP/AVP;port=8002 Transport: RTP/AVP;port=8002
Session: 1234 Session: 1234
M->C: RTSP/1.0 200 3 OK M->C: RTSP/1.0 200 OK
CSeq: 3
Session: 1234 Session: 1234
C->M: PLAY rtsp://foo/twister RTSP/1.0 4 C->M: PLAY rtsp://foo/twister RTSP/1.0
CSeq: 4
Range: npt=0- Range: npt=0-
Session: 1234 Session: 1234
M->C: RTSP/1.0 200 OK
M->C: RTSP/1.0 200 4 OK CSeq: 4
Session: 1234 Session: 1234
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0 5 C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
CSeq: 5
Session: 1234 Session: 1234
M->C: RTSP/1.0 4xx 5 Only aggregate operation allowed M->C: RTSP/1.0 4xx Only aggregate operation allowed
CSeq: 5
C->M: PAUSE rtsp://foo/twister RTSP/1.0 6 C->M: PAUSE rtsp://foo/twister RTSP/1.0
CSeq: 6
Session: 1234 Session: 1234
M->C: RTSP/1.0 200 6 OK M->C: RTSP/1.0 200 OK
CSeq: 6
Session: 1234 Session: 1234
C->M: SETUP rtsp://foo/twister RTSP/1.0 7 C->M: SETUP rtsp://foo/twister RTSP/1.0
CSeq: 7
Transport: RTP/AVP;port=10000 Transport: RTP/AVP;port=10000
M->C: RTSP/1.0 4xx 7 Aggregate operation not allowed M->C: RTSP/1.0 4xx Aggregate operation not allowed
CSeq: 7
In the first instance of failure, the client tries to pause one In the first instance of failure, the client tries to pause one stream
stream(in this case video) of the presentation which is disallowed for (in this case video) of the presentation which is disallowed for that
that presentation by the server. In the second instance, the aggregate presentation by the server. In the second instance, the aggregate URL
URL may not be used for SETUP and one control message is required per may not be used for SETUP and one control message is required per
stream to setup transport parameters. stream to setup transport parameters.
This keeps the syntax of the Transport header simple, and allows This keeps the syntax of the Transport header simple, and allows
easy parsing of transport information by firewalls. easy parsing of transport information by firewalls.
14.3 Live Media Presentation Using Multicast 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.
skipping to change at line 2600 skipping to change at line 2901
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.
C->W: GET /concert.sdp 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 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
CSeq: 1
M->C: RTSP/1.0 200 1 OK M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp Content-Type: application/sdp
v=0 v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202 o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session s=RTSP Session
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
c=IN IP4 224.2.0.1/16 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
CSeq: 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 M->C: RTSP/1.0 200 OK
CSeq: 2
M->C: RTSP/1.0 200 3 OK C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 3
M->C: RTSP/1.0 200 OK
CSeq: 3
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.
14.4 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: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0 1 C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 1
Accept: application/sdp Accept: application/sdp
M->C: RTSP/1.0 200 1 OK M->C: RTSP/1.0 200 1 OK
Content-type: application/rtsl Content-type: application/rtsl
v=0 v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202 o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session s=RTSP Session
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
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
Conference: 218kadjk CSeq: 2
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
14.5 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 90 C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0
CSeq: 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: RTSP/1.0 200 90 OK M->C: RTSP/1.0 200 OK
CSeq: 90
C->S: SETUP rtsp://server.example.com/meeting RTSP/1.0 91 C->S: SETUP rtsp://server.example.com/meeting RTSP/1.0
CSeq: 91
Transport: RTP/AVP;mode=record Transport: RTP/AVP;mode=record
S->C: RTSP/1.0 200 91 OK S->C: RTSP/1.0 200 OK
CSeq: 91
Transport: RTP/AVP;port=3244;mode=record Transport: RTP/AVP;port=3244;mode=record
Session: 508876 Session: 508876
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 92 C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
CSeq: 92
Session: 508876 Session: 508876
Range: clock 19961110T1925-19961110T2015 Range: clock 19961110T1925-19961110T2015
S->C: RTSP/1.0 200 OK
CSeq: 92
15 Syntax 15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF) The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 (HTTP/1.1). as used in RFC 2068 (HTTP/1.1).
15.1 Base Syntax 15.1 Base Syntax
OCTET = <any 8-bit sequence of data> OCTET = <any 8-bit sequence of data>
CHAR = <any US-ASCII character (octets 0 - 127)> CHAR = <any US-ASCII character (octets 0 - 127)>
UPALPHA = <any US-ASCII uppercase letter "A".."Z"> UPALPHA = <any US-ASCII uppercase letter "A".."Z">
skipping to change at line 2749 skipping to change at line 3066
The RTSP client and server state machines describe the behavior of The RTSP client and server state machines describe the behavior of
the protocol from RTSP session initialization through RTSP session the protocol from RTSP session initialization through RTSP session
termination. termination.
State is defined on a per object basis. An object is uniquely State is defined on a per object basis. An object is uniquely
identified by the stream URL and the RTSP session identifier. Any identified by the stream URL and the RTSP session identifier. Any
request/reply using aggregate URLs denoting RTSP presentations request/reply using aggregate URLs denoting RTSP presentations
comprised of multiple streams will have an effect on the individual comprised of multiple streams will have an effect on the individual
states of all the streams. For example, if the presentation /movie states of all the streams. For example, if the presentation /movie
contains two streams /movie/audio and /movie/video, then the following contains two streams, /movie/audio and /movie/video, then the
command: following command:
PLAY rtsp://foo.com/movie RTSP/1.0 559 PLAY rtsp://foo.com/movie RTSP/1.0
CSeq: 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 streams in This example does not imply a standard way to represent streams in
URLs or a relation to the filesystem. See Section 3.2. URLs or a relation to the filesystem. See Section 3.2.
The requests OPTIONS, DESCRIBE, GET_PARAMETER, SET_PARAMETER do not The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER, SET_PARAMETER
have any effect on client or server state and are therefore not listed do not have any effect on client or server state and are therefore not
in the state tables. listed in the state tables.
A.1 Client State Machine A.1 Client State Machine
The client can assume the following states: The client can assume the following states:
Init: Init:
SETUP has been sent, waiting for reply. SETUP has been sent, waiting for reply.
Ready: Ready:
SETUP reply received OR after playing, PAUSE reply received. SETUP reply received or PAUSE reply received while in Playing
state.
Playing: Playing:
PLAY reply received PLAY reply received
Recording: Recording:
RECORD reply received RECORD reply received
In general, the client changes state on receipt of replies to In general, the client changes state on receipt of replies to
requests. Note that some requests are effective at a future time or requests. Note that some requests are effective at a future time or
position(such as a PAUSE), and state also changes accordingly. If no position(such as a PAUSE), and state also changes accordingly. If no
explicit SETUP is required for the object (for example, it is explicit SETUP is required for the object (for example, it is
available via a multicast group), state begins at READY. In this case, available via a multicast group), state begins at Ready. In this case,
there are only two states, READY and PLAYING. there are only two states, Ready and Playing.
The client also changes state from Playing/Recording to Ready when the The client also changes state from Playing/Recording to Ready when the
end of the requested range is reached. end of the requested range is reached.
The ``next state'' column indicates the state assumed after receiving The ``next state'' column indicates the state assumed after receiving
a success response (2xx). If a request yields a status code of 3xx, 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 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 state. Messages not listed for each state MUST NOT be issued by the
client in that state, with the exception of messages not affecting client in that state, with the exception of messages not affecting
state, as listed above. Receiving a REDIRECT from the server is state, as listed above. Receiving a REDIRECT from the server is
equivalent to receiving a 3xx redirect status from the server. equivalent to receiving a 3xx redirect status from the server.
state message 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
SETUP Ready
Playing PAUSE Ready Playing PAUSE Ready
TEARDOWN Init TEARDOWN Init
PLAY Playing PLAY Playing
SETUP Playing (changed transport) SETUP Playing (changed transport)
Recording PAUSE Ready Recording PAUSE Ready
TEARDOWN Init TEARDOWN Init
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: Init:
The initial state, no valid SETUP received. The initial state, no valid SETUP has been received yet.
Ready: Ready:
Last SETUP received was successful, reply sent or after 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: Playing:
Last PLAY received was successful, reply sent. Data is being Last PLAY received was successful, reply sent. Data is being
sent. sent.
Recording: Recording:
The server is recording media data. The server is recording media data.
In general,the server changes state on receiving requests. If the In general,the server changes state on receiving requests. If the
server is in state Playing or Recording and in unicast mode, it MAY server is in state Playing or Recording and in unicast mode, it MAY
revert to Init and tear down the RTSP session if it has not received revert to Init and tear down the RTSP session if it has not received
``wellness'' information, such as RTCP reports, from the client for a ``wellness'' information, such as RTCP reports or RTSP commands, from
defined interval, with a default of one minute. If the server is in the client for a defined interval, with a default of one minute. The
state Ready, it MAY revert to Init if it does not receive an RTSP server can declare another timeout value in the Session response
request for an interval of more than one minute. Note that some header (Section 12.37). If the server is in state Ready, it MAY revert
requests(such as PAUSE) may be effective at a future time or position, to Init if it does not receive an RTSP request for an interval of more
and server state transitions at the appropriate time. The server than one minute. Note that some requests (such as PAUSE) may be
reverts from state Playing or Recording to state Ready at the end of effective at a future time or position, and server state transitions
the range requested by the client. at the appropriate time. The server reverts from state Playing or
Recording to state Ready at the end of the range requested by the
client.
The REDIRECT message, when sent, is effective immediately unless it The REDIRECT message, when sent, is effective immediately unless it
has a Range: header specifying when the redirect is effective. In such has a Range header specifying when the redirect is effective. In such
a case, server state will also change at the appropriate time. a case, server state will also change at the appropriate time.
If no explicit SETUP is required for the object, state starts at If no explicit SETUP is required for the object, the state starts at
READY, there are only two states READY and PLAYING. Ready and there are only two states, Ready and Playing.
The ``next state'' column indicates the state assumed after sending a The ``next state'' column indicates the state assumed after sending a
success response (2xx). If a request results in a status code of 3xx, success response (2xx). If a request results in a status code of 3xx,
the state becomes Init. A status code of 4xx results in no change. the state becomes Init. A status code of 4xx results in no change.
state message 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
skipping to change at line 2870 skipping to change at line 3192
RECORD Recording RECORD Recording
Playing PLAY Playing Playing PLAY Playing
PAUSE Ready PAUSE Ready
TEARDOWN Init TEARDOWN Init
SETUP Playing SETUP Playing
Recording RECORD Recording Recording RECORD Recording
PAUSE Ready PAUSE Ready
TEARDOWN Init TEARDOWN Init
SETUP Recording SETUP Recording
B Open Issues B Interaction with RTP
1. RTSP allows to play selected, non-contiguous sections of a
Define text/rtsp-parameter MIME type. presentation. The media client playing back the RTP stream should not
2. be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
Reverse: Scale: -1, with reversed start times, or both? timestamps MUST be continuous and monotonic across jumps of NPT.
3.
HS believes that RTSP should only control individual media As an example, assume a clock frequency of 8000 Hz, a packetization
objects rather than aggregates. This avoids disconnects between interval of 100 ms and an initial sequence number and timestamp of
presentation descriptions and streams and avoids having to deal zero. First we play NPT 10 through 15, then skip ahead and play NPT 18
separately with single-host and multi-host case. Cost: several through 20. The first segment is presented as RTP packets with
PLAY/PAUSE/RECORD in one packet, one for each stream. sequence numbers 0 through 49 and timestamp 0 through 39,200. The
4. second segment consists of RTP packets with sequence number 50 through
Allow changing of transport for a stream that's playing? May 69, with timestamps 40,000 through 55,200.
not be a great idea since the same can be accomplished by tear
down and re-setup. Exception: near-video-on-demand, where the We cannot assume that the RTSP client can communicate with the RTP
server changes the address in a PLAY response. Servers may not media agent, as the two may be independent processes. If the RTP
be able to reliably send TEARDOWN to clients and the client timestamp shows the same gap as the NPT, the media agent will
wouldn't know why this happened in any event. assume that there is a pause in the presentation. If the jump in
5. NPT is large enough, the RTP timestamp may roll over and the media
How does the server get back to the client unless a persistent agent may believe later packets to be duplicates of packets just
played out.
For scaling (see Section 12.34), RTP timestamps should correspond to
the playback timing. For example, when playing video recorded at 30
frames/second at a scale of two and speed (Section 12.35) of one, the
server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame.
The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning. The
sequence parameter of the RTP-Info (Section 12.33 header provides the
last sequence number of the previous segment.
C Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP [6]) may be used to describe
streams or presentations in RTSP. Such usage is limited to specifying
means of access and encoding(s) for:
* Scenario A: A presentation comprised of streams from one or more
servers that are not available for aggregate control. Such a
description is typically retrieved by HTTP or other non-RTSP
means. However, they MAY be received with ANNOUNCE methods.
* Scenario B: A presentation comprised of multiple streams from a
single server that are available for aggregate control. Such a
description is typically returned in reply to a DESCRIBE request
on a URL, or received in an ANNOUNCE method.
Specifically, this appendix addresses the usage of SDP (for example,
embedded in a web page) that triggers a RTSP session, and the usage in
replies to RTSP DESCRIBE requests. However, it does not address the
issue of media or encoding negotiation within such descriptions.
C.1 Specification
The terms ``session-level'', ``media-level'' and other key/attribute
names and values used in this appendix are as defined in [6]. SDP
fields not specifically mentioned in this section are assumed to have
their usual meaning.
C.1.1 Control URL
The ``a=control:'' field is used to convey the control URL. This
field is used both at the media-level to provide a means to reference
individual streams, and at the session-level to signify a global URL
for aggregate control, providing the URL to be used on aggregate
commands (PLAY, PAUSE, etc.).
Example:
a=control:rtsp://example.com/foo
This field may contain both relative and absolute URLs, following the
rules and conventions set out in RFC 1808 ([16]). Specifically, the
order for which implementations should look for a base URL is as
follows:
* The RTSP Content-Base field
* The RTSP Content-Location field
* The RTSP request URL
If this field contains only an asterix (*), then the URL is treated as
if it were an empty embedded URL, and thus inherits the entire base
URL.
C.1.2 Media streams
The ``m='' field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate
synchronization. If the session is unicast, the port number simply
serves as a recommendation, and would still need to be conveyed to the
server via a SETUP request. The port number may be specified as 0, in
which case the client makes the choice of the port.
Example:
m=audio 0 RTP/AVP 31
C.1.3 Payload type(s)
The payload type(s) are specified in the ``m='' field. In case the
payload type is a static payload type from RFC 1890([1]), no other
information is required. In case it is a dynamic payload type, the
media attribute ``rtpmap'' is used to specify what the media is. The
``encoding name'' within the ``rtpmap'' attribute may be one of those
specified in RFC 1890(Sections 5 and 6), or an experimental encoding
with a ``X-'' prefix as specified in [6]. Codec-specific parameters
are not specified in this field but the ``fmtp'' attribute described
below. Implementors seeking to register new encodings should follow
the procedure in RFC 1890. If the media type is not suited to the RTP
AV profile, then it is recommended that a new profile be created and
the appropriate profile name must be used in lieu of ``RTP/AVP'' in
the ``m='' field. An informational document may be published in lieu
of this if the usage is expected to be limited or experimental.
C.1.4 Format specific parameters
This is accomplished using the ``fmtp'' media attribute. The syntax of
the ``fmtp'' attribute is specific to the encoding(s) that the
attribute refers to. This is with the exception of the number of
samples per packet, which is conveyed using the ``ptime'' attribute.
C.1.5 Length of presentation
This is applicable to non-live sessions(typically on-demand retreivals
of stored files) only and is specified using a media-level
``a=length'' field. It defines the total length of the presentation in
time. The unit is specified first, followed by the value. The units
and their values are as defined in Section 3.
Example :
a=length:npt=34.4368
C.1.6 Time of availability
It is required that suitable values for the start and stop times for
the ``t='' field be used for both scnearios. In Scenario B, the server
SHOULD indicate a stop time value for which it guarantees the
description to be valid, and a start time that is equal to or before
the time at which the DESCRIBE request was received.(It MAY also
indicate start and stop times of 0, meaning that the session is always
available). In Scenario A, the values should reflect the actual period
for which the session is avaiable in keeping with SDP semantics, and
not depend on other means(such as the life of the web page containing
the description) for this purpose.
C.1.7 Connection Information
In some cases, the mandatory ``c='' field may have no well-defined
interpretation. This is since all the necessary information may be
conveyed by the control URL and subsequent RTSP operations. In such
cases, the address within this field must be set to a suitable null
value. For address of type ``IP4'', this value is ``0.0.0.0''.
C.1.8 Entity Tag
Because RTSP supports the If-Match field (see section 12.22) in a
session-description-independent fashion, it's necessary to embed an
entirely opaque uniqueness field in the specification. The contents of
this tag is totally implementation specific, so long as it serves as a
unique identifier for this exact description of the media. Support of
this tag is optional.
Example :
a=etag:''158bb3e7c7fd62ce67f12b533f06b83a''
One could argue that the o= field provides identical functionality.
However, it does so in a manner that would put constraints on
servers that need to support multiple session description types
other than SDP for the same piece of media content.
C.2 Scenario A
Multiple media sections are specified, and each section MUST have the
control URL specified via the ``a=control:''field.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I came from a web page
t=0 0
c=IN IP4 0.0.0.0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid
Note that the control URL in this case implies that the client
establishes seperate RTSP control sessions to the servers audio.com
and video.com.
C.3 Scenario B
In this scenario, the server has multiple streams that are available
for aggregate control. In this case, there is both a media-level
``a=control:'' field which is used to specify the stream URL, and a
session-level ``a=control:'' field which is used as a global handle
for aggregate control. The media-level URLs may be relative, in which
case they resolve to absolute URLs as defined in C.1.1 above.
If the session comprises only a single stream, the media-level
``a=control:'' field may be omitted altogether. In case more than one
stream is present, the ``a=control:'' field MUST be used.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I contain
i=<more info>
t=0 0
c=IN IP4 0.0.0.0
a=control:rtsp://example.com/movie/
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
In this example, the client is required to establish a single RTSP
session to the server, and uses the URLs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to setup the media streams, and
rtsp://example.com/movie/ to control it.
D Minimal RTSP implementation
D.1 Client
A client implementation MUST be able to do the following :
* Generate the following requests : SETUP, TEARDOWN, and one of
PLAY(ie. a minimal playback client) or RECORD(ie. a minimal
recording client). If RECORD is implemented, ANNOUNCE must be
implemented as well.
* Include the following headers in requests: Connection, Session,
Transport. If ANNOUNCE is implemented, the capability to include
headers Content-Language, Content-Encoding, Content-Length,
Content-Type should be as well.
* Parse and understand the following headers in responses:
Connection, Session, Transport, Content-Language,
Content-Encoding, Content-Length, Content-Type. If RECORD is
implemented, the Location header must be understood as well.
RTP-complient implementations should also implement RTP-Info.
* Understand the class of each error code received and notify the
end-user, if one is present, of error codes in classes 4xx and
5xx. The notification requirement may be relaxed if the end-user
explicity does not want it for one or all status codes.
* Expect and respond to asynchronous requests from the server, such
as ANNOUNCE. This does not necessarily mean that it should
implement the ANNOUNCE method, merely that it MUST respond
positively or negatively to any request received from the server
* Implement RTP transport.
Inclusion of the User-Agent header is recommended.
The following capability sets are defined over and above the minimal
implementation :
D.1.1 Basic Playback
The client MUST additionally be able to do the following:
* Include and parse the Range header, with ``npt'' units.
* Generate the PAUSE reqeust.
* Implement the REDIRECT method, and the Location header.
* Implement the OPTIONS method, and the Public header.
* Understand SDP session descriptions as defined in Appendix C
Implementation of DESCRIBE is highly recommended for this case.
D.1.2 Authentication-enabled
The client MUST additionally be able to do the following:
* Recognize the 401 status code.
* Parse and include the WWW-Authenicate header
* Implement Basic and Digest authentication
D.2 Server
A minimal server implementation MUST be able to do the following:
* Implement SETUP, TEARDOWN, OPTIONS and one of the PLAY(ie. a
minimal playback server) or RECORD(ie. a minimal recording server)
methods. If RECORD is implemented, ANNOUNCE should be implemented
as well.
* Include the following headers in responses: Connection,
Content-Length, Content-Type, Content-Language, Content-Encoding,
Transport, Public. The capability to include the Location header
should be implemented if the RECORD method is. RTP-complient
implementations should also implement the RTP-Info field.
* Parse and respond appropriately to the following headers in
requests: Connection, Session, Transport, Require.
* Implement RTP transport.
Inclusion of the Server header is recommended.
The following capability sets are defined over and above the minimal
implementation :
D.2.1 Basic Playback
The server MUST additionally be able to do the following:
* Include and parse the Range header, with ``npt'' units.
Implementation of ``smpte'' units is recommended.
* Implement the PAUSE method.
* Implement the REDIRECT method, and the Location header.
Implementation of DESCRIBE and generation of SDP descriptions as
defined in Appendix C is highly recommended for this case.
D.2.2 Authentication-enabled
The server MUST additionally be able to do the following:
* Generate the 401 status code when authentication is required for
the resource.
* Parse and include the WWW-Authenicate header
* Implement Basic and Digest authentication
E Open Issues
1. Define text/rtsp-parameter MIME type.
2. Allow byte offsets for Range (Prasoon Tiwari).
3. Reverse: Scale: -1, with reversed start times, or both?
4. 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. 5. Server issues TEARDOWN and other 'event' notifications to
Server issues TEARDOWN and other 'event' notifications to
client? This raises the problem discussed in the previous open client? This raises the problem discussed in the previous open
issue, but is useful for the client if the data stream contains issue, but is useful for the client if the data stream contains
no end indication. no end indication.
C Changes F Changes
Since the March 1997 version, the following changes were made: Since draft03 (July 30, 1997 version) of RTSP, the following changes
were made:
* PEP was removed, ``Require'' header returns
* Usage of SDP within RTSP is specified as an appendix
* Minimal RTSP implementation specified as an appendix
* The RTSP control sequence number was moved off of the request and
response lines, and put into a new CSeq: header.
* Interaction with RTP appendix added
* Several changes to Transport: and RTP-Info: fields (RTP-Info: was
formerly Transport-Info:)
Between draft02 (March, 1997) and draft03 (July, 1997), the following
changes were made:
* Definition of RTP behavior.
* Definition of behavior for container files.
* Remove server-to-client DESCRIBE request.
* Allowing the Transport header to direct media streams to unicast * Allowing the Transport header to direct media streams to unicast
and multicast addresses, with an appropriate warning about and multicast addresses, with an appropriate warning about
denial-of-service attacks. denial-of-service attacks.
* Add mode parameter to Transport header to allow RECORD or PLAY. * Add mode parameter to Transport header to allow RECORD or PLAY.
* The Embedded binary data section was modified to clearly indicate * The Embedded binary data section was modified to clearly indicate
the stream the data corresponds to, and a reference to the the stream the data corresponds to, and a reference to the
Transport header was added. Transport header was added.
* The Transport header format has been changed to use a more general * The Transport header format has been changed to use a more general
means to specify data channel and application level protocol. It means to specify data channel and application level protocol. It
also conveys the port to be used at the server for RTCP messages, also conveys the port to be used at the server for RTCP messages,
skipping to change at line 2929 skipping to change at line 3567
* Example showing the use of aggregate/presentation control using a * Example showing the use of aggregate/presentation control using a
single RTSP session has been added. single RTSP session has been added.
* Support for the PEP(Protocol Extension Protocol) headers has been * Support for the PEP(Protocol Extension Protocol) headers has been
added. added.
* Server-Client DESCRIBE messages have been renamed to ANNOUNCE for * Server-Client DESCRIBE messages have been renamed to ANNOUNCE for
better clarity and differentiation. better clarity and differentiation.
Note that this list does not reflect minor changes in wording or Note that this list does not reflect minor changes in wording or
correction of typographical errors. correction of typographical errors.
D Author Addresses G 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 501 E. Middlefield Road
Mountain View, CA 94043 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 H Acknowledgements
This draft is based on the functionality of the original RTSP draft This draft is based on the functionality of the original RTSP draft
submitted in October 96. It also borrows format and descriptions from submitted in October 96. It also borrows format and descriptions from
HTTP/1.1. HTTP/1.1.
This document has benefited greatly from the comments of all those This document has benefited greatly from the comments of all those
participating in the MMUSIC-WG. In addition to those already participating in the MMUSIC-WG. In addition to those already
mentioned, the following individuals have contributed to this mentioned, the following individuals have contributed to this
specification: specification:
Rahul Agarwal Eduardo F. Llach Rahul Agarwal, Bruce Butterfield, Steve Casner, Francisco Cortes,
Bruce Butterfield Rob McCool Martin Dunsmuir, Eric Fleischman, V. Guruprasad, Peter Haight, Mark
Steve Casner David Oran Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka, Anders Klemets,
Martin Dunsmuir Sujal Patel Ruth Lang, Stephanie Leif, Eduardo F. Llach, Rob McCool, David Oran,
Eric Fleischman Sujal Patel, Alagu Periyannan, Igor Plotnikov, Pinaki Shah, Jeff
Mark Handley Igor Plotnikov Smith, Alexander Sokolsky, Dale Stammen, and John Francis Stracke.
Peter Haight Pinaki Shah
Brad Hefta-Gaub Jeff Smith
John K. Ho Alexander Sokolsky
Ruth Lang Dale Stammen
Stephanie Leif John Francis Stracke
References References
1 1 H. Schulzrinne, ``RTP profile for audio and video conferences
H. Schulzrinne, ``RTP profile for audio and video conferences
with minimal control,'' RFC 1890, Internet Engineering Task with minimal control,'' RFC 1890, Internet Engineering Task
Force, Jan. 1996. Force, Jan. 1996.
2
D. Kristol and L. Montulli, ``HTTP state management 2 D. Kristol and L. Montulli, ``HTTP state management
mechanism,'' RFC 2109, Internet Engineering Task Force, Feb. mechanism,'' RFC 2109, Internet Engineering Task Force, Feb.
1997. 1997.
3
F. Yergeau, G. Nicol, G. Adams, and M. Duerst, 3 F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
``Internationalization of the hypertext markup language,'' RFC ``Internationalization of the hypertext markup language,'' RFC
2070, Internet Engineering Task Force, Jan. 1997. 2070, Internet Engineering Task Force, Jan. 1997.
4
S. Bradner, ``Key words for use in RFCs to indicate requirement 4 S. Bradner, ``Key words for use in RFCs to indicate requirement
levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997. levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997.
5
R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. 5 R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T.
Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC
2068, Internet Engineering Task Force, Jan. 1997. 2068, Internet Engineering Task Force, Jan. 1997.
6
M. Handley, ``SDP: Session description protocol,'' Internet 6 M. Handley, ``SDP: Session description protocol,'' Internet
Draft, Internet Engineering Task Force, Nov. 1996. Draft, Internet Engineering Task Force, Nov. 1996.
Work in progress. Work in progress.
7
A. Freier, P. Karlton, and P. Kocher, ``The TLS protocol,'' 7 A. Freier, P. Karlton, and P. Kocher, ``The TLS protocol,''
Internet Draft, Internet Engineering Task Force, Dec. 1996. Internet Draft, Internet Engineering Task Force, Dec. 1996.
Work in progress. Work in progress.
8
J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and 8 J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and
E. L. Stewart, ``An extension to HTTP: digest access E. L. Stewart, ``An extension to HTTP: digest access
authentication,'' RFC 2069, Internet Engineering Task Force, authentication,'' RFC 2069, Internet Engineering Task Force,
Jan. 1997. Jan. 1997.
9
J. Postel, ``User datagram protocol,'' STD 6, RFC 768, Internet 9 J. Postel, ``User datagram protocol,'' STD 6, RFC 768, Internet
Engineering Task Force, Aug. 1980. Engineering Task Force, Aug. 1980.
10
R. Hinden and C. Partridge, ``Version 2 of the reliable data 10 R. Hinden and C. Partridge, ``Version 2 of the reliable data
protocol (RDP),'' RFC 1151, Internet Engineering Task Force, protocol (RDP),'' RFC 1151, Internet Engineering Task Force,
Apr. 1990. Apr. 1990.
11
J. Postel, ``Transmission control protocol,'' STD 7, RFC 793, 11 J. Postel, ``Transmission control protocol,'' STD 7, RFC 793,
Internet Engineering Task Force, Sept. 1981. Internet Engineering Task Force, Sept. 1981.
12
M. Handley, H. Schulzrinne, and E. Schooler, ``SIP: Session 12 M. Handley, H. Schulzrinne, and E. Schooler, ``SIP: Session
initiation protocol,'' Internet Draft, Internet Engineering initiation protocol,'' Internet Draft, Internet Engineering
Task Force, Dec. 1996. Task Force, Dec. 1996.
Work in progress. Work in progress.
13
P. McMahon, ``GSS-API authentication method for SOCKS version 13 P. McMahon, ``GSS-API authentication method for SOCKS version
5,'' RFC 1961, Internet Engineering Task Force, June 1996. 5,'' RFC 1961, Internet Engineering Task Force, June 1996.
14
D. Crocker, ``Augmented BNF for syntax specifications: ABNF,'' 14 D. Crocker, ``Augmented BNF for syntax specifications: ABNF,''
Internet Draft, Internet Engineering Task Force, Oct. 1996. Internet Draft, Internet Engineering Task Force, Oct. 1996.
Work in progress. Work in progress.
15
R. Elz, ``A compact representation of IPv6 addresses,'' RFC 15 R. Elz, ``A compact representation of IPv6 addresses,'' RFC
1924, Internet Engineering Task Force, Apr. 1996. 1924, Internet Engineering Task Force, Apr. 1996.
16
T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform 16 R. Fielding, ``Relative Uniform Resource Locators,'' RFC 1808,
Internet Engineering Task Force, June 1995.
17 T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform
resource locators (URL),'' RFC 1738, Internet Engineering Task resource locators (URL),'' RFC 1738, Internet Engineering Task
Force, Dec. 1994. Force, Dec. 1994.
17
International Telecommunication Union, ``Visual telephone 18 International Telecommunication Union, ``Visual telephone
systems and equipment for local area networks which provide a systems and equipment for local area networks which provide a
non-guaranteed quality of service,'' Recommendation H.323, non-guaranteed quality of service,'' Recommendation H.323,
Telecommunication Standardization Sector of ITU, Geneva, Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, May 1996. Switzerland, May 1996.
18
ISO/IEC, ``Information technology - generic coding of moving 19 ISO/IEC, ``Information technology - generic coding of moving
pictures and associated audio informaiton - part 6: extension pictures and associated audio informaiton - part 6: extension
for digital storage media and control,'' Draft International for digital storage media and control,'' Draft International
Standard ISO 13818-6, International Organization for Standard ISO 13818-6, International Organization for
Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland, Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
Nov. 1995. Nov. 1995.
19
H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, 20 H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
``RTP: a transport protocol for real-time applications,'' RFC ``RTP: a transport protocol for real-time applications,'' RFC
1889, Internet Engineering Task Force, Jan. 1996. 1889, Internet Engineering Task Force, Jan. 1996.
20
J. Miller, P. Resnick, and D. Singer, ``Rating Services and 21 J. Miller, P. Resnick, and D. Singer, ``Rating Services and
Rating Systems(and Their Machine Readable Descriptions), '' Rating Systems(and Their Machine Readable Descriptions), ''
REC-PICS-services-961031, Worldwide Web Consortium, Oct. 1996. REC-PICS-services-961031, Worldwide Web Consortium, Oct. 1996.
21
D. Connolly, R. Khare, H.F. Nielsen, ``PEP - an Extension 22 D. Connolly, R. Khare, H.F. Nielsen, ``PEP - an Extension
Mechanism for HTTP", Internet draft, work-in-progress. W3C Mechanism for HTTP", Internet draft, work-in-progress. W3C
Draft WD-http-pep-970714 Draft WD-http-pep-970714
http://www.w3.org/TR/WD-http-pep-970714, July, 1996. http://www.w3.org/TR/WD-http-pep-970714, July, 1996.
 End of changes. 

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