draft-ietf-mmusic-rtsp-04.txt   draft-ietf-mmusic-rtsp-05.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-04.txt Columbia U./Netscape/Progressive Networks draft-ietf-mmusic-rtsp-05.txt Columbia U./Netscape/RealNetworks
September 17, 1997 Expires: March 17, 1998 October 28, 1997 Expires: April 28, 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 37 skipping to change at line 37
Abstract: Abstract:
The Real Time Streaming Protocol, or RTSP, is an application-level The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP, sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC multicast UDP and TCP, and provide a means for choosing delivery
1889). mechanisms based upon RTP (RFC 1889).
This is a snapshot of the current draft which will become the next This is a snapshot of the current draft which will become the next
version of the ``official'' Internet Draft. version of the ``official'' Internet Draft.
Copyright Notice:
Copyright (C) The Internet Society (1997). All Rights Reserved.
H. Schulzrinne, A. Rao, R. Lanphier Page 1 H. Schulzrinne, A. Rao, R. Lanphier Page 1
Contents Table of 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
* 3 Protocol Parameters * 3 Protocol Parameters
+ 3.1 RTSP Version + 3.1 RTSP Version
+ 3.2 RTSP URL + 3.2 RTSP URL
+ 3.3 Conference Identifiers + 3.3 Conference Identifiers
+ 3.4 Session Identifiers + 3.4 Session Identifiers
+ 3.5 SMPTE Relative Timestamps + 3.5 SMPTE Relative Timestamps
+ 3.6 Normal Play Time + 3.6 Normal Play Time
+ 3.7 Absolute Time + 3.7 Absolute Time
+ 3.8 Option Tags
o 3.8.1 Registering New Option Tags With IANA
* 4 RTSP Message * 4 RTSP Message
+ 4.1 Message Types + 4.1 Message Types
+ 4.2 Message Headers + 4.2 Message Headers
+ 4.3 Message Body + 4.3 Message Body
+ 4.4 Message Length + 4.4 Message Length
* 5 General Header Fields * 5 General Header Fields
* 6 Request * 6 Request
+ 6.1 Request Line + 6.1 Request Line
+ 6.2 Request Header Fields + 6.2 Request Header Fields
* 7 Response * 7 Response
skipping to change at line 89 skipping to change at line 95
* 8 Entity * 8 Entity
+ 8.1 Entity Header Fields + 8.1 Entity Header Fields
+ 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
+ 10.5 PLAY
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+ 10.4 SETUP
+ 10.5 PLAY
+ 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
+ 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 Success 2xx
+ 11.2 Client Error 4xx o 11.1.1 250 Low on Storage Space
o 11.2.1 405 Method Not Allowed + 11.2 Redirection 3xx
o 11.2.2 451 Parameter Not Understood + 11.3 Client Error 4xx
o 11.2.3 452 Conference Not Found o 11.3.1 405 Method Not Allowed
o 11.2.4 453 Not Enough Bandwidth o 11.3.2 451 Parameter Not Understood
o 11.2.5 454 Session Not Found o 11.3.3 452 Conference Not Found
o 11.2.6 455 Method Not Valid in This State o 11.3.4 453 Not Enough Bandwidth
o 11.2.7 456 Header Field Not Valid for Resource o 11.3.5 454 Session Not Found
o 11.2.8 457 Invalid Range o 11.3.6 455 Method Not Valid in This State
o 11.2.9 458 Parameter Is Read-Only o 11.3.7 456 Header Field Not Valid for Resource
o 11.2.10 459 Aggregate operation not allowed o 11.3.8 457 Invalid Range
o 11.2.11 460 Only aggregate operation allowed o 11.3.9 458 Parameter Is Read-Only
o 11.3.10 459 Aggregate Operation Not Allowed
o 11.3.11 460 Only Aggregate Operation Allowed
o 11.3.12 461 Unsupported Transport
o 11.3.13 462 Destination Unreachable
o 11.3.14 551 Option not supported
* 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 Cache-Control + 12.8 Cache-Control
+ 12.9 Conference + 12.9 Conference
+ 12.10 Connection + 12.10 Connection
+ 12.11 Content-Base + 12.11 Content-Base
+ 12.12 Content-Encoding + 12.12 Content-Encoding
+ 12.13 Content-Language + 12.13 Content-Language
+ 12.14 Content-Length + 12.14 Content-Length
+ 12.15 Content-Location + 12.15 Content-Location
+ 12.16 Content-Type + 12.16 Content-Type
+ 12.17 CSeq + 12.17 CSeq
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+ 12.18 Date + 12.18 Date
+ 12.19 Expires + 12.19 Expires
+ 12.20 From + 12.20 From
+ 12.21 Host + 12.21 Host
+ 12.22 If-Match + 12.22 If-Match
+ 12.23 If-Modified-Since + 12.23 If-Modified-Since
+ 12.24 Last-Modified + 12.24 Last-Modified
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+ 12.25 Location + 12.25 Location
+ 12.26 Proxy-Authenticate + 12.26 Proxy-Authenticate
+ 12.27 Proxy-Require + 12.27 Proxy-Require
+ 12.28 Public + 12.28 Public
+ 12.29 Range + 12.29 Range
+ 12.30 Referer + 12.30 Referer
+ 12.31 Retry-After + 12.31 Retry-After
+ 12.32 Require + 12.32 Require
+ 12.33 RTP-Info + 12.33 RTP-Info
+ 12.34 Scale + 12.34 Scale
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+ 12.39 Transport + 12.39 Transport
+ 12.40 Unsupported + 12.40 Unsupported
+ 12.41 User-Agent + 12.41 User-Agent
+ 12.42 Vary + 12.42 Vary
+ 12.43 Via + 12.43 Via
+ 12.44 WWW-Authenticate + 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 Single Stream Container Files
+ 14.4 Playing media into an existing session + 14.4 Live Media Presentation Using Multicast
+ 14.5 Recording + 14.5 Playing media into an existing session
+ 14.6 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 Interaction with RTP * B Interaction with RTP
* C Use of SDP for RTSP Session Descriptions * C Use of SDP for RTSP Session Descriptions
+ C.1 Specification + C.1 Definitions
o C.1.1 Control URL o C.1.1 Control URL
o C.1.2 Media streams o C.1.2 Media streams
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o C.1.3 Payload type(s) o C.1.3 Payload type(s)
o C.1.4 Format specific parameters o C.1.4 Format-specific parameters
o C.1.5 Length of presentation o C.1.5 Length of presentation
o C.1.6 Time of availability o C.1.6 Time of availability
o C.1.7 Connection Information o C.1.7 Connection Information
o C.1.8 Entity Tag o C.1.8 Entity Tag
+ C.2 Scenario A + C.2 Aggregate Control Not Available
+ C.3 Scenario B + C.3 Aggregate Control Available
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* D Minimal RTSP implementation * D Minimal RTSP implementation
+ D.1 Client + D.1 Client
o D.1.1 Basic Playback o D.1.1 Basic Playback
o D.1.2 Authentication-enabled o D.1.2 Authentication-enabled
+ D.2 Server + D.2 Server
o D.2.1 Basic Playback o D.2.1 Basic Playback
o D.2.2 Authentication-enabled o D.2.2 Authentication-enabled
* E Open Issues * E Changes
* F Changes * F Author Addresses
* G Author Addresses * G Acknowledgements
* 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 10.11).
In other words, RTSP acts as a ``network remote control'' for In other words, RTSP acts as a ``network remote control'' for
multimedia servers. multimedia servers.
The set of streams to be controlled is defined by a presentation The set of streams to be controlled is defined by a presentation
description. This memorandum does not define a format for a description. This memorandum does not define a format for a
presentation description. presentation description.
There is no notion of an RTSP connection; instead, a server maintains There is no notion of an RTSP connection; instead, a server maintains
a session labeled by an identifier. An RTSP session is in no way tied a session labeled by an identifier. An RTSP session is in no way tied
to a transport-level connection such as a TCP connection. During an to a transport-level connection such as a TCP connection. During an
RTSP session, an RTSP client may open and close many reliable RTSP session, an RTSP client may open and close many reliable
transport connections to the server to issue RTSP requests. transport connections to the server to issue RTSP requests.
Alternatively, it may use a connectionless transport protocol such as Alternatively, it may use a connectionless transport protocol such as
UDP. UDP.
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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:
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* RTSP introduces a number of new methods and has a different * RTSP introduces a number of new methods and has a different
protocol identifier. protocol identifier.
* An RTSP server needs to maintain state by default in almost all * An RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP. cases, as opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests. * Both an RTSP server and client can issue requests.
* Data is carried out-of-band, by a different protocol. (There is an * Data is carried out-of-band by a different protocol. (There is an
exception to this.) exception to this.)
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts [3]. consistent with current HTML internationalization efforts [3].
* The Request-URI always contains the absolute URI. Because of * The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1 carries backward compatibility with a historical blunder, HTTP/1.1 carries
only the absolute path in the request and puts the host name in a only the absolute path in the request and puts the host name in a
separate header field. separate header field.
This makes ``virtual hosting'' easier, where a single host with one This makes ``virtual hosting'' easier, where a single host with one
IP address hosts several document trees. IP address hosts several document trees.
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provides the destination for security reasons. provides the destination for security reasons.
Invitation of a media server to a conference: Invitation of a media server to a conference:
A media server can be ``invited'' to join an existing A media server can be ``invited'' to join an existing
conference, either to play back media into the presentation or conference, either to play back media into the presentation or
to record all or a subset of the media in a presentation. This to record all or a subset of the media in a presentation. This
mode is useful for distributed teaching applications. Several mode is useful for distributed teaching applications. Several
parties in the conference may take turns ``pushing the remote parties in the conference may take turns ``pushing the remote
control buttons''. control buttons''.
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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.
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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
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greater than or equal to one. greater than or equal to one.
Client: Client:
The client requests continuous media data from the media The client requests continuous media data from the media
server. server.
Connection: Connection:
A transport layer virtual circuit established between two A transport layer virtual circuit established between two
programs for the purpose of communication. programs for the purpose of communication.
Container file:
A file which may contain multiple media streams which often
comprise a presentation when played together. RTSP servers may
offer aggregate control on these files, though the concept of a
container file is not embedded in the protocol.
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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 relationship
that existed at the source. The most common examples of that existed at the source. The most common examples of
continuous media are audio and motion video. Continuous media continuous media are audio and motion video. Continuous media
can be realtime (interactive), where there is a ``tight'' can be real-time (interactive), where there is a ``tight''
timing relationship between source and sink, or streaming timing relationship between source and sink, or streaming
(playback), where the relationship is less strict. (playback), where the relationship is less strict.
Media initialization: Media initialization:
Datatype/codec specific initialization. This includes such Datatype/codec specific initialization. This includes such
things as clockrates, color tables, etc. Any things as clockrates, color tables, etc. Any
transport-independent information which is required by a client transport-independent information which is required by a client
for playback of a media stream occurs in the media for playback of a media stream occurs in the media
initialization phase of stream setup. initialization phase of stream setup.
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Media parameter: Media parameter:
Parameter specific to a media type that may be changed while Parameter specific to a media type that may be changed before
the stream is being played or prior to it. or during stream playback.
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 server indirection: Media server indirection:
Redirection of a media client to a different media server. Redirection of a media client to a different media server.
skipping to change at line 355 skipping to change at line 371
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([19]). 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: Participant:
Participants are members of conferences. A participant may be a Member of a conference. A participant may be a machine, e.g., a
machine, e.g., a media record or playback server. media record or playback server.
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Presentation: Presentation:
A set of one or more streams presented to the client as a A set of one or more streams presented to the client as a
complete media feed, using a presentation description as complete media feed, using a presentation description as
defined below. In most cases in the RTSP context, this implies defined below. In most cases in the RTSP context, this implies
aggregate control of those streams, but doesn't have to. aggregate control of those streams, but doesn't have to.
Presentation description: Presentation description:
A presentation description contains information about one or A presentation description contains information about one or
more media streams within a presentation, such as the set of more media streams within a presentation, such as the set of
encodings, network addresses and information about the content. encodings, network addresses and information about the 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.
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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: Transport initialization:
The negotiation of transport information (i.e. port numbers, The negotiation of transport information (e.g., port numbers,
transport protocols, etc) between the client and the server. transport protocols) between the client and the server.
1.4 Protocol Properties 1.4 Protocol Properties
RTSP has the following properties: RTSP has the following properties:
Extendable: Extendable:
New methods and parameters can be easily added to RTSP. New methods and parameters can be easily added to RTSP.
Easy to parse: Easy to parse:
RTSP can be parsed by standard HTTP or MIME parsers. RTSP can be parsed by standard HTTP or MIME parsers.
H. Schulzrinne, A. Rao, R. Lanphier Page 9
Secure: Secure:
RTSP re-uses web security mechanisms, either at the transport RTSP re-uses web security mechanisms, either at the transport
level (TLS [7]) or within the protocol itself. All HTTP level (TLS [7]) or within the protocol itself. All HTTP
authentication mechanisms such as basic [5, 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.
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.
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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
skipping to change at line 451 skipping to change at line 467
format to be used. However, the presentation description must format to be used. However, the presentation description must
contain at least one RTSP URI. contain at least one RTSP URI.
Proxy and firewall friendly: Proxy and firewall friendly:
The protocol should be readily handled by both application and The protocol should be readily handled by both application and
transport-layer (SOCKS [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 reuses HTTP concepts, so that the existing
existing infrastructure can be re-used. This infrastructure infrastructure can be reused. This infrastructure includes PICS
includes PICS (Platform for Internet Content Selection [21]) (Platform for Internet Content Selection [21]) for associating
for associating labels with content. However, RTSP does not labels with content. However, RTSP does not just add methods to
just add methods to HTTP, since the controlling continuous HTTP since the controlling continuous media requires server
media requires server state in most cases. 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.
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.
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scenario. Stream identifiers can be used by several control scenario. Stream identifiers can be used by several control
streams, so that ``passing the remote'' would be possible. The streams, so that ``passing the remote'' would be possible. The
protocol would not address how several clients negotiate access; protocol would not address how several clients negotiate access;
this is left to either a ``social protocol'' or some other floor this is left to either a ``social protocol'' or some other floor
control mechanism. control mechanism.
1.5 Extending RTSP 1.5 Extending RTSP
Since not all media servers have the same functionality, media servers Since not all media servers have the same functionality, media servers
by necessity will support different sets of requests. For example: by necessity will support different sets of requests. For example:
* A server may only be capable of playback, not recording and thus * A server may only be capable of playback thus has no need to
has no need to support the RECORD request. support the RECORD request.
* A server may not be capable of seeking (absolute positioning), * A server may not be capable of seeking (absolute positioning) if
say, if it is to support live events only. it is to support live events only.
* Some servers may not support setting stream parameters and thus * Some servers may not support setting stream parameters and thus
not support GET_PARAMETER and SET_PARAMETER. not support GET_PARAMETER and SET_PARAMETER.
A server SHOULD implement all header fields described in Section 12. A server SHOULD implement all header fields described in Section 12.
It is up to the creators of presentation descriptions not to ask the It is up to the creators of presentation descriptions not to ask the
impossible of a server. This situation is similar in HTTP/1.1, 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 here in order of the
of changes supported: magnitude of changes supported:
* Existing methods can be extended with new parameters, as long as * Existing methods can be extended with new parameters, as long as
these parameters can be safely ignored by the recipient. (This is these parameters can be safely ignored by the recipient. (This is
equivalent to adding new parameters to an HTML tag.) If the client equivalent to adding new parameters to an HTML tag.) If the client
needs negative acknowledgement when a method extension is not needs negative acknowledgement when a method extension is not
supported, a tag corresponding to the extension may be added in supported, a tag corresponding to the extension may be added in
the Require: field (see Section 12.32). 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
skipping to change at line 595 skipping to change at line 611
TEARDOWN. TEARDOWN.
SETUP: SETUP:
Causes the server to allocate resources for a stream and start Causes the server to allocate resources for a stream and start
an RTSP session. an RTSP session.
PLAY and RECORD: PLAY and RECORD:
Starts data transmission on a stream allocated via SETUP. Starts data transmission on a stream allocated via SETUP.
PAUSE: PAUSE:
Temporarily halts a stream, without freeing server resources. Temporarily halts a stream without freeing server resources.
TEARDOWN: TEARDOWN:
Frees resources associated with the stream. The RTSP session Frees resources associated with the stream. The RTSP session
ceases to exist on the server. ceases to exist on the server.
1.8 Relationship with Other Protocols 1.8 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may interact RTSP has some overlap in functionality with HTTP. It also may interact
with HTTP in that the initial contact with streaming content is often with HTTP in that the initial contact with streaming content is often
to be made through a web page. The current protocol specification aims to be made through a web page. The current protocol specification aims
to allow different hand-off points between a web server and the media to allow different hand-off points between a web server and the media
server implementing RTSP. For example, the presentation description server implementing RTSP. For example, the presentation description
can be retrieved using HTTP or RTSP. Having the presentation can be retrieved using HTTP or RTSP, which reduces roundtrips in
description be returned by the web server makes it possible to have web-browser-based scenarios, yet also allows for standalone RTSP
the web server take care of authentication and billing, by handing out servers and clients which don't rely on HTTP at all.
a presentation description whose media identifier includes an
encrypted version of the requestor's IP address and a timestamp, with
a shared secret between web and media server.
However, RTSP differs fundamentally from HTTP in that data delivery However, RTSP differs fundamentally from HTTP in that data delivery
takes place out-of-band, in a different protocol. HTTP is an takes place out-of-band in a different protocol. HTTP is an asymmetric
asymmetric protocol, where the client issues requests and the server protocol where the client issues requests and the server responds. In
responds. In RTSP, both the media client and media server can issue RTSP, both the media client and media server can issue requests. RTSP
requests. RTSP requests are also not stateless, in that they may set requests are also not stateless; they may set parameters and continue
parameters and continue to control a media stream long after the to control a media stream long after the request has been
request has been acknowledged. acknowledged.
Re-using HTTP functionality has advantages in at least two areas, Re-using HTTP functionality has advantages in at least two areas,
namely security and proxies. The requirements are very similar, so namely security and proxies. The requirements are very similar, so
having the ability to adopt HTTP work on caches, proxies and having the ability to adopt HTTP work on caches, proxies and
authentication is valuable. authentication is valuable.
While most real-time media will use RTP as a transport protocol, RTSP While most real-time media will use RTP as a transport protocol, RTSP
is not tied to RTP. is not tied to RTP.
RTSP assumes the existence of a presentation description format that RTSP assumes the existence of a presentation description format that
skipping to change at line 683 skipping to change at line 696
Note that fragment and query identifiers do not have a well-defined Note that fragment and query identifiers do not have a well-defined
meaning at this time, with the interpretation left to the RTSP meaning at this time, with the interpretation left to the RTSP
server. server.
The scheme rtsp requires that commands are issued via a reliable The scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtspu identifies protocol (within the Internet, TCP), while the scheme rtspu identifies
an unreliable protocol (within the Internet, UDP). The scheme rtsps an unreliable protocol (within the Internet, UDP). The scheme rtsps
indicates that a TCP connection secured by TLS [7] must be used. indicates that a TCP connection secured by TLS [7] must be used.
If the port is empty or not given, port 554 is assumed. The semantics If the port is empty or not given, port 554 is assumed. The semantics
are that the identified resource can be controlled be RTSP at the are that the identified resource can be controlled by RTSP at the
server listening for TCP (scheme ``rtsp'') connections or UDP (scheme server listening for TCP (scheme ``rtsp'') connections or UDP (scheme
``rtspu'') packets on that port of host, and the Request-URI for the ``rtspu'') packets on that port of host, and the Request-URI for the
resource is rtsp_URL. resource is rtsp_URL.
The use of IP addresses in URLs SHOULD be avoided whenever possible The use of IP addresses in URLs SHOULD be avoided whenever possible
(see RFC 1924 [15]). (see RFC 1924 [15]).
A presentation or a stream is identified by an textual media A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions [H3.2] of identifier, using the character set and escape conventions [H3.2] of
URLs [17]. 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,
presentation. Accordingly, requests described in Section 10 can apply i.e., a presentation. Accordingly, requests described in Section 10
to either the whole presentation or an individual stream within the can apply to either the whole presentation or an individual stream
presentation. Note that some request methods can only be applied to within the presentation. Note that some request methods can only be
streams, not presentations and vice versa. applied to streams, not presentations and vice versa.
For example, the RTSP URL For example, the RTSP URL:
rtsp://media.example.com:554/twister/audiotrack rtsp://media.example.com:554/twister/audiotrack
identifies the audio stream within the presentation ``twister'', which identifies the audio stream within the presentation ``twister'', which
can be controlled via RTSP requests issued over a TCP connection to can be controlled via RTSP requests issued over a TCP connection to
port 554 of host media.example.com. port 554 of host media.example.com.
Also, the RTSP URL Also, the RTSP URL:
rtsp://media.example.com:554/twister rtsp://media.example.com:554/twister
identifies the presentation ``twister'', which may be composed of identifies the presentation ``twister'', which may be composed of
audio and video streams. audio and video streams.
This does not imply a standard way to reference streams in URLs. This does not imply a standard way to reference streams in URLs.
The presentation description defines the hierarchical relationships The presentation description defines the hierarchical relationships
in the presentation and the URLs for the individual streams. A in the presentation and the URLs for the individual streams. A
presentation description may name a stream 'a.mov' and the whole presentation description may name a stream ``a.mov'' and the whole
presentation 'b.mov'. presentation ``b.mov''.
The path components of the RTSP URL are opaque to the client and do The path components of the RTSP URL are opaque to the client and do
not imply any particular file system structure for the server. not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be used This decoupling also allows presentation descriptions to be used
with non-RTSP media control protocols, simply by replacing the with non-RTSP media control protocols simply by replacing the
scheme in the URL. scheme in the URL.
3.3 Conference Identifiers 3.3 Conference Identifiers
Conference identifiers are opaque to RTSP and are encoded using Conference identifiers are opaque to RTSP and are encoded using
standard URI encoding methods (i.e., LWS is escaped with %). They can standard URI encoding methods (i.e., LWS is escaped with %). They can
contain any octet value. The conference identifier MUST be globally contain any octet value. The conference identifier MUST be globally
unique. For H.323, the conferenceID value is to be used. unique. For H.323, the conferenceID value is to be used.
conference-id = 1*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 RTSP sessions to obtain
obtain parameters from multimedia conferences the media server is parameters from multimedia conferences the media server is
participating in. These conferences are created by protocols participating in. These conferences are created by protocols
outside the scope of this specification, e.g., H.323 [18] 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 the conference
conference participant and the initiator of the RTSP commands are participant and the initiator of the RTSP commands from being two
two different entities. different entities.
3.4 Session Identifiers 3.4 Session Identifiers
Session identifiers are opaque strings of arbitrary length. Linear Session identifiers are opaque strings of arbitrary length. Linear
white space must be URL-escaped. A session identifier SHOULD be chosen white space must be URL-escaped. A session identifier SHOULD be chosen
randomly and SHOULD be at least eight octets long to make guessing it randomly and SHOULD be at least eight octets long to make guessing it
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 timestamp expresses time relative to the start of
the clip. Relative timestamps are expressed as SMPTE time codes for the clip. Relative timestamps are expressed as SMPTE time codes for
frame-level access accuracy. The time code has the format frame-level access accuracy. The time code has the format
hours:minutes:seconds:frames.subframes, with the origin at the start hours:minutes:seconds:frames.subframes, with the origin at the start
of the clip. RTSP uses the ``SMPTE 30 drop'' format. The frame rate is of the clip. RTSP uses the ``SMPTE 30 drop'' format. The frame rate is
29.97 frames per second. The ``frames'' field in the time value can 29.97 frames per second. The ``frames'' field in the time value can
assume the values 0 through 29. The difference between 30 and 29.97 assume the values 0 through 29. The difference between 30 and 29.97
frames per second is handled by dropping the first two frame indices frames per second is handled by dropping the first two frame indices
(values 00 and 01) of every minute, except every tenth minute. If the (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 frame value is zero, it may be omitted. Subframes are measured in
one-hundredth of a frame. one-hundredth of a frame.
skipping to change at line 781 skipping to change at line 793
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
to the beginning of the presentation. The timestamp consists is a relative to the beginning of the presentation. The timestamp consists
decimal fraction. The part left of the decimal may be expressed in of a decimal fraction. The part left of the decimal may be expressed
either seconds or hours, minutes and seconds. The part right of the in either seconds or hours, minutes, and seconds. The part right of
decimal point measures fractions of a second. the decimal point measures fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative The beginning of a presentation corresponds to 0.0 seconds. Negative
values are not defined. The special constant now is defined as the values are not defined. The special constant now is defined as the
current instant of a live event. It may be used only for live events. current instant of a live event. It may be used only for live events.
NPT is defined as in DSM-CC: ``Intuitively, NPT is the clock the NPT is defined as in DSM-CC: ``Intuitively, NPT is the clock the
viewer associates with a program. It is often digitally displayed on a viewer associates with a program. It is often digitally displayed on a
VCR. NPT advances normally when in normal play mode (scale = 1), VCR. NPT advances normally when in normal play mode (scale = 1),
advances at a faster rate when in fast scan forward (high positive advances at a faster rate when in fast scan forward (high positive
scale ratio), decrements when in scan reverse (high negative scale scale ratio), decrements when in scan reverse (high negative scale
skipping to change at line 815 skipping to change at line 826
Examples: Examples:
npt=123.45-125 npt=123.45-125
npt=12:05:35.3 npt=12:05:35.3
npt=now npt=now
The syntax conforms to ISO 8601. The npt-sec notation is optimized The syntax conforms to ISO 8601. The npt-sec notation is optimized
for automatic generation, the ntp-hhmmss notation for consumption for automatic generation, the ntp-hhmmss notation for consumption
by human readers. The ``now'' constant allows clients to request to by human readers. The ``now'' constant allows clients to request to
receive the live feed rather than the stored or time-delayed receive the live feed rather than the stored or time-delayed
version. This is needed since neither absolute time, nor zero time version. This is needed since neither absolute time nor zero time
are appropriate for this case. are appropriate for this case.
3.7 Absolute Time 3.7 Absolute Time
Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
Fractions of a second may be indicated. Fractions of a second may be indicated.
utc-range = "clock" "=" utc-time "-" [ utc-time ] utc-range = "clock" "=" utc-time "-" [ utc-time ]
utc-time = utc-date "T" utc-time "Z" utc-time = utc-date "T" utc-time "Z"
utc-date = 8DIGIT ; < YYYYMMDD > utc-date = 8DIGIT ; < YYYYMMDD >
utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction > utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC: UTC:
19961108T143720.25Z 19961108T143720.25Z
3.8 Option Tags
Option tags are unique identifiers used to designate new options in
RTSP. These tags are used in correspondance with Require
(Section 12.32) and Proxy-Require (Section 12.27) fields.
Syntax:
}{
option-tag &=& 1*OCTET &; LWS must be URL-escaped
The creator of a new RTSP option should either prefix the option with
a reverse domain name (e.g., ``com.foo.mynewfeature'' is apt name for
a feature whose inventor can be reached at ``foo.com''), or register
the new option with the Internet Assigned Numbers Authority (IANA).
3.8.1 Registering New Option Tags With IANA
When registering a new RTSP option, the following information should
be provided:
* Name and description of option. The name may be of any length, but
SHOULD be no more than a twenty-character descriptive term.
* Indication of who has change control over the option (for example,
IETF, ISO, ITU-T, other international standardization bodies, a
consortium or a particular company or group of companies);
* A reference to a further description, if available, for example
(in order of preference) an RFC, a published paper, a patent
filing, a technical report, documented source code or a computer
manual;
* For proprietary options, contact information (postal and email
address);
4 RTSP Message 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
concern. Text-based protocols, if done carefully, also allow easy concern. Text-based protocols, if done carefully, also allow easy
implementation of research prototypes in scripting languages such implementation of research prototypes in scripting languages such
as Tcl, Visual Basic and Perl. as Tcl, Visual Basic and Perl.
The 10646 character set avoids tricky character set switching, but The 10646 character set avoids tricky character set switching, but
is invisible to the application as long as US-ASCII is being used. is invisible to the application as long as US-ASCII is being used.
This is also the encoding used for RTCP. ISO 8859-1 translates This is also the encoding used for RTCP. ISO 8859-1 translates
directly into Unicode, with a high-order octet of zero. ISO 8859-1 directly into Unicode with a high-order octet of zero. ISO 8859-1
characters with the most-significant bit set are represented as characters with the most-significant bit set are represented as
1100001x 10xxxxxx. 1100001x 10xxxxxx.
RTSP messages can be carried over any lower-layer transport protocol RTSP messages can be carried over any lower-layer transport protocol
that is 8-bit clean. that is 8-bit clean.
Requests contain methods, the object the method is operating upon and Requests contain methods, the object the method is operating upon and
parameters to further describe the method. Methods are idempotent, parameters to further describe the method. Methods are idempotent,
unless otherwise noted. Methods are also designed to require little or unless otherwise noted. Methods are also designed to require little or
no state maintenance at the media server. no state maintenance at the media server.
skipping to change at line 876 skipping to change at line 921
4.2 Message Headers 4.2 Message Headers
See [H4.2] See [H4.2]
4.3 Message Body 4.3 Message Body
See [H4.3] See [H4.3]
4.4 Message Length 4.4 Message Length
When a message-body is included with a message, the length of that When a message body is included with a message, the length of that
body is determined by one of the following (in order of precedence): body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message-body 1.
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. If a Content-Length header field (section 12.14) is present, 2.
If a Content-Length header field (section 12.14) is present,
its value in bytes represents the length of the message-body. its value in bytes represents the length of the message-body.
If this header field is not present, a value of zero is If this header field is not present, a value of zero is
assumed. assumed.
3. By the server closing the connection. (Closing the connection 3.
By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a that would leave no possibility for the server to send back a
response.) response.)
Note that RTSP does not (at present) support the HTTP/1.1 ``chunked'' Note that RTSP does not (at present) support the HTTP/1.1 ``chunked''
transfer coding(see [H3.6]) and requires the presence of the transfer coding(see [H3.6]) and requires the presence of the
Content-Length header field. Content-Length header field.
Given the moderate length of presentation descriptions returned, Given the moderate length of presentation descriptions returned,
the server should always be able to determine its length, even if the server should always be able to determine its length, even if
skipping to change at line 938 skipping to change at line 986
6.1 Request Line 6.1 Request Line
Request-Line = Method SP Request-URI SP RTSP-Version 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 10.10
| "REDIRECT" ; Section 10.10 | "REDIRECT" ; Section
| "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
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
skipping to change at line 974 skipping to change at line 1023
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
apply to a particular resource, but to the server itself, and is only apply to a particular resource, but to the server itself, and is only
allowed when the method used does not necessarily apply to a resource. allowed when the method used does not necessarily apply to a resource.
One example would be One example would be:
OPTIONS * RTSP/1.0 OPTIONS * RTSP/1.0
7 Response 7 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version. [H6] applies except that HTTP-Version is replaced by RTSP-Version.
Also, RTSP defines additional status codes and does not define some Also, RTSP defines additional status codes and does not define some
HTTP codes. The valid response codes and the methods they can be used HTTP codes. The valid response codes and the methods they can be used
with are defined in the table 1. with are defined in Table 1.
After receiving and interpreting a request message, the recipient After receiving and interpreting a request message, the recipient
responds with an RTSP response message. responds with an RTSP response message.
Response = Status-Line ; Section 7.1 Response = Status-Line ; Section 7.1
*( general-header ; Section 5 *( general-header ; Section 5
| response-header ; Section 7.1.2 | response-header ; Section 7.1.2
| entity-header ) ; Section 8.1 | entity-header ) ; Section 8.1
CRLF CRLF
[ message-body ] ; Section 4.3 [ message-body ] ; Section 4.3
7.1 Status-Line 7.1 Status-Line
The first line of a Response message is the Status-Line, consisting of The first line of a Response message is the Status-Line, consisting
the protocol version followed by a numeric status code, and the of the protocol version followed by a numeric status code, and the
textual phrase associated with the status code, with each element textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the final separated by SP characters. No CR or LF is allowed except in the final
CRLF sequence. CRLF sequence.
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
7.1.1 Status Code and Reason Phrase 7.1.1 Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully attempt to understand and satisfy the request. These codes are fully
defined in section11. The Reason-Phrase is intended to give a short defined in Section 11. The Reason-Phrase is intended to give a short
textual description of the Status-Code. The Status-Code is intended textual description of the Status-Code. The Status-Code is intended
for use by automata and the Reason-Phrase is intended for the human for use by automata and the Reason-Phrase is intended for the human
user. The client is not required to examine or display the user. The client is not required to examine or display the
Reason-Phrase. Reason-Phrase.
The first digit of the Status-Code defines the class of response. The The first digit of the Status-Code defines the class of response. The
last two digits do not have any categorization role. There are 5 last two digits do not have any categorization role. There are 5
values for the first digit: values for the first digit:
* 1xx: Informational - Request received, continuing process * 1xx: Informational - Request received, continuing process
skipping to change at line 1039 skipping to change at line 1089
RTSP/1.0, and an example set of corresponding Reason-Phrase's, are RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
presented below. The reason phrases listed here are only recommended - presented below. The reason phrases listed here are only recommended -
they may be replaced by local equivalents without affecting the they may be replaced by local equivalents without affecting the
protocol. Note that RTSP adopts most HTTP/1.1 status codes and adds protocol. Note that RTSP adopts most HTTP/1.1 status codes and adds
RTSP-specific status codes in the starting at 450 to avoid conflicts RTSP-specific status codes in the starting at 450 to avoid conflicts
with newly defined HTTP status codes. with newly defined HTTP status codes.
Status-Code = "100" ; Continue Status-Code = "100" ; Continue
| "200" ; OK | "200" ; OK
| "201" ; Created | "201" ; Created
| "250" ; Low on Storage Space
| "300" ; Multiple Choices | "300" ; Multiple Choices
| "301" ; Moved Permanently | "301" ; Moved Permanently
| "302" ; Moved Temporarily | "302" ; Moved Temporarily
| "303" ; See Other | "303" ; See Other
| "304" ; Not Modified | "304" ; Not Modified
| "305" ; Use Proxy | "305" ; Use Proxy
| "400" ; Bad Request | "400" ; Bad Request
| "401" ; Unauthorized | "401" ; Unauthorized
| "402" ; Payment Required | "402" ; Payment Required
| "403" ; Forbidden | "403" ; Forbidden
| "404" ; Not Found | "404" ; Not Found
| "405" ; Method Not Allowed | "405" ; Method Not Allowed
| "406" ; Not Acceptable | "406" ; Not Acceptable
| "407" ; Proxy Authentication Required | "407" ; Proxy Authentication Required
| "408" ; Request Time-out | "408" ; Request Time-out
| "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
| "454" ; Session Not Found | "454" ; Session Not Found
| "455" ; Method Not Valid in This State | "455" ; Method Not Valid in This State
| "456" ; Header Field Not Valid for Resource | "456" ; Header Field Not Valid for Resource
| "457" ; Invalid Range | "457" ; Invalid Range
| "458" ; Parameter Is Read-Only | "458" ; Parameter Is Read-Only
| "459" ; Aggregate operation not allowed | "459" ; Aggregate operation not allowed
| "460" ; Only aggregate operation allowed | "460" ; Only aggregate operation allowed
| "461" ; Unsupported transport
| "462" ; Destination unreachable
| "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
| "551" ; Option not supported
| extension-code | extension-code
extension-code = 3DIGIT extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF> Reason-Phrase = *<TEXT, excluding CR, LF>
RTSP status codes are extensible. RTSP applications are not required RTSP status codes are extensible. RTSP applications are not required
to understand the meaning of all registered status codes, though such to understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an unrecognized x00 status code of that class, with the exception that an unrecognized
skipping to change at line 1102 skipping to change at line 1156
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
250 Low on Storage Space RECORD
300 Multiple Choices all 300 Multiple Choices all
301 Moved Permanently all 301 Moved Permanently all
302 Moved Temporarily all 302 Moved Temporarily all
303 See Other all 303 See Other all
305 Use Proxy all 305 Use Proxy all
400 Bad Request all 400 Bad Request all
401 Unauthorized all 401 Unauthorized all
402 Payment Required all 402 Payment Required all
403 Forbidden all 403 Forbidden all
404 Not Found all 404 Not Found all
405 Method Not Allowed all 405 Method Not Allowed all
406 Not Acceptable all 406 Not Acceptable all
407 Proxy Authentication Required all 407 Proxy Authentication Required all
408 Request Timeout all 408 Request Timeout all
409 Conflict RECORD
410 Gone all 410 Gone all
411 Length Required SETUP 411 Length Required all
412 Precondition Failed DESCRIBE, SETUP 412 Precondition Failed DESCRIBE, SETUP
413 Request Entity Too Large SETUP 413 Request Entity Too Large all
414 Request-URI Too Long all 414 Request-URI Too Long all
415 Unsupported Media Type SETUP 415 Unsupported Media Type all
451 Invalid parameter SETUP 451 Invalid parameter SETUP
452 Illegal Conference Identifier SETUP 452 Illegal Conference Identifier SETUP
453 Not Enough Bandwidth SETUP 453 Not Enough Bandwidth SETUP
454 Session not found all 454 Session Not Found all
455 Method Not Valid In This State all 455 Method Not Valid In This State all
456 Header Field Not Valid all 456 Header Field Not Valid all
457 Invalid Range PLAY 457 Invalid Range PLAY
458 Parameter Is Read-Only SET_PARAMETER 458 Parameter Is Read-Only SET_PARAMETER
459 Aggregate operation not allowed all 459 Aggregate Operation Not Allowed all
460 Only aggregate operation allowed all 460 Only Aggregate Operation Allowed all
461 Unsupported Transport all
462 Destination Unreachable 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
551 Option not support all
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.
skipping to change at line 1214 skipping to change at line 1271
RTSP requests can be transmitted in several different ways: RTSP requests can be transmitted in several different ways:
* persistent transport connections used for several request-response * persistent transport connections used for several request-response
transactions; transactions;
* one connection per request/response transaction; * one connection per request/response transaction;
* connectionless mode. * connectionless mode.
The type of transport connection is defined by the RTSP URI The type of transport connection is defined by the RTSP URI
(Section 3.2). For the scheme ``rtsp'', a persistent connection is (Section 3.2). For the scheme ``rtsp'', a persistent connection is
assumed, while the scheme ``rtspu'' calls for RTSP requests to be send assumed, while the scheme ``rtspu'' calls for RTSP requests to be sent
without setting up a connection. without setting up a connection.
Unlike HTTP, RTSP allows the media server to send requests to the Unlike HTTP, RTSP allows the media server to send requests to the
media client. However, this is only supported for persistent media client. However, this is only supported for persistent
connections, as the media server otherwise has no reliable way of connections, as the media server otherwise has no reliable way of
reaching the client. Also, this is the only way that requests from reaching the client. Also, this is the only way that requests from
media server to client are likely to traverse firewalls. media server to client are likely to traverse firewalls.
9.1 Pipelining 9.1 Pipelining
skipping to change at line 1243 skipping to change at line 1300
multicast group. If there is no acknowledgement, the sender may resend multicast group. If there is no acknowledgement, the sender may resend
the same message after a timeout of one round-trip time (RTT). The the same message after a timeout of one round-trip time (RTT). The
round-trip time is estimated as in TCP (RFC TBD), with an initial round-trip time is estimated as in TCP (RFC TBD), with an initial
round-trip value of 500 ms. An implementation MAY cache the last RTT round-trip value of 500 ms. An implementation MAY cache the last RTT
measurement as the initial value for future connections. If a reliable measurement as the initial value for future connections. If a reliable
transport protocol is used to carry RTSP, the timeout value MAY be set transport protocol is used to carry RTSP, the timeout value MAY be set
to an arbitrarily large value. to an arbitrarily large value.
This can greatly increase responsiveness for proxies operating in This can greatly increase responsiveness for proxies operating in
local-area networks with small RTTs. The mechanism is defined such local-area networks with small RTTs. The mechanism is defined such
that the client implementation does not have be aware of whether a that the client implementation does not have to be aware of whether
reliable or unreliable transport protocol is being used. It is a reliable or unreliable transport protocol is being used. It is
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.
skipping to change at line 1312 skipping to change at line 1369
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 nonstandard request. It does not influence server state.
Example : Example :
C->S: OPTIONS * RTSP/1.0 C->S: OPTIONS * RTSP/1.0
CSeq: 1 CSeq: 1
Require: implicit-play Require: implicit-play
Proxy-Require: gzipped-messages Proxy-Require: gzipped-messages
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are necessarily fictional features (one would hope Note that these are necessarily fictional features (one would hope
that we would not purposefully overlook a truly useful feature just so that we would not purposefully overlook a truly useful feature just so
that we could have a strong example in this section). that we could have a strong example in this section).
DESCRIBE 10.2 DESCRIBE
The DESCRIBE method retrieves the description of a presentation or The DESCRIBE method retrieves the description of a presentation or
media object identified by the request URL from a server. It may use media object identified by the request URL from a server. It may use
the Accept header to specify the description formats that the client the Accept header to specify the description formats that the client
understands. The server responds with a description of the requested understands. The server responds with a description of the requested
resource. resource. The DESCRIBE reply-response pair constitutes the media
initialization phase of RTSP.
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 C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312 CSeq: 312
Accept: application/sdp, application/rtsl, application/mheg Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 312 CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
skipping to change at line 1381 skipping to change at line 1439
of media indirection. of media indirection.
Clear ground rules need to be established so that clients have an Clear ground rules need to be established so that clients have an
unambiguous means of knowing when to request media initialization unambiguous means of knowing when to request media initialization
information via DESCRIBE, and when not to. By forcing a DESCRIBE information via DESCRIBE, and when not to. By forcing a DESCRIBE
response to contain all media initialization for the set of streams response to contain all media initialization for the set of streams
that it describes, and discouraging use of DESCRIBE for media that it describes, and discouraging use of DESCRIBE for media
indirection, we avoid looping problems that might result from other indirection, we avoid looping problems that might result from other
approaches. approaches.
ANNOUNCE Media initialization is a requirement for any RTSP-based system,
but the RTSP specification doesn't dictate that this must be done
via the DESCRIBE method. There are three ways that an RTSP client
may receive initialization information:
* Via RTSP's DESCRIBE method
* Via some other protocol (HTTP, email attachment, etc.)
* Via the command line or standard input (thus working as a browser
helper application launched with an SDP file or other media
initialization format)
In the interest of practical interoperability, it is highly
recommended that minimal servers support the DESCRIBE method, and
highly recommended that minimal clients support the ability to act
as a ``helper application'' that accepts a media initialization
file from standard input, command line, and/or other means that are
appropriate to the operating environment of the client.
10.3 ANNOUNCE
The ANNOUNCE method serves two purposes: The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a When sent from client to server, ANNOUNCE posts the description of a
presentation or media object identified by the request URL to a presentation or media object identified by the request URL to a
server. When sent from server to client, ANNOUNCE updates the session server. When sent from server to client, ANNOUNCE updates the session
description in real-time. description in real-time.
If a new media stream is added to a presentation (e.g., during a live If a new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent presentation), the whole presentation description should be sent
skipping to change at line 1419 skipping to change at line 1495
e=mjh@isi.edu (Mark Handley) e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127 c=IN IP4 224.2.17.12/127
t=2873397496 2873404696 t=2873397496 2873404696
a=recvonly a=recvonly
m=audio 3456 RTP/AVP 0 m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31 m=video 2232 RTP/AVP 31
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 312 CSeq: 312
SETUP 10.4 SETUP
The SETUP request for a URI specifies the transport mechanism to be The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. A client can issue a SETUP request for a used for the streamed media. A client can issue a SETUP request for a
stream that is already playing to change transport parameters, which a stream that is already playing to change transport parameters, which a
server MAY allow(If it does not allow it, it must respond with error server MAY allow. If it does not allow this, it MUST respond with
``455 Method not valid in this state'' ). For the benefit of any error ``455 Method Not Valid In This State''. For the benefit of any
intervening firewalls, a client must indicate the transport parameters intervening firewalls, a client must indicate the transport 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 Since SETUP includes all transport initialization information,
transport information in SETUP avoids having firewall to parse firewalls and other intermediate network devices (which need this
numerous different presentation description formats for information information) are spared the more arduous task of parsing the
which is irrelevant to transport. DESCRIBE response, which has been reserved for media
initialization.
The Transport header specifies the transport parameters acceptable to The Transport header specifies the transport parameters acceptable to
the client for data transmission; the response will contain the the client for data transmission; the response will contain the
transport parameters selected by the server. transport parameters selected by the server.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302 CSeq: 302
Transport: RTP/AVP;port=4588 Transport: RTP/AVP;unicast;client_port=4588-4589
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 302 CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
Transport: RTP/AVP;port=4588 Session: 4711
Transport: RTP/AVP;unicast;client_port=4588-4589
;server_port=6256-6257
PLAY 10.5 PLAY
The PLAY method tells the server to start sending data via the The PLAY method tells the server to start sending data via the
mechanism specified in SETUP. A client MUST NOT issue a PLAY request mechanism specified in SETUP. A client MUST NOT issue a PLAY request
until any outstanding SETUP requests have been acknowledged as until any outstanding SETUP requests have been acknowledged as
successful. successful.
The PLAY request positions the normal play time to the beginning of The PLAY request positions the normal play time to the beginning of
the range specified and delivers stream data until the end of the the range specified and delivers stream data until the end of the
range is reached. PLAY requests may be pipelined (queued); a server range is reached. PLAY requests may be pipelined (queued); a server
MUST queue PLAY requests to be executed in order. That is, a PLAY MUST queue PLAY requests to be executed in order. That is, a PLAY
request arriving while a previous PLAY request is still active is request arriving while a previous PLAY request is still active is
delayed until the first has been completed. delayed until the first has been completed.
This allows precise editing. This allows precise editing.
For example, regardless of how closely spaced the two PLAY 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 first play seconds 10
15 and then, immediately following, seconds 20 to 25 and finally through 15, 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 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 835 CSeq: 835
Range: npt=10-15 Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 836 CSeq: 836
Range: npt=20-25 Range: npt=20-25
skipping to change at line 1491 skipping to change at line 1570
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
message is received after the specified time, playback is started message is received after the specified time, playback is started
immediately. The time parameter may be used to aid in synchronisation immediately. The time parameter may be used to aid in synchronization
of streams obtained from different sources. of streams obtained from different sources.
For a on-demand stream, the server replies back with the actual range For a on-demand stream, the server replies with the actual range that
that will be played back. This may differ from the requested range if will be played back. This may differ from the requested range if
alignment of the requested range to valid frame boundaries is required alignment of the requested range to valid frame boundaries is required
for the media source. If no range is specified in the request, the for the media source. If no range is specified in the request, the
current position is returned in the reply. The unit of the range in 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
skipping to change at line 1531 skipping to change at line 1610
CSeq: 835 CSeq: 835
Range: clock=19961108T142300Z-19961108T143520Z Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 835 CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the npt format A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats. and MAY support the clock and smpte formats.
PAUSE 10.6 PAUSE
The PAUSE request causes the stream delivery to be interrupted The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only playback (halted) temporarily. If the request URL names a stream, only 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
is equivalent to muting. If the request URL names a presentation or is equivalent to muting. If the request URL names a presentation or
group of streams, delivery of all currently active streams within the group of streams, delivery of all currently active streams within the
presentation or group is halted. After resuming playback or recording, presentation or group is halted. After resuming playback or recording,
synchronization of the tracks MUST be maintained. Any server resources synchronization of the tracks MUST be maintained. Any server resources
are kept. are kept.
The PAUSE request may contain a Range header specifying when the The PAUSE request may contain a Range header specifying when the
stream or presentation is to be halted. The header must contain stream or presentation is to be halted. The header must contain
exactly one value rather than a time range. The normal play time for exactly one value rather than a time range. The normal play time for
the stream is set to that value. The pause request becomes effective the stream is set to that value. The pause request becomes effective
the first time the server is encountering the time point specified. If the first time the server is encountering the time point specified in
this header is missing, stream delivery is interrupted immediately on any of the currently pending PLAY requests. If the Range header
receipt of the message. specifies a time outside any currently pending PLAY requests, the
error ``457 Invalid Range'' is returned. If this header is missing,
stream delivery is interrupted immediately on receipt of the message.
For example, if the server has play requests for ranges 10 to 15 and For example, if the server has play requests for ranges 10 to 15 and
20 to 29 pending and then receives a pause request for NPT 21, it 20 to 29 pending and then receives a pause request for NPT 21, it
would start playing the second range and stop at NPT 21. If the pause would start playing the second range and stop at NPT 21. If the pause
request is for NPT 12 and the server is playing at NPT 13 serving the request is for NPT 12 and the server is playing at NPT 13 serving the
first play request, it stops immediately. If the pause request is for first play request, the server stops immediately. If the pause request
NPT 16, it stops after completing the first play request and discards is for NPT 16, the server stops after completing the first play
the second play request. request and discards the second play request.
As another example, if a server has received requests to play ranges As another example, if a server has received requests to play ranges
10 to 15 and then 13 to 20, that is, overlapping ranges, the PAUSE 10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
request for NPT=14 would take effect while playing the first range, request for NPT=14 would take effect while the server plays the first
with the second PLAY request effectively being ignored, assuming the range, with the second PLAY request effectively being ignored,
PAUSE request arrives before the server has started playing the assuming the PAUSE request arrives before the server has started
second, overlapping range. Regardless of when the PAUSE request playing the second, overlapping range. Regardless of when the PAUSE
arrives, it sets the NPT to 14. request arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the If the server has already sent data beyond the time specified in the
Range header, a PLAY would still resume at that point in time, as it Range header, a PLAY would still resume at that point in time, as it
is assumed that the client has discarded data after that point. This is assumed that the client has discarded data after that point. This
ensures continuous pause/play cycling without gaps. ensures continuous pause/play cycling without gaps.
Example: Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834 CSeq: 834
Session: 1234 Session: 1234
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 834 CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT Date: 23 Jan 1997 15:35:06 GMT
TEARDOWN 10.7 TEARDOWN
Stop the stream delivery for the given URI, freeing the resources The TEARDOWN request stops the stream delivery for the given URI,
associated with it. If the URI is the presentation URI for this freeing the resources associated with it. If the URI is the
presentation, any RTSP session identifier associated with the session presentation URI for this presentation, any RTSP session identifier
is no longer valid. Unless all transport parameters are defined by the associated with the session is no longer valid. Unless all transport
session description, a SETUP request has to be issued before the parameters are defined by the session description, a SETUP request has
session can be played again. to be issued before the session can be played again.
Example: Example:
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 892 CSeq: 892
Session: 1234 Session: 1234
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 892 CSeq: 892
GET_PARAMETER 10.8 GET_PARAMETER
The requests retrieves the value of a parameter of a presentation or The GET_PARAMETER request retrieves the value of a parameter of a
stream specified in the URI. Multiple parameters can be requested in presentation or stream specified in the URI.
the message body using the content type text/rtsp-parameters. Note
that parameters include server and client statistics. IANA registers The content of the reply and response is left to the implementation.
parameter names for statistics and other purposes. GET_PARAMETER with GET_PARAMETER with no entity body may be used to test client or server
no entity body may be used to test client or server liveness liveness (``ping'').
(``ping'').
Example: Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 431 CSeq: 431
Content-Type: text/rtsp-parameters Content-Type: text/x-rtsp-parameters
Session: 1234 Session: 1234
Content-Length: 15 Content-Length: 15
packets_received packets_received
jitter jitter
C->S: RTSP/1.0 200 OK C->S: RTSP/1.0 200 OK
CSeq: 431 CSeq: 431
Content-Length: 46 Content-Length: 46
Content-Type: text/rtsp-parameters Content-Type: text/rtsp-parameters
packets_received: 10 packets_received: 10
jitter: 0.3838 jitter: 0.3838
SET_PARAMETER The ``text/x-rtsp-parameters'' section is only an example type for
parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be
defined after further experimentation.
10.9 SET_PARAMETER
This method requests to set the value of a parameter for a This method requests to set the value of a parameter for a
presentation or stream specified by the URI. presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client A request SHOULD only contain a single parameter to allow the client
to determine why a particular request failed. A server MUST allow a to determine why a particular request failed. A server MUST allow a
parameter to be set repeatedly to the same value, but it MAY disallow parameter to be set repeatedly to the same value, but it MAY disallow
changing parameter values. changing parameter values.
Note: transport parameters for the media stream MUST only be set with Note: transport parameters for the media stream MUST only be set with
skipping to change at line 1658 skipping to change at line 1743
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 C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 421 CSeq: 421
Content-type: text/rtsp-parameters Content-type: text/x-rtsp-parameters
barparam: barstuff barparam: barstuff
S->C: RTSP/1.0 450 Invalid Parameter S->C: RTSP/1.0 451 Invalid Parameter
CSeq: 421 CSeq: 421
Content-Length: 6 Content-Length: 6
barparam barparam
REDIRECT The ``text/x-rtsp-parameters'' section is only an example type for
parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be
defined after further experimentation.
10.10 REDIRECT
A redirect request informs the client that it must connect to 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. If the client wants to continue to send or receive media
for this URI, the client MUST issue a TEARDOWN request for the current
session and a SETUP for the new session at the designated host.
This example request redirects traffic for this URI to the new server This example request redirects traffic for this URI to the new server
at the given play time: at the given play time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0 S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 732 CSeq: 732
Location: rtsp://bigserver.com:8001 Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z- Range: clock=19960213T143205Z-
RECORD 10.11 RECORD
This method initiates recording a range of media data according to This method initiates recording a range of media data according to
the presentation description. The timestamp reflects start and end the presentation description. The timestamp reflects start and end
time (UTC). If no time range is given, use the start or end time time (UTC). If no time range is given, use the start or end time
provided in the presentation description. If the session has already provided in the presentation description. If the session has already
started, commence recording immediately. started, commence recording immediately.
The server decides whether to store the recorded data under the The server decides whether to store the recorded data under the
request-URI or another URI. If the server does not use the request-URI or another URI. If the server does not use the
request-URI, the response SHOULD be 201 (Created) and contain an request-URI, the response SHOULD be 201 (Created) and contain an
skipping to change at line 1752 skipping to change at line 1844
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 3 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 reused. Status codes
that have the same meaning are not repeated here. See Table 1 for a that have the same meaning are not repeated here. See Table 1 for a
listing of which status codes may be returned by which request. listing of which status codes may be returned by which requests.
11.1 Redirection 3xx 11.1 Success 2xx
11.1.1 250 Low on Storage Space
The server returns this warning after receiving a RECORD request that
it may not be able to fulfill completely due to insufficient storage
space. If possible, the server should use the Range header to indicate
what time period it may still be able to record. Since other processes
on the server may be consuming storage space simultaneously, a client
should take this only as an estimate.
11.2 Redirection 3xx
See [H10.3]. See [H10.3].
Within RTSP, redirection may be used for load balancing or redirecting Within RTSP, redirection may be used for load balancing or redirecting
stream requests to a server topologically closer to the client. stream requests to a server topologically closer to the client.
Mechanisms to determine topological proximity are beyond the scope of Mechanisms to determine topological proximity are beyond the scope of
this specification. this specification.
11.2 Client Error 4xx 11.3 Client Error 4xx
11.2.1 405 Method Not Allowed 11.3.1 405 Method Not Allowed
The method specified in the request is not allowed for the resource The method specified in the request is not allowed for the resource
identified by the request URI. The response MUST include an Allow identified by the request URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource. header containing a list of valid methods for the requested resource.
This status code is also to be used if a request attempts to use a This status code is also to be used if a request attempts to use a
method not indicated during SETUP, e.g., if a RECORD request is issued method not indicated during SETUP, e.g., if a RECORD request is issued
even though the mode parameter in the Transport header only specified even though the mode parameter in the Transport header only specified
PLAY. PLAY.
11.2.2 451 Parameter Not Understood 11.3.2 451 Parameter Not Understood
The recipient of the request does not support one or more parameters The recipient of the request does not support one or more parameters
contained in the request. contained in the request.
11.2.3 452 Conference Not Found 11.3.3 452 Conference Not Found
The conference indicated by a Conference header field is unknown to The conference indicated by a Conference header field is unknown to
the media server. the media server.
11.2.4 453 Not Enough Bandwidth 11.3.4 453 Not Enough Bandwidth
The request was refused since there was insufficient bandwidth. This The request was refused because 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.
11.2.5 454 Session Not Found 11.3.5 454 Session Not Found
The RTSP session identifier is invalid or has timed out. The RTSP session identifier in the Session header is missing, invalid,
or has timed out.
11.2.6 455 Method Not Valid in This State 11.3.6 455 Method Not Valid in This State
The client or server cannot process this request in its current state. The client or server cannot process this request in its current state.
11.2.7 456 Header Field Not Valid for Resource 11.3.7 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example, if The server could not act on a required request header. For example, 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.
11.2.8 457 Invalid Range 11.3.8 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the The Range value given is out of bounds, e.g., beyond the end of the
presentation. presentation.
11.2.9 458 Parameter Is Read-Only 11.3.9 458 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 be read but not modified.
modified.
11.2.10 459 Aggregate operation not allowed 11.3.10 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URL in question since The requested method may not be applied on the URL in question since
it is an aggregate(presentation) URL. The method may be applied on a it is an aggregate(presentation) URL. The method may be applied on a
stream URL. stream URL.
11.2.11 460 Only aggregate operation allowed 11.3.11 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URL in question since The requested method may not be applied on the URL in question since
it is not an aggregate(presentation) URL. The method may be applied on it is not an aggregate (presentation) URL. The method may be applied
the presentation URL. on the presentation URL.
11.3.12 461 Unsupported Transport
The Transport field did not contain a supported transport
specification.
11.3.13 462 Destination Unreachable
The data transmission channel could not be established because the
client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid Destination
parameter in the Transport field.
11.3.14 551 Option not supported
An option given in the Require or the Proxy-Require fields was not
supported.
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.
Tables 3 summarizes the header fields used by RTSP. Type ``g'' Table 3 summarizes the header fields used by RTSP. Type ``g''
designates general request headers, to be found in both requests and designates general request headers to be found in both requests and
responses, type ``R'' designates request headers, type ``r'' response responses, type ``R'' designates request headers, type ``r''
headers, type ``e'' entity header fields. Fields marked with ``req.'' designates response headers, and type ``e'' designates entity header
in the column labeled ``support'' MUST be implemented by the recipient fields. Fields marked with ``req.'' in the column labeled ``support''
for a particular method, while fields marked ``opt.'' are optional. MUST be implemented by the recipient for a particular method, while
Note that not all fields marked 'r' will be send in every request of fields marked ``opt.'' are optional. Note that not all fields marked
this type; merely, that client (for response headers) and server (for 'r' will be sent in every request of this type. The ``r'' means only
request headers) MUST implement them. The last column lists the method that client (for response headers) and server (for request headers)
for which this header field is meaningful; the designation ``entity'' MUST implement the fields. The last column lists the method for which
refers to all methods that return a message body. Within this this header field is meaningful; the designation ``entity'' refers to
specification, DESCRIBE and GET_PARAMETER fall into this class. all methods that return a message body. Within this specification,
DESCRIBE and GET_PARAMETER fall into this class.
If the field content does not apply to the particular resource, the If the field content does not apply to the particular resource, the
server MUST return status 456 (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
skipping to change at line 1953 skipping to change at line 2074
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
the media specific size if necessary. The block size is a strictly with the media-specific size if necessary. The block size MUST be a
positive decimal number and measured in octets. The server only positive decimal number, measured in octets. The server only returns
returns an error (416) if the value is syntactically invalid. an error (416) if the value is syntactically invalid.
12.8 Cache-Control 12.8 Cache-Control
The Cache-Control general header field is used to specify directives The Cache-Control general header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the that MUST be obeyed by all caching mechanisms along the
request/response chain. request/response chain.
Cache directives must be passed through by a proxy or gateway Cache directives must be passed through by a proxy or gateway
application, regardless of their significance to that application, application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the since the directives may be applicable to all recipients along the
skipping to change at line 1977 skipping to change at line 2098
request/response chain. It is not possible to specify a cache- request/response chain. It is not possible to specify a cache-
directive for a specific cache. directive for a specific cache.
Cache-Control should only be specified in a SETUP request and its Cache-Control should only be specified in a SETUP request and its
response. Note: Cache-Control does not govern the caching of responses response. Note: Cache-Control does not govern the caching of responses
as for HTTP, but rather of the stream identified by the SETUP request. as for HTTP, but rather of the stream identified by the SETUP request.
Responses to RTSP requests are not cacheable, except for responses to Responses to RTSP requests are not cacheable, except for responses to
DESCRIBE. DESCRIBE.
Cache-Control = "Cache-Control" ":" 1#cache-directive Cache-Control = "Cache-Control" ":" 1#cache-directive
cache-directive = cache-request-directive cache-directive = cache-request-directive
| cache-response-directive | cache-response-directive
cache-request-directive = "no-cache"
cache-request-directive =
"no-cache"
| "max-stale" | "max-stale"
| "min-fresh" | "min-fresh"
| "only-if-cached" | "only-if-cached"
| cache-extension | cache-extension
cache-response-directive = "public"
cache-response-directive =
"public"
| "private" | "private"
| "no-cache" | "no-cache"
| "no-transform" | "no-transform"
| "must-revalidate" | "must-revalidate"
| "proxy-revalidate" | "proxy-revalidate"
| "max-age" "=" delta-seconds | "max-age" "=" delta-seconds
| cache-extension | cache-extension
cache-extension = token [ "=" ( token | quoted-string ) ] cache-extension = token [ "=" ( token | quoted-string ) ]
no-cache: no-cache:
Indicates that the media stream MUST NOT be cached anywhere. Indicates that the media stream MUST NOT be cached anywhere.
This allows an origin server to prevent caching even by caches This allows an origin server to prevent caching even by caches
that have been configured to return stale responses to client that have been configured to return stale responses to client
requests. requests.
public: public:
Indicates that the media stream is cachable by any cache. Indicates that the media stream is cacheable by any cache.
private: private:
Indicates that the media stream is intended for a single user Indicates that the media stream is intended for a single user
and MUST NOT be cached by a shared cache. A private and MUST NOT be cached by a shared cache. A private
(non-shared) cache may cache the media stream. (non-shared) cache may cache the media stream.
no-transform: no-transform:
An intermediate cache (proxy) may find it useful to convert the An intermediate cache (proxy) may find it useful to convert the
media type of certain stream. A proxy might, for example, media type of a certain stream. A proxy might, for example,
convert between video formats to save cache space or to reduce convert between video formats to save cache space or to reduce
the amount of traffic on a slow link. Serious operational the amount of traffic on a slow link. Serious operational
problems may occur, however, when these transformations have problems may occur, however, when these transformations have
been applied to streams intended for certain kinds of been applied to streams intended for certain kinds of
applications. For example, applications for medical imaging, applications. For example, applications for medical imaging,
scientific data analysis and those using end-to-end scientific data analysis and those using end-to-end
authentication, all depend on receiving a stream that is bit authentication all depend on receiving a stream that is
for bit identical to the original entity-body. Therefore, if a bit-for-bit identical to the original entity-body. Therefore,
response includes the no-transform directive, an intermediate if a response includes the no-transform directive, an
cache or proxy MUST NOT change the encoding of the stream. intermediate cache or proxy MUST NOT change the encoding of the
Unlike HTTP, RTSP does not provide for partial transformation stream. Unlike HTTP, RTSP does not provide for partial
at this point, e.g., allowing translation into a different transformation at this point, e.g., allowing translation into a
language. different language.
only-if-cached: only-if-cached:
In some cases, such as times of extremely poor network In some cases, such as times of extremely poor network
connectivity, a client may want a cache to return only those connectivity, a client may want a cache to return only those
media streams that it currently has stored, and not to receive media streams that it currently has stored, and not to receive
these from the origin server. To do this, the client may these from the origin server. To do this, the client may
include the only-if-cached directive in a request. If it include the only-if-cached directive in a request. If it
receives this directive, a cache SHOULD either respond using a receives this directive, a cache SHOULD either respond using a
cached media stream that is consistent with the other cached media stream that is consistent with the other
constraints of the request, or respond with a 504 (Gateway constraints of the request, or respond with a 504 (Gateway
skipping to change at line 2063 skipping to change at line 2178
Indicates that the client is willing to accept a media stream Indicates that the client is willing to accept a media stream
whose freshness lifetime is no less than its current age plus whose freshness lifetime is no less than its current age plus
the specified time in seconds. That is, the client wants a the specified time in seconds. That is, the client wants a
response that will still be fresh for at least the specified response that will still be fresh for at least the specified
number of seconds. number of seconds.
must-revalidate: must-revalidate:
When the must-revalidate directive is present in a SETUP When the must-revalidate directive is present in a SETUP
response received by a cache, that cache MUST NOT use the entry response received by a cache, that cache MUST NOT use the 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. That is,
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.)
12.9 Conference 12.9 Conference
This request header field establishes a logical connection between a This request header field establishes a logical connection between a
conference, established using non-RTSP means, and an RTSP stream. The pre-established conference and an RTSP stream. The conference-id must
conference-id must not be changed for the same RTSP session. not be changed for the same RTSP session.
Conference = "Conference" ":" conference-id Conference = "Conference" ":" conference-id
Example: Example:
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
A response code of 452 (452 Conference Not Found) is returned if the A response code of 452 (452 Conference Not Found) is returned if the
conference-id is not valid. conference-id is not valid.
12.10 Connection 12.10 Connection
skipping to change at line 2105 skipping to change at line 2220
12.13 Content-Language 12.13 Content-Language
See [H14.13] See [H14.13]
12.14 Content-Length 12.14 Content-Length
This field contains the length of the content of the method (i.e. This field contains the length of the content of the method (i.e.
after the double CRLF following the last header). Unlike HTTP, it MUST after the double CRLF following the last header). Unlike HTTP, it 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. If it is missing, a default value of zero is
assumed. It is interpreted according to [H14.14].
12.15 Content-Location 12.15 Content-Location
See [H14.15] See [H14.15]
12.16 Content-Type 12.16 Content-Type
See [H14.18]. Note that the content types suitable for RTSP are See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and likely to be restricted in practice to presentation descriptions and
parameter-value types. parameter-value types.
12.17 CSeq 12.17 CSeq
This field is a mandatory field that specifies the sequence number The CSeq field specifies the sequence number for an RTSP
for an RTSP request-response pair. For every RTSP request containing request-response pair. This field MUST be present in all requests and
the given sequence number, there will be a corresponding response responses. For every RTSP request containing the given sequence
having the same number. number, there will be a corresponding response having the same number.
12.18 Date 12.18 Date
See [H14.19]. See [H14.19].
12.19 Expires 12.19 Expires
The Expires entity-header field gives a date and time after which The Expires entity-header field gives a date and time after which
the description or media-stream should be considered stale. the description or media-stream should be considered stale. The
interpretation depends on the method:
The interpretation depends on the method:
DESCRIBE response: DESCRIBE response:
The Expires header indicates a date and time after which the The Expires header indicates a date and time after which the
description should be considered stale. description should be considered stale.
A stale cache entry may not normally be returned by a cache (either a A stale cache entry may not normally be returned by a cache (either a
proxy cache or an user agent cache) unless it is first validated with proxy cache or an user agent cache) unless it is first validated with
the origin server (or with an intermediate cache that has a fresh copy the origin server (or with an intermediate cache that has a fresh copy
of the entity). See section 13 for further discussion of the of the entity). See section 13 for further discussion of the
expiration model. expiration model.
skipping to change at line 2158 skipping to change at line 2273
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 having occured in the past
expired''). (i.e., ``already expired'').
To mark a response as ``already expired,'' an origin server should use To mark a response as ``already expired,'' an origin server should use
an Expires date that is equal to the Date header value. To mark a an Expires date that is equal to the Date header value. To mark a
response as ``never expires,'' an origin server should use an Expires response as ``never expires,'' an origin server should use an Expires
date approximately one year from the time the response is sent. date approximately one year from the time the response is sent.
RTSP/1.0 servers should not send Expires dates more than one year in RTSP/1.0 servers should not send Expires dates more than one year in
the future. the future.
The presence of an Expires header field with a date value of some time The presence of an Expires header field with a date value of some 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 cacheable, unless
indicated otherwise by a Cache-Control header field (Section 12.8). indicated otherwise by a Cache-Control header field (Section 12.8).
12.20 From 12.20 From
See [H14.22]. See [H14.22].
12.21 Host 12.21 Host
This HTTP request header field is not needed for RTSP. It should be This HTTP request header field is not needed for RTSP. It should be
silently ignored if sent. silently ignored if sent.
12.22 If-Match 12.22 If-Match
See [H14.25]. See [H14.25].
This field is especially useful for ensuring the integrity of the This field is especially useful for ensuring the integrity of the
presentation description, in both the case where it is fetched via presentation description, in both the case where it is fetched via
means external to RTSP (such as HTTP), or in the case where the server means external to RTSP (such as HTTP), or in the case where the server
implementation is guaranteeing the integrety of the description implementation is guaranteeing the integrity of the description
between the time of the DESCRIBE message and the SETUP message. between the time of the DESCRIBE message and the SETUP message.
The identifier is an opaque identifier, and thus is not specific to The identifier is an opaque identifier, and thus is not specific to
any particular session description language. any particular session description language.
12.23 If-Modified-Since 12.23 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional: if the requested variant and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a description will not be returned from the server (DESCRIBE) or a
stream will not be setup (SETUP); instead, a 304 (not modified) stream will not be set up (SETUP). Instead, a 304 (not modified)
response will be returned without any message-body. response will be returned without any message-body.
If-Modified-Since = "If-Modified-Since" ":" HTTP-date If-Modified-Since = "If-Modified-Since" ":" HTTP-date
An example of the field is: An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
12.24 Last-Modified 12.24 Last-Modified
skipping to change at line 2229 skipping to change at line 2344
See [H14.30]. See [H14.30].
12.26 Proxy-Authenticate 12.26 Proxy-Authenticate
See [H14.33]. See [H14.33].
12.27 Proxy-Require 12.27 Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive The Proxy-Require header is used to indicate proxy-sensitive
features that MUST be stripped by the proxy to the server if not features that MUST be supported by the proxy. Any Proxy-Require header
supported. Furthermore, any Proxy-Require header features that are not features that are not supported by the proxy MUST be negatively
supported by the proxy MUST be negatively acknowledged by the proxy to acknowledged by the proxy to the client if not supported. Servers
the client if not supported. should treat this field identically to the Require field.
See Section 12.32 for more details on the mechanics of this message See Section 12.32 for more details on the mechanics of this message
and a usage example. and a usage example.
We explored using the W3C's PEP proposal [22] for this We explored using the W3C's PEP proposal [22] for this
functionality. However, we determined that such a device was too functionality. However, Require, Proxy-Require, and Unsupported
complex for our needs. allow the addition of extensions with far less complexity.
This field roughly corresponds to the C-PEP field in the PEP draft. This field roughly corresponds to the C-PEP field in the PEP draft.
12.28 Public 12.28 Public
See [H14.35]. See [H14.35].
12.29 Range 12.29 Range
This request and response header field specifies a range of time. This request and response header field specifies a range of time.
The range can be specified in a number of units. This specification The range can be specified in a number of units. This specification
defines the smpte (see Section 3.5) and clock (see Section 3.7) range defines the smpte (see Section 3.5) and clock (see Section 3.7) range
units. Within RTSP, byte ranges [H14.36.1] are not meaningful and MUST units. Within RTSP, byte ranges [H14.36.1] are not meaningful and MUST
NOT be used. The header may also contain a time parameter in UTC, NOT be used. The header may also contain a time parameter in UTC,
specifying the time at which the operation is to be made effective. specifying the time at which the operation is to be made effective.
Servers supporting the Range header MUST understand the NPT range Servers supporting the Range header MUST understand the NPT range
format and SHOULD understand the SMPTE range format. The Range format and SHOULD understand the SMPTE range format. The Range
response header indicates what range of time is actually being played response header indicates what range of time is actually being played
or recorded. 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 byterange
allows to select a clip from the media object, to play from a given header. It allows clients to select an excerpt from the media
point to the end and from the current location to a given point. object, and to play from a given point to the end as well as from
The start of playback can be scheduled for at any time in the the current location to a given point. The start of playback can be
future, although a server may refuse to keep server resources for scheduled for any time in the future, although a server may refuse
extended idle periods. to keep server resources for extended idle periods.
12.30 Referer 12.30 Referer
See [H14.37]. The URL refers to that of the presentation See [H14.37]. The URL refers to that of the presentation
description, typically retrieved via HTTP. description, typically retrieved via HTTP.
12.31 Retry-After 12.31 Retry-After
See [H14.38]. See [H14.38].
12.32 Require 12.32 Require
The Require header is used by clients to query the server about The Require header is used by clients to query the server about
features that it may or may not support. The server MUST respond to options that it may or may not support. The server MUST respond to
this header by negatively acknowledging those features which are NOT this header by using the Unsupported header to negatively acknowledge
supported in the Unsupported header. those options which are NOT supported.
For example Require = "Require" ":" 1#option-tag
Example:
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0 C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302 CSeq: 302
Require: funky-feature Require: funky-feature
Funky-Parameter: funkystuff Funky-Parameter: funkystuff
S->C: RTSP/1.0 200 Option not supported S->C: RTSP/1.0 551 Option not supported
CSeq: 302 CSeq: 302
Unsupported: funky-feature Unsupported: funky-feature
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0 C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 303 CSeq: 303
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 303 CSeq: 303
This is to make sure that the client-server interaction will proceed This is to make sure that the client-server interaction will proceed
optimally when all options are understood by both sides, and only slow 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 down if options aren't understood (as in the case above). For a
well-matched client-server pair, the interaction proceeds quickly, well-matched client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation mechanisms. In saving a round-trip often required by negotiation mechanisms. In
addition, it also removes state ambiguity when the client requires addition, it also removes state ambiguity when the client requires
features that the server doesn't understand. features that the server doesn't understand.
We explored using the W3C's PEP proposal [22] for this We explored using the W3C's PEP proposal [22] for this
functionality. However, we determined that such a device was too functionality. However, Require, Proxy-Require, and Unsupported
complex for our needs. allow the addition of extensions with far less complexity.
This field roughly corresponds to the PEP field in the PEP draft. This field roughly corresponds to the PEP field in the PEP draft.
Proxies and other intermediary devices SHOULD ignore features that are Proxies and other intermediary devices SHOULD ignore features that are
not understood in this field. If a particular extension requires that not understood in this field. If a particular extension requires that
intermediate devices support it, the extension should be tagged in the intermediate devices support it, the extension should be tagged in the
Proxy-Require field instead (see Section 3.4). Proxy-Require field instead (see Section 3.4).
12.33 RTP-Info 12.33 RTP-Info
skipping to change at line 2346 skipping to change at line 2463
stream. This allows clients to gracefully deal with packets stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that packets that originated before the seek from packets that
originated after the seek. originated after the seek.
rtptime: rtptime:
Indicates the RTP timestamp of the first packet of the stream. Indicates the RTP timestamp of the first packet of the stream.
The client uses this value to calculate the mapping of RTP time The client uses this value to calculate the mapping of RTP time
to NPT. to NPT.
This information is also available in RTCP timestamps. However, in A mapping from RTP timestamps to NTP timestamps (wall clock) is
available via RTCP. However, this information is not sufficient to
generate a mapping from RTP timestamps to NPT. Furthermore, in
order to ensure that this information is available at the necessary order to ensure that this information is available at the necessary
time (immediately at startup or after a seek), and that it is time (immediately at startup or after a seek), and that it is
delivered reliably, it is placed in the RTSP control channel as delivered reliably, this mapping is placed in the RTSP control
well. channel.
RTP-Info = "RTP-Info" ":" In order to compensate for drift for long, uninterrupted
1#stream-url ";" presentations, RTSP clients should additionally map NPT to NTP,
*parameter using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping.
Syntax:
RTP-Info = "RTP-Info" ":" 1#stream-url ";" *parameter
stream-url = "url" "=" url stream-url = "url" "=" url
parameter = ";" "seq" "=" sequence-number parameter = ";" "seq" "=" sequence-number
sequence-number = 1*16(DIGIT) sequence-number = 1*(DIGIT)
Example: Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=43754027, RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=43754027,
url=rtsp://foo.com/bar.avi/streamid=1;seq=34834738 url=rtsp://foo.com/bar.avi/streamid=1;seq=34834738
12.34 Scale 12.34 Scale
A scale value of 1 indicates normal play or record at the normal A scale value of 1 indicates normal play or record at the normal
forward viewing rate. If not 1, the value corresponds to the rate with forward viewing rate. If not 1, the value corresponds to the rate 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
skipping to change at line 2427 skipping to change at line 2551
may be necessary. When data is delivered over UDP, it is highly may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss recommended that means such as RTCP be used to track packet loss
rates. rates.
12.36 Server 12.36 Server
See [H14.39] See [H14.39]
12.37 Session 12.37 Session
This request and response header field identifies an RTSP session, This request and response header field identifies an RTSP session
started by the media server in a SETUP response and concluded by started by the media server in a SETUP response and concluded by
TEARDOWN on the presentation URL. The session identifier is chosen by TEARDOWN on the presentation URL. The session identifier is chosen by
the media server (see Section 3.4). Once a client receives a Session the media server (see Section 3.4). Once a client receives a Session
identifier, it MUST return it for any request related to that session. identifier, it MUST return it for any request related to that session.
A server does not have to set up a session identifier if it has other 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. means of identifying a session, such as dynamically generated URLs.
Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ] Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
The timeout parameter is only allowed in a response header. The server The timeout parameter is only allowed in a response header. The server
uses it to indicate to the client how long the server is prepared to 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 wait between RTSP commands before closing the session due to lack of
activity (see Section A). The timeout is measured in seconds, with a activity (see Section A). The timeout is measured in seconds, with a
default of 60 seconds (1 minute). default of 60 seconds (1 minute).
skipping to change at line 2448 skipping to change at line 2571
The timeout parameter is only allowed in a response header. The server 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 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 wait between RTSP commands before closing the session due to lack of
activity (see Section A). The timeout is measured in seconds, with a activity (see Section A). The timeout is measured in seconds, with a
default of 60 seconds (1 minute). 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 constituting a presentation, as long as all the streams come
from the same server. (See example in Section 14). However, multiple from the same server. (See example in Section 14). However, multiple
``user'' sessions for the same URL from the same client MUST use ``user'' sessions for the same URL from the same client MUST use
different session identifiers. different session identifiers.
The session identifier is needed to distinguish several delivery The session identifier is needed to distinguish several delivery
requests for the same URL coming from the same client. requests for the same URL coming from the same client.
The response 454 (Session Not Found) is returned if the session The response 454 (Session Not Found) is returned if the session
identifier is invalid. identifier is invalid.
skipping to change at line 2470 skipping to change at line 2593
The timestamp general header describes when the client sent the The timestamp general header describes when the client sent the
request to the server. The value of the timestamp is of significance request to the server. The value of the timestamp is of significance
only to the client and may use any timescale. The server MUST echo the only to the client and may use any timescale. The server MUST echo the
exact same value and MAY, if it has accurate information about this, exact same value and MAY, if it has accurate information about this,
add a floating point number indicating the number of seconds that has add a floating point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp is used by 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 the client to compute the round-trip time to the server so that it can
adjust the timeout value for retransmissions. adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
[ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ] delay = *(DIGIT) [ "." *(DIGIT) ]
12.39 Transport 12.39 Transport
This request header indicates which transport protocol is to be used This request header indicates which transport protocol is to be used
and configures its parameters such as destination address, and configures its parameters such as destination address,
compression, multicast time-to-live and destination port for a single compression, multicast time-to-live and destination port for a single
stream. It sets those values not already determined by a presentation stream. It sets those values not already determined by a presentation
description. description.
Transports are comma separated, listed in order of preference. Transports are comma separated, listed in order of preference.
Parameters may be added to each tranpsort, separated by a semicolon. Parameters may be added to each transport, separated by a semicolon.
The Transport header MAY also be used to change certain transport The Transport header MAY also be used to change certain transport
parameters. A server MAY refuse to change parameters of an existing parameters. A server MAY refuse to change parameters of an existing
stream. stream.
The server MAY return a Transport response header in the response to The server MAY return a Transport response header in the response to
indicate the values actually chosen. indicate the values actually chosen.
A Transport request header field may contain a list of transport A Transport request header field may contain a list of transport
options acceptable to the client. In that case, the server MUST return options acceptable to the client. In that case, the server MUST return
skipping to change at line 2507 skipping to change at line 2629
transport/profile/lower-transport. transport/profile/lower-transport.
The default value for the ``lower-transport'' parameters is specific The default value for the ``lower-transport'' parameters is specific
to the profile. For RTP/AVP, the default is UDP. 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: unicast | multicast
Mutually exclusive indication of whether unicast or multicast : mutually exclusive indication of whether unicast or multicast
delivery will be attempted. Default value is multicast. Clients delivery will be attempted. Default value is multicast. Clients
that are capable of handling both unicast and multicast that are capable of handling both unicast and multicast
transmission MUST indicate such capability by including two transmission MUST indicate such capability by including two
full transport-specs with separate parameters for each. 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.
server SHOULD authenticate the client and SHOULD log such To avoid becoming the unwitting perpetrator of a
attempts before allowing the client to direct a media stream to remote-controlled denial-of-service attack, a server SHOULD
an address not chosen by the server to avoid becoming the authenticate the client and SHOULD log such attempts before
unwitting perpetrator of a remote-controlled denial-of-service allowing the client to direct a media stream to an address not
attack. This is particularly important if RTSP commands are chosen by the server. This is particularly important if RTSP
issued via UDP, but TCP cannot be relied upon as reliable means commands are issued via UDP, but implementations cannot rely on
of client identification by itself. A server SHOULD not allow a TCP as reliable means of client identification by itself. A
client to direct media streams to an address that differs from server SHOULD not allow a client to direct media streams to an
the address commands are coming from. address that differs from the address commands are coming from.
source: source:
Unicast only. If the source address for the stream is different Unicast only. If the source address for the stream is different
than can be derived from the RTSP endpoint address (the server than can be derived from the RTSP endpoint address (the server
in playback or the client in recording), the source MAY be in playback or the client in recording), the source MAY be
specified. specified.
This information may also be available through SDP, however, since This information may also be available through SDP. However, since
this is more a feature of transport than media initialization, the this is more a feature of transport than media initialization, the
authoritative source for this information should be in the SETUP authoritative source for this information should be in the SETUP
response. response.
layers: layers:
The number of multicast layers to be used for this media The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses starting stream. The layers are sent to consecutive addresses starting
at the destination address. 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 append to the existing
existing resource rather than overwriting it. If appending is resource rather than overwrite it. If appending is requested
requested and the server does not support this, it MUST refuse and the server does not support this, it MUST refuse the
the request rather than overwrite the resouce identified by the request rather than overwrite the resource identified by the
URI. The append parameter is ignored if the mode parameter does URI. The append parameter is ignored if the mode parameter 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, using the mechanism defined in Section 10.12. control stream, using the mechanism defined in Section 10.11.
The argument provides the the channel number to be used in the The argument provides the channel number to be used in the $
$ statement. 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:
the RTP/RTCP port pair for a multicast session. Specified as a This parameter provides the RTP/RTCP port pair for a multicast
range (e.g. port=3456-3457). session. It is specified as a range, e.g., port=3456-3457.
client_port: client_port:
the RTP/RTCP port pair on the server in the unicast model. This parameter provides the unicast RTP/RTCP port pair on the
Specified as a range (e.g. port=3456-3457). client where media data and control information is to be sent.
It is specified as a range, e.g., port=3456-3457.
server_port: server_port:
the RTP/RTCP port pair on the server in the unicast model. This parameter provides the unicast RTP/RTCP port pair on the
Specified as a range (e.g. port=3456-3457). server where media data and control information is to be sent.
It is specified as a range, e.g., port=3456-3457.
ssrc: ssrc:
Indicates the RTP SSRC [20, Sec. 3] value that should be The ssrc parameter indicates the RTP SSRC [20, Sec. 3] value
(request) or will be (response) used by the media server. This that should be (request) or will be (response) used by the
parameter is only valid for unicast transmission. It identifies media server. This parameter is only valid for unicast
the synchronization source to be associated with the media transmission. It identifies the synchronization source to be
stream. associated with the media stream.
Transport = "Transport" ":" Transport &=& "Transport" ":"
1\#transport-spec & & 1#transport-spec
transport-spec = transport-protocol/profile[/lower-transport] transport-spec = transport-protocol/profile[/lower-transport]
*parameter *parameter
transport-protocol = "RTP" transport-protocol = "RTP"
profile = "AVP" profile = "AVP"
lower-transport = "TCP" | "UDP" lower-transport = "TCP" | "UDP"
parameter = ( "unicast" | "multicast" ) parameter = ( "unicast" | "multicast" )
| ";" "destination" [ "=" address ] | ";" "destination" [ "=" address ]
| ";" "compressed" | ";" "compressed"
| ";" "interleaved" "=" channel | ";" "interleaved" "=" channel
| ";" "append" | ";" "append"
| ";" "ttl" "=" ttl | ";" "ttl" "=" ttl
| ";" "layers" "=" 1*DIGIT | ";" "layers" "=" 1*DIGIT
| ";" "port" "=" port [ "-" port ] | ";" "port" "=" port [ "-" port ]
| ";" "client_port" "=" port [ "-" port ] | ";" "client_port" "=" port [ "-" port ]
| ";" "server_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;multicast;compressed;ttl=127;mode="PLAY", Transport: RTP/AVP;multicast;compressed;ttl=127;mode="PLAY",
RTP/AVP;unicast;compressed;client_port=3456-3457;mode="PLAY" 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.
skipping to change at line 2631 skipping to change at line 2753
The Transport header is restricted to describing a single RTP The Transport header is restricted to describing a single RTP
stream. (RTSP can also control multiple streams as a single stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of of session description formats greatly simplifies designs of
firewalls. firewalls.
12.40 Unsupported 12.40 Unsupported
Negative acknowledgement of features not supported by the server. In Negative acknowledgement of features not supported by the server. In
the case where the feature was specified via the Proxy-Require: field 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 (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 and the server, the proxy MUST insert a message reply with an error
message 506 (Feature not supported). message ``551 Option Not Supported''.
We explored using the W3C's PEP proposal [22] for this We explored using the W3C's PEP proposal [22] for this
functionality. However, we determined that such a device was too functionality. However, Require, Proxy-Require, and Unsupported
complex for our needs. allow the addition of extensions with far less complexity.
This field roughly corresponds to the PEP-Info and C-PEP-Info in This field roughly corresponds to the PEP-Info and C-PEP-Info in
the PEP draft. the PEP draft.
See Section 12.32 for a usage example. See Section 12.32 for a usage example.
12.41 User-Agent 12.41 User-Agent
See [H14.42] See [H14.42]
skipping to change at line 2664 skipping to change at line 2786
See [H14.44]. See [H14.44].
12.44 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 cacheable, with the
exception of the stream description returned by DESCRIBE. (Since the exception of the presentation description returned by DESCRIBE or
responses for anything but DESCRIBE and GET_PARAMETER do not return included with ANNOUNCE. (Since the responses for anything but DESCRIBE
any data, caching is not really an issue for these requests.) However, and GET_PARAMETER do not return any data, caching is not really an
it is desirable for the continuous media data, typically delivered issue for these requests.) However, it is desirable for the continuous
out-of-band with respect to RTSP, to be cached. media data, typically delivered out-of-band with respect to RTSP, to
be cached, as well as the session description.
On receiving a SETUP or PLAY request, the proxy would ascertain as to On receiving a SETUP or PLAY request, a proxy ascertains whether it
whether it has an up-to-date copy of the continuous media content. If has an up-to-date copy of the continuous media content and its
not, it would modify the SETUP transport parameters as appropriate and description. It can determine whether the copy is up-to-date by
forward the request to the origin server. Subsequent control commands issuing a SETUP or DESCRIBE request, respectively, and comparing the
such as PLAY or PAUSE would pass the proxy unmodified. The proxy would Last-Modified header with that of the cached copy. If the copy is not
then pass the continuous media data to the client, while possibly up-to-date, it modifies the SETUP transport parameters as appropriate
making a local copy for later re-use. The exact behavior allowed to and forwards the request to the origin server. Subsequent control
the cache is given by the cache-response directives described in commands such as PLAY or PAUSE then pass the proxy unmodified. The
proxy delivers the continuous media data to the client, while possibly
making a local copy for later reuse. The exact behavior allowed to the
cache is given by the cache-response directives described in
Section 12.8. 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.
To the client, an RTSP proxy cache would appear like a regular media To the client, an RTSP proxy cache appears like a regular media
server, to the media origin server like a client. Just like an HTTP server, to the media origin server like a client. Just as an HTTP
cache has to store the content type, content language, etc. for the cache has to store the content type, content language, and so on for
objects it caches, a media cache has to store the presentation the objects it caches, a media cache has to store the presentation
description. Typically, a cache would eliminate all description. Typically, a cache eliminates all transport-references
transport-references (that is, multicast information) from the (that is, multicast information) from the presentation description,
presentation description, since these are independent of the data since these are independent of the data delivery from the cache to the
delivery from the cache to the client. Information on the encodings client. Information on the encodings remains the same. If the cache is
remains the same. If the cache is able to translate the cached media able to translate the cached media data, it would create a new
data, it would create a new presentation description with all the presentation description with all the encoding possibilities it can
encoding possibilities it can offer. offer.
14 Examples 14 Examples
The following examples reference stream description formats that are The following examples refer to stream description formats that are
not finalized, such as RTSL and SDP. Please do not use these examples not standards, such as RTSL. The following examples are not to be used
as a reference for those formats. as a reference for those formats.
14.1 Media on Demand (Unicast) 14.1 Media on Demand (Unicast)
Client C requests a movie from media servers A ( audio.example.com) Client C requests a movie from media servers A ( audio.example.com)
and V (video.example.com). The media description is stored on a web and V (video.example.com). The media description is stored on a web
server W . The media description contains descriptions of the server W . The media description contains descriptions of the
presentation and all its streams, including the codecs that are presentation and all its streams, including the codecs that are
available, dynamic RTP payload types, the protocol stack and content available, dynamic RTP payload types, the protocol stack, and content
information such as language or copyright restrictions. It may also information such as language or copyright restrictions. It may also
give an indication about the time line of the movie. give an indication about the time line of the movie.
In our example, the client is only interested in the last part of the In this example, the client is only interested in the last part of the
movie. The server requires authentication for this movie. movie.
C->W: GET /twister.sdp HTTP/1.1 C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com Host: www.example.com
Accept: application/sdp Accept: application/sdp
W->C: HTTP/1.0 200 OK W->C: HTTP/1.0 200 OK
Content-Type: application/sdp Content-Type: application/sdp
v=0 v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202 o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session s=RTSP Session
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=murl:rtsp://audio.example.com/twister/audio.en a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31 m=video 0 RTP/AVP 31
a=murl:rtsp://audio.example.com/twister/video a=control:rtsp://audio.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1 CSeq: 1
Transport: rtp/udp;port=3056 Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
A->C: RTSP/1.0 200 OK A->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Session: 1234 Session: 1234
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
server_port=5000-5001
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 1 CSeq: 1
Transport: rtp/udp;port=3058 Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
V->C: RTSP/1.0 200 OK V->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Session: 1235 Session: 1235
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
server_port=5002-5003
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 2 CSeq: 2
Session: 1235 Session: 1235
Range: smpte=0:10:00- Range: smpte=0:10:00-
V->C: RTSP/1.0 200 OK V->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Session: 1235
Range: smpte=0:10:00-0:20:00
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 2 CSeq: 2
Session: 1234 Session: 1234
Range: smpte=0:10:00- Range: smpte=0:10:00-
A->C: RTSP/1.0 200 OK A->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Session: 1234
Range: smpte=0:10:00-0:20:00
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 3 CSeq: 3
Session: 1234 Session: 1234
A->C: RTSP/1.0 200 OK A->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 3 CSeq: 4
Session: 1235 Session: 1235
V->C: RTSP/1.0 200 OK V->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 4
Even though the audio and video track are on two different servers, Even though the audio and video track are on two different servers,
and may start at slightly different times and may drift with respect and may start at slightly different times and may drift with respect
to each other, the client can synchronize the two using standard RTP to each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender methods, in particular the time scale contained in the RTCP sender
reports. reports.
14.2 Streaming of a Container file 14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in For purposes of this example, a container file is a storage entity in
which multiple continuous media types pertaining to the same end-user which multiple continuous media types pertaining to the same end-user
presentation are present. In effect, the container file represents a presentation are present. In effect, the container file represents a
RTSP presentation, with each of its components being RTSP streams. RTSP presentation, with each of its components being RTSP streams.
Container files are a widely used means to store such presentations. Container files are a widely used means to store such presentations.
While the components are essentially transported as independant While the components are transported as independent streams, it is
streams, it is desirable to maintain a common context for those desirable to maintain a common context for those streams at the server
streams at the server end. end.
This enables the server to keep a single storage handle open This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case of easily. It also allows treating all the streams equally in case of
any prioritization of streams by the server. any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by client in order to preserve the selective retrieval of the streams by the client in order to preserve
artistic effect of the combined media presentation. Similarly, in such the artistic effect of the combined media presentation. Similarly, in
a tightly bound presentation, it is desirable to be able to control such a tightly bound presentation, it is desirable to be able to
all the streams via a single control message using an aggregate URL. control all the streams via a single control message using an
aggregate URL.
The following is an example of using a single RTSP session to control 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 C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
CSeq: 1 CSeq: 1
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 164 Content-Length: 164
v=0 v=0
o=- 2890844256 2890842807 IN IP4 172.16.2.93 o=- 2890844256 2890842807 IN IP4 172.16.2.93
s=RTSP Session s=RTSP Session
i=An Example of RTSP Session Usage i=An Example of RTSP Session Usage
a=control:rtsp://foo/twister # aggregate URL a=control:rtsp://foo/twister
t=0 0 t=0 0
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=control:rtsp://foo/twister/audio a=control:rtsp://foo/twister/audio
m=video 0 RTP/AVP 26 m=video 0 RTP/AVP 26
a=control:rtsp://foo/twister/video a=control:rtsp://foo/twister/video
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0 C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
CSeq: 2 CSeq: 2
Transport: RTP/AVP;port=8000 Transport: RTP/AVP;unicast;client_port=8000-8001
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001;
server_port=9000-9001
Session: 1234 Session: 1234
C->M: SETUP rtsp://foo/twister/video RTSP/1.0 C->M: SETUP rtsp://foo/twister/video RTSP/1.0
CSeq: 3 CSeq: 3
Transport: RTP/AVP;port=8002 Transport: RTP/AVP;unicast;client_port=8002-8003
Session: 1234 Session: 1234
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003;
server_port=9004-9005
Session: 1234 Session: 1234
C->M: PLAY rtsp://foo/twister RTSP/1.0 C->M: PLAY rtsp://foo/twister RTSP/1.0
CSeq: 4 CSeq: 4
Range: npt=0- Range: npt=0-
Session: 1234 Session: 1234
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 4 CSeq: 4
Session: 1234 Session: 1234
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0 C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
CSeq: 5 CSeq: 5
Session: 1234 Session: 1234
M->C: RTSP/1.0 4xx Only aggregate operation allowed M->C: RTSP/1.0 460 Only aggregate operation allowed
CSeq: 5 CSeq: 5
C->M: PAUSE rtsp://foo/twister RTSP/1.0 C->M: PAUSE rtsp://foo/twister RTSP/1.0
CSeq: 6 CSeq: 6
Session: 1234 Session: 1234
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 6 CSeq: 6
Session: 1234 Session: 1234
C->M: SETUP rtsp://foo/twister RTSP/1.0 C->M: SETUP rtsp://foo/twister RTSP/1.0
CSeq: 7 CSeq: 7
Transport: RTP/AVP;port=10000 Transport: RTP/AVP;unicast;client_port=10000
M->C: RTSP/1.0 4xx Aggregate operation not allowed M->C: RTSP/1.0 459 Aggregate operation not allowed
CSeq: 7 CSeq: 7
In the first instance of failure, the client tries to pause one stream In the first instance of failure, the client tries to pause one stream
(in this case video) of the presentation which is disallowed for that (in this case video) of the presentation. This is disallowed for that
presentation by the server. In the second instance, the aggregate URL presentation by the server. In the second instance, the aggregate 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 Single Stream Container Files
Some RTSP servers may treat all files as though they are ``container
files'', yet other servers may not support such a concept. Because of
this, clients SHOULD use the rules set forth in the session
description for request URLs, rather than assuming that a consistant
URL may always be used throughout. Here's an example of how a
multi-stream server might expect a single-stream file to be served:
C->S DESCRIBE rtsp://foo.com/test.wav RTSP/1.0
Accept: application/x-rtsp-mh, application/sdp
CSeq: 2
S->C RTSP/1.0 200 OK
CSeq: 2
Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp
Content-length: 48
v=0
o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file
i=audio test
t=0 0
m=audio 0 RTP/AVP 0
a=control:streamid=0
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;mode=play
CSeq: 3
S->C RTSP/1.0 200 OK
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
server_port=6970-6971;mode=play
CSeq: 3
Session: 2034820394
C->S PLAY rtsp://foo.com/test.wav RTSP/1.0
CSeq: 4
Session: 2034820394
S->C RTSP/1.0 200 OK
CSeq: 4
Note the different URL in the SETUP command, and then the switch back
to the aggregate URL in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but is less
than intuitive in the special case where the number of streams is one.
In this special case, it is recommended that servers be forgiving of
implementations that send:
C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
CSeq: 4
In the worst case, servers should send back:
S->C RTSP/1.0 460 Only aggregate operation allowed
CSeq: 4
One would also hope that server implementations are also forgiving of
the following:
C->S SETUP rtsp://foo.com/test.wav RTSP/1.0
Transport: rtp/avp/udp;client_port=6970-6971;mode=play
CSeq: 3
Since there is only a single stream in this file, it's not ambiguous
what this means.
14.4 Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, we 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/x-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 C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 1 CSeq: 1
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 44
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
a=control:rtsp://live.example.com/concert/audio
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 C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 2 CSeq: 2
Transport: multicast=224.2.0.1; port=3456; ttl=16 Transport: RTP/AVP;multicast
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Transport: RTP/AVP;multicast;destination=224.2.0.1;port=3456;ttl=16
Session: 0456804596
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 3 CSeq: 3
Session: 0456804596
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
Session: 0456804596
The attempt to position the stream fails since this is a live 14.5 Playing media into an existing session
presentation.
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. C indicates to the media
presentation description, C indicates to the media server that the server that the network addresses and encryption keys are already
network addresses and encryption keys are already given by the given by the conference, so they should not be chosen by the server.
conference, so they should not be chosen by the server. The example 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 C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 1 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/sdp
Content-Length: 44
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
i=See above
t=0 0
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 C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 2 CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
port=7000-7001;ttl=127
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
14.5 Recording M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
port=7000-7001;ttl=127
Session: 91389234234
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
The conference participant C asks the media server M to record a C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
meeting. If the presentation description contains any alternatives, CSeq: 3
the server records them all. Session: 91389234234
C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 M->C: RTSP/1.0 200 OK
CSeq: 3
14.6 Recording
The conference participant client C asks the media server M to record
the audio portion of a meeting. The client uses the ANNOUNCE method to
provide meta-information about the recorded session to the server.
C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
CSeq: 90 CSeq: 90
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 121
v=0 v=0
s=Mbone Audio o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
i=Discussion of Mbone Engineering Issues s=IETF Meeting, Munich - 1
i=The thirty-ninth IETF meeting will be held in Munich, Germany
u=http://www.ietf.org/meetings/Munich.html
e=IETF Channel 1 <ietf39-mbone@uni-koeln.de>
p=IETF Channel 1 +49-172-2312 451
c=IN IP4 224.0.1.11/127
t=3080271600 3080703600
a=tool:sdr v2.4a6
a=type:test
m=audio 21010 RTP/AVP 5
c=IN IP4 224.0.1.11/127
a=ptime:40
m=video 61010 RTP/AVP 31
c=IN IP4 224.0.1.12/127
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 90 CSeq: 90
C->S: SETUP rtsp://server.example.com/meeting RTSP/1.0 C->S: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
CSeq: 91 CSeq: 91
Transport: RTP/AVP;mode=record Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=21010-21011;
mode=record;ttl=127
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 91 CSeq: 91
Transport: RTP/AVP;port=3244;mode=record
Session: 508876 Session: 508876
Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=21010-21011;
mode=record;ttl=127
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 C->S: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
CSeq: 92 CSeq: 92
Session: 508876 Session: 508876
Range: clock 19961110T1925-19961110T2015 Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=61010-61011;
mode=record;ttl=127
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 92 CSeq: 92
Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=61010-61011;
mode=record;ttl=127
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
CSeq: 93
Session: 508876
Range: clock 19961110T1925-19961110T2015
S->C: RTSP/1.0 200 OK
CSeq: 93
15 Syntax 15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF) The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 (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">
LOALPHA = <any US-ASCII lowercase letter "a".."z"> LOALPHA = <any US-ASCII lowercase letter "a".."z">
ALPHA = UPALPHA | LOALPHA ALPHA = UPALPHA | LOALPHA
DIGIT = <any US-ASCII digit "0".."9"> DIGIT = <any US-ASCII digit "0".."9">
CTL = <any US-ASCII control character CTL = <any US-ASCII control character
(octets 0 - 31) and DEL (127)> (octets 0 - 31) and DEL (127)>
CR = <US-ASCII CR, carriage return (13)> CR = <US-ASCII CR, carriage return (13)>
LF = <US-ASCII LF, linefeed (10)> LF = <US-ASCII LF, linefeed (10)>
SP = <US-ASCII SP, space (32)> SP = <US-ASCII SP, space (32)>
skipping to change at line 3022 skipping to change at line 3276
CRLF = CR LF CRLF = CR LF
LWS = [CRLF] 1*( SP | HT ) LWS = [CRLF] 1*( SP | HT )
TEXT = <any OCTET except CTLs> TEXT = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@" tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | "\" | <"> | "," | ";" | ":" | "\" | <">
| "/" | "[" | "]" | "?" | "=" | "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT | "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials> token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> ) quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">> qdtext = <any TEXT except <">>
quoted-pair = "\" CHAR quoted-pair = "CHAR
message-header = field-name ":" [ field-value ] CRLF message-header = field-name ":" [ field-value ] CRLF
field-name = token field-name = token
field-value = *( field-content | LWS ) field-value = *( field-content | LWS )
field-content = <the OCTETs making up the field-value and consisting field-content = <the OCTETs making up the field-value and
of either *TEXT or combinations of token, tspecials, consisting of either *TEXT or
and quoted-string> combinations of token, tspecials, and
quoted-string>
16 Security Considerations 16 Security Considerations
The protocol offers the opportunity for a remote-controlled The protocol offers the opportunity for a remote-controlled
denial-of-service attack. denial-of-service attack.
The attacker, using a forged source IP address, can ask for a stream The attacker may initiate traffic flows to one or more IP addresses by
to be played back to that forged IP address. Thus, an RTSP server specifying them as the destination in SETUP requests. While the
SHOULD only allow client-specified destinations for RTSP-initiated attacker's IP address may be known in this case, this is not always
traffic flows if the server has verified the client's identity, e.g., useful in prevention of more attacks or ascertaining the attackers
using the RTSP authentication mechanisms. identity. Thus, an RTSP server SHOULD only allow client-specified
destinations for RTSP-initiated traffic flows if the server has
verified the client's identity, either against a database of known
users using RTSP authentication mechanisms (preferrably digest
authentication or stronger), or other secure means.
Since there is no relation between a transport layer connection and an Since there is no relation between a transport layer connection and an
RTSP session, it is possible for a malicious client to issue requests RTSP session, it is possible for a malicious client to issue requests
with random session identifiers which would affect unsuspecting with random session identifiers which would affect unsuspecting
clients. This does not require spoofing of network packet addresses. clients. The server SHOULD use a large, random and non-sequential
The server SHOULD use a large random session identifier to make this session identifier to minimize the possibility of this kind of attack.
attack more difficult.
Both problems can be be prevented by appropriate authentication.
Servers SHOULD implement both basic and digest [8] authentication. Servers SHOULD implement both basic and digest [8] authentication. In
environments requring tighter security for the control messages,
transport layer mechanims such as [7] SHOULD be used.
In addition, the security considerations outlined in [H15] apply. In addition, the security considerations outlined in [H15] apply.
A RTSP Protocol State Machines A RTSP Protocol State Machines
The RTSP client and server state machines describe the behavior of The RTSP client and server state machines describe the behavior of
the protocol from RTSP session initialization through RTSP session the protocol from RTSP session initialization through RTSP session
termination. termination.
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 composed of multiple streams will have an effect on the individual
states of all the streams. For example, if the presentation /movie states of all the streams. For example, if the presentation /movie
contains two streams, /movie/audio and /movie/video, then the contains two streams, /movie/audio and /movie/video, then the
following command: following command:
PLAY rtsp://foo.com/movie RTSP/1.0 PLAY rtsp://foo.com/movie RTSP/1.0
CSeq: 559 CSeq: 559
Session: 12345 Session: 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.
skipping to change at line 3104 skipping to change at line 3361
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 end of the requested
The client also changes state from Playing/Recording to Ready when the 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 sent next state after response
Init SETUP Ready Init SETUP Ready
TEARDOWN Init TEARDOWN Init
Ready PLAY Playing Ready PLAY Playing
RECORD Recording RECORD Recording
TEARDOWN Init TEARDOWN Init
SETUP Ready SETUP Ready
Playing PAUSE Ready Playing PAUSE Ready
TEARDOWN Init TEARDOWN Init
PLAY Playing PLAY Playing
SETUP Playing (changed transport) SETUP Playing (changed transport)
skipping to change at line 3160 skipping to change at line 3416
In general,the server changes state on receiving requests. If the In general,the server changes state on receiving requests. If the
server is in state Playing or Recording and in unicast mode, it MAY server is in state Playing or Recording and in unicast mode, it MAY
revert to Init and tear down the RTSP session if it has not received revert to Init and tear down the RTSP session if it has not received
``wellness'' information, such as RTCP reports or RTSP commands, from ``wellness'' information, such as RTCP reports or RTSP commands, from
the client for a defined interval, with a default of one minute. The the client for a defined interval, with a default of one minute. The
server can declare another timeout value in the Session response server can declare another timeout value in the Session response
header (Section 12.37). If the server is in state Ready, it MAY revert header (Section 12.37). If the server is in state Ready, it MAY revert
to Init if it does not receive an RTSP request for an interval of more to Init if it does not receive an RTSP request for an interval of more
than one minute. Note that some requests (such as PAUSE) may be than one minute. Note that some requests (such as PAUSE) may be
effective at a future time or position, and server state transitions effective at a future time or position, and server state changes at
at the appropriate time. The server reverts from state Playing or the appropriate time. The server reverts from state Playing or
Recording to state Ready at the end of the range requested by the Recording to state Ready at the end of the range requested by the
client. client.
The REDIRECT message, when sent, is effective immediately unless it The REDIRECT message, when sent, is effective immediately unless it
has a Range header specifying when the redirect is effective. In such has a Range header specifying when the redirect is effective. In such
a case, server state will also change at the appropriate time. a case, server state will also change at the appropriate time.
If no explicit SETUP is required for the object, the state starts at If no explicit SETUP is required for the object, the state starts at
Ready and there are only two states, Ready and Playing. Ready and there are only two states, Ready and Playing.
The ``next state'' column indicates the state assumed after sending a The ``next state'' column indicates the state assumed after sending a
success response (2xx). If a request results in a status code of 3xx, success response (2xx). If a request results in a status code of 3xx,
the state becomes Init. A status code of 4xx results in no change. the state becomes Init. A status code of 4xx results in no change.
state message next state state message received next state
Init SETUP Ready Init SETUP Ready
TEARDOWN Init TEARDOWN Init
Ready PLAY Playing Ready PLAY Playing
SETUP Ready SETUP Ready
TEARDOWN Init TEARDOWN Init
RECORD Recording RECORD Recording
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 Interaction with RTP B Interaction with RTP
RTSP allows to play selected, non-contiguous sections of a RTSP allows media clients to control selected, non-contiguous
presentation. The media client playing back the RTP stream should not sections of media presentations, rendering those streams with an RTP
media layer[20]. The media layer rendering the RTP stream should not
be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
timestamps MUST be continuous and monotonic across jumps of NPT. timestamps MUST be continuous and monotonic across jumps of NPT.
As an example, assume a clock frequency of 8000 Hz, a packetization As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms and an initial sequence number and timestamp of interval of 100 ms and an initial sequence number and timestamp of
zero. First we play NPT 10 through 15, then skip ahead and play NPT 18 zero. First we play NPT 10 through 15, then skip ahead and play NPT 18
through 20. The first segment is presented as RTP packets with through 20. The first segment is presented as RTP packets with
sequence numbers 0 through 49 and timestamp 0 through 39,200. The sequence numbers 0 through 49 and timestamp 0 through 39,200. The
second segment consists of RTP packets with sequence number 50 through second segment consists of RTP packets with sequence number 50 through
69, with timestamps 40,000 through 55,200. 69, with timestamps 40,000 through 55,200.
We cannot assume that the RTSP client can communicate with the RTP We cannot assume that the RTSP client can communicate with the RTP
media agent, as the two may be independent processes. If the RTP media agent, as the two may be independent processes. If the RTP
timestamp shows the same gap as the NPT, the media agent will timestamp shows the same gap as the NPT, the media agent will
assume that there is a pause in the presentation. If the jump in assume that there is a pause in the presentation. If the jump in
NPT is large enough, the RTP timestamp may roll over and the media NPT is large enough, the RTP timestamp may roll over and the media
agent may believe later packets to be duplicates of packets just agent may believe later packets to be duplicates of packets just
played out. played out.
For certain datatypes, tight integration between the RTSP layer and
the RTP layer will be necessary. This by no means precludes the
above restriction. Combined RTSP/RTP media clients should use the
RTP-Info field to determine whether incoming RTP packets were sent
before or after a seek.
For continuous audio, the server SHOULD set the RTP marker bit at the
beginning of serving a new PLAY request. This allows the client to
perform playout delay adaptation.
For scaling (see Section 12.34), RTP timestamps should correspond to For scaling (see Section 12.34), RTP timestamps should correspond to
the playback timing. For example, when playing video recorded at 30 the playback timing. For example, when playing video recorded at 30
frames/second at a scale of two and speed (Section 12.35) of one, the frames/second at a scale of two and speed (Section 12.35) of one, the
server would drop every second frame to maintain and deliver video server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame. would increase by 1/15 second for each video frame.
The client can maintain a correct display of NPT by noting the RTP The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning. The timestamp value of the first packet arriving after repositioning. The
sequence parameter of the RTP-Info (Section 12.33 header provides the sequence parameter of the RTP-Info (Section 12.33) header provides the
last sequence number of the previous segment. last sequence number of the previous segment.
C Use of SDP for RTSP Session Descriptions C Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP [6]) may be used to describe The Session Description Protocol (SDP [6]) may be used to describe
streams or presentations in RTSP. Such usage is limited to specifying streams or presentations in RTSP. Such usage is limited to specifying
means of access and encoding(s) for: means of access and encoding(s) for:
* Scenario A: A presentation comprised of streams from one or more aggregate control:
servers that are not available for aggregate control. Such a A presentation composed of streams from one or more servers
that are not available for aggregate control. Such a
description is typically retrieved by HTTP or other non-RTSP description is typically retrieved by HTTP or other non-RTSP
means. However, they MAY be received with ANNOUNCE methods. means. However, they may be received with ANNOUNCE methods.
* 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, non-aggregate control:
embedded in a web page) that triggers a RTSP session, and the usage in A presentation composed of multiple streams from a single
replies to RTSP DESCRIBE requests. However, it does not address the server that are available for aggregate control. Such a
issue of media or encoding negotiation within such descriptions. description is typically returned in reply to a DESCRIBE
request on a URL, or received in an ANNOUNCE method.
C.1 Specification This appendix describes how an SDP file, retrieved, for example,
through HTTP, determines the operation of an RTSP session. It also
describes how a client should interpret SDP content returned in reply
to a DESCRIBE request. SDP provides no mechanism by which a client can
distinguish, without human guidance, between several media streams to
be rendered simultaneously and a set of alternatives (e.g., two audio
streams spoken in different languages).
C.1 Definitions
The terms ``session-level'', ``media-level'' and other key/attribute The terms ``session-level'', ``media-level'' and other key/attribute
names and values used in this appendix are as defined in [6]. SDP names and values used in this appendix are to be used as defined in
fields not specifically mentioned in this section are assumed to have [6].
their usual meaning.
C.1.1 Control URL C.1.1 Control URL
The ``a=control:'' field is used to convey the control URL. This The ``a=control:'' attribute is used to convey the control URL. This
field is used both at the media-level to provide a means to reference attribute is used both for the session and media descriptions. If used
individual streams, and at the session-level to signify a global URL for individual media, it indicates the URL to be used for controlling
for aggregate control, providing the URL to be used on aggregate that particular media stream. If found at the session level, the
commands (PLAY, PAUSE, etc.). attribute indicates the URL for aggregate control.
Example: Example:
a=control:rtsp://example.com/foo a=control:rtsp://example.com/foo
This field may contain both relative and absolute URLs, following the This attribute may contain either relative and absolute URLs,
rules and conventions set out in RFC 1808 ([16]). Specifically, the following the rules and conventions set out in RFC 1808 ([16]).
order for which implementations should look for a base URL is as Implementations should look for a base URL in the following order:
follows:
* The RTSP Content-Base field 1. The RTSP Content-Base field
* The RTSP Content-Location field 2. The RTSP Content-Location field
* The RTSP request URL 3. The RTSP request URL
If this field contains only an asterix (*), then the URL is treated as If this attribute contains only an asterisk (*), then the URL is
if it were an empty embedded URL, and thus inherits the entire base treated as if it were an empty embedded URL, and thus inherits the
URL. entire base URL.
C.1.2 Media streams C.1.2 Media streams
The ``m='' field is used to enumerate the streams. It is expected that The ``m='' field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate all the specified streams will be rendered with appropriate
synchronization. If the session is unicast, the port number simply synchronization. If the session is unicast, the port number serves as
serves as a recommendation, and would still need to be conveyed to the a recommendation from the server to the client; the client still has
server via a SETUP request. The port number may be specified as 0, in to include it in its SETUP request and may ignore this recommendation.
which case the client makes the choice of the port. If the server has no preference, it SHOULD set the port number value
to zero.
Example: Example:
m=audio 0 RTP/AVP 31 m=audio 0 RTP/AVP 31
C.1.3 Payload type(s) C.1.3 Payload type(s)
The payload type(s) are specified in the ``m='' field. In case the The payload type(s) are specified in the ``m='' field. In case the
payload type is a static payload type from RFC 1890([1]), no other payload type is a static payload type from RFC 1890([1]), no other
information is required. In case it is a dynamic payload type, the information is required. In case it is a dynamic payload type, the
media attribute ``rtpmap'' is used to specify what the media is. The media attribute ``rtpmap'' is used to specify what the media is. The
``encoding name'' within the ``rtpmap'' attribute may be one of those ``encoding name'' within the ``rtpmap'' attribute may be one of those
specified in RFC 1890(Sections 5 and 6), or an experimental encoding specified in RFC 1890(Sections 5 and 6), or an experimental encoding
with a ``X-'' prefix as specified in [6]. Codec-specific parameters with a ``X-'' prefix as specified in [6]. Codec-specific parameters
are not specified in this field but the ``fmtp'' attribute described are not specified in this field, but rather in the ``fmtp'' attribute
below. Implementors seeking to register new encodings should follow described below. Implementors seeking to register new encodings should
the procedure in RFC 1890. If the media type is not suited to the RTP follow the procedure in RFC 1890. If the media type is not suited to
AV profile, then it is recommended that a new profile be created and the RTP AV profile, then it is recommended that a new profile be
the appropriate profile name must be used in lieu of ``RTP/AVP'' in created and the appropriate profile name be used in lieu of
the ``m='' field. An informational document may be published in lieu ``RTP/AVP'' in the ``m='' field.
of this if the usage is expected to be limited or experimental.
C.1.4 Format specific parameters C.1.4 Format-specific parameters
This is accomplished using the ``fmtp'' media attribute. The syntax of Format-specific parameters are conveyed using the ``fmtp'' media
the ``fmtp'' attribute is specific to the encoding(s) that the attribute. The syntax of the ``fmtp'' attribute is specific to the
attribute refers to. This is with the exception of the number of encoding(s) that the attribute refers to. Note that the packetization
samples per packet, which is conveyed using the ``ptime'' attribute. interval is conveyed using the ``ptime'' attribute.
C.1.5 Length of presentation C.1.5 Length of presentation
This is applicable to non-live sessions(typically on-demand retreivals The ``a=length'' attribute defines the total length of stored
of stored files) only and is specified using a media-level sessions. (The length of live sessions can be deduced from the ``t''
``a=length'' field. It defines the total length of the presentation in and ``r'' parameters.) Unless the presentation contains media streams
time. The unit is specified first, followed by the value. The units of different durations, the length attribute is a session-level
and their values are as defined in Section 3. attribute. The unit is specified first, followed by the value. The
units and their values are as defined in Section 3.5, 3.6 and 3.7.
Example : Example :
a=length:npt=34.4368 a=length:npt=34.4368
C.1.6 Time of availability C.1.6 Time of availability
It is required that suitable values for the start and stop times for The ``t='' field MUST contain suitable values for the start and stop
the ``t='' field be used for both scnearios. In Scenario B, the server times for both aggregate and non-aggregate stream control. With
SHOULD indicate a stop time value for which it guarantees the aggregate control, the server SHOULD indicate a stop time value for
description to be valid, and a start time that is equal to or before which it guarantees the description to be valid, and a start time that
the time at which the DESCRIBE request was received.(It MAY also is equal to or before the time at which the DESCRIBE request was
indicate start and stop times of 0, meaning that the session is always received. It MAY also indicate start and stop times of 0, meaning that
available). In Scenario A, the values should reflect the actual period the session is always available. With non-aggregate control, the
for which the session is avaiable in keeping with SDP semantics, and values should reflect the actual period for which the session is
not depend on other means(such as the life of the web page containing available in keeping with SDP semantics, and not depend on other means
the description) for this purpose. (such as the life of the web page containing the description) for this
purpose.
C.1.7 Connection Information C.1.7 Connection Information
In some cases, the mandatory ``c='' field may have no well-defined In SDP, the ``c='' field contains the destination address for the
interpretation. This is since all the necessary information may be media stream. However, for on-demand unicast streams and some
conveyed by the control URL and subsequent RTSP operations. In such multicast streams, the destination address is specified by the client
cases, the address within this field must be set to a suitable null via the SETUP request. Unless the media content has a fixed
value. For address of type ``IP4'', this value is ``0.0.0.0''. destination address, the ``c='' field is to be set to a suitable null
value. For addresses of type ``IP4'', this value is ``0.0.0.0''.
C.1.8 Entity Tag C.1.8 Entity Tag
Because RTSP supports the If-Match field (see section 12.22) in a The optional ``a=etag'' attribute identifies a version of the session
session-description-independent fashion, it's necessary to embed an description. It is opaque to the client. SETUP requests may include
entirely opaque uniqueness field in the specification. The contents of this identifier in the If-Match field (see section 12.22) to only
this tag is totally implementation specific, so long as it serves as a allow session establishment if this attribute value still corresponds
unique identifier for this exact description of the media. Support of to that of the current description. The attribute value is opaque and
this tag is optional. may contain any character allowed within SDP attribute values.
Example : Example :
a=etag:''158bb3e7c7fd62ce67f12b533f06b83a'' a=etag:158bb3e7c7fd62ce67f12b533f06b83a
One could argue that the o= field provides identical functionality. One could argue that the ``o='' field provides identical
However, it does so in a manner that would put constraints on functionality. However, it does so in a manner that would put
servers that need to support multiple session description types constraints on servers that need to support multiple session
other than SDP for the same piece of media content. description types other than SDP for the same piece of media
content.
C.2 Scenario A C.2 Aggregate Control Not Available
Multiple media sections are specified, and each section MUST have the If a presentation does not support aggregate control and multiple
control URL specified via the ``a=control:''field. media sections are specified, each section MUST have the control URL
specified via the ``a=control:'' attribute.
Example: Example:
v=0 v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32 o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I came from a web page s=I came from a web page
t=0 0 t=0 0
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
m=video 8002 RTP/AVP 31 m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3 m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid a=control:rtsp://video.com/movie.vid
skipping to change at line 3379 skipping to change at line 3653
v=0 v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32 o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I came from a web page s=I came from a web page
t=0 0 t=0 0
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
m=video 8002 RTP/AVP 31 m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3 m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid a=control:rtsp://video.com/movie.vid
Note that the control URL in this case implies that the client Note that the position of the control URL in the description implies
establishes seperate RTSP control sessions to the servers audio.com that the client establishes separate RTSP control sessions to the
and video.com. servers audio.com and video.com.
C.3 Scenario B It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media client
through non-RTSP means. This is necessary as there is no mechanism to
indicate that the client should request more detailed media stream
information via DESCRIBE.
In this scenario, the server has multiple streams that are available C.3 Aggregate Control 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 In this scenario, the server has multiple streams that can be
``a=control:'' field may be omitted altogether. In case more than one controlled as a whole. In this case, there are both a media-level
stream is present, the ``a=control:'' field MUST be used. ``a=control:'' attributes, which are used to specify the stream URLs,
and a session-level ``a=control:'' attribute which is used as the
request URL for aggregate control. If the media-level URL is relative,
it is resolved to absolute URLs according to Section C.1.1 above.
Example: If the presentation comprises only a single stream, the media-level
``a=control:'' attribute may be omitted altogether. However, if the
presentation contains more than one stream, each media stream section
MUST contain its own ``a=control'' attribute.
Example:
v=0 v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32 o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I contain s=I contain
i=<more info> i=<more info>
t=0 0 t=0 0
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=control:rtsp://example.com/movie/ a=control:rtsp://example.com/movie/
m=video 8002 RTP/AVP 31 m=video 8002 RTP/AVP 31
a=control:trackID=1 a=control:trackID=1
m=audio 8004 RTP/AVP 3 m=audio 8004 RTP/AVP 3
skipping to change at line 3409 skipping to change at line 3689
o=- 2890844256 2890842807 IN IP4 204.34.34.32 o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I contain s=I contain
i=<more info> i=<more info>
t=0 0 t=0 0
c=IN IP4 0.0.0.0 c=IN IP4 0.0.0.0
a=control:rtsp://example.com/movie/ a=control:rtsp://example.com/movie/
m=video 8002 RTP/AVP 31 m=video 8002 RTP/AVP 31
a=control:trackID=1 a=control:trackID=1
m=audio 8004 RTP/AVP 3 m=audio 8004 RTP/AVP 3
a=control:trackID=2 a=control:trackID=2
In this example, the client is required to establish a single RTSP In this example, the client is required to establish a single RTSP
session to the server, and uses the URLs session to the server, and uses the URLs
rtsp://example.com/movie/trackID=1 and rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to setup the media streams, and rtsp://example.com/movie/trackID=2 to set up the video and audio
rtsp://example.com/movie/ to control it. streams, respectively. The URL rtsp://example.com/movie/ controls the
whole movie.
D Minimal RTSP implementation D Minimal RTSP implementation
D.1 Client D.1 Client
A client implementation MUST be able to do the following : A client implementation MUST be able to do the following :
* Generate the following requests : SETUP, TEARDOWN, and one of * Generate the following requests :
PLAY(ie. a minimal playback client) or RECORD(ie. a minimal SETUP, TEARDOWN, and one of PLAY (i.e., a minimal playback client)
recording client). If RECORD is implemented, ANNOUNCE must be or RECORD (i.e., a minimal recording client). If RECORD is
implemented as well. implemented, ANNOUNCE must be implemented as well.
* Include the following headers in requests: Connection, Session, * Include the following headers in requests:
Transport. If ANNOUNCE is implemented, the capability to include CSeq, Connection, Session, Transport. If ANNOUNCE is implemented,
headers Content-Language, Content-Encoding, Content-Length, the capability to include headers Content-Language,
Content-Type should be as well. Content-Encoding, Content-Length, and Content-Type should be as
* Parse and understand the following headers in responses: well.
* Parse and understand the following headers in responses: CSeq,
Connection, Session, Transport, Content-Language, Connection, Session, Transport, Content-Language,
Content-Encoding, Content-Length, Content-Type. If RECORD is Content-Encoding, Content-Length, Content-Type. If RECORD is
implemented, the Location header must be understood as well. implemented, the Location header must be understood as well.
RTP-complient implementations should also implement RTP-Info. RTP-compliant implementations should also implement RTP-Info.
* Understand the class of each error code received and notify the * Understand the class of each error code received and notify the
end-user, if one is present, of error codes in classes 4xx and end-user, if one is present, of error codes in classes 4xx and
5xx. The notification requirement may be relaxed if the end-user 5xx. The notification requirement may be relaxed if the end-user
explicity does not want it for one or all status codes. explicitly does not want it for one or all status codes.
* Expect and respond to asynchronous requests from the server, such * Expect and respond to asynchronous requests from the server, such
as ANNOUNCE. This does not necessarily mean that it should as ANNOUNCE. This does not necessarily mean that it should
implement the ANNOUNCE method, merely that it MUST respond implement the ANNOUNCE method, merely that it MUST respond
positively or negatively to any request received from the server positively or negatively to any request received from the server.
* Implement RTP transport.
Inclusion of the User-Agent header is recommended. Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial
implementations and/or to be a ``good citizen''.
The following capability sets are defined over and above the minimal * Implement RTP/AVP/UDP as a valid transport.
implementation : * Inclusion of the User-Agent header.
* Understand SDP session descriptions as defined in Appendix C
* Accept media initialization formats (such as SDP) from standard
input, command line, or other means appropriate to the operating
environment to act as a ``helper application'' for other
applications (such as web browsers).
There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict
requirements.
D.1.1 Basic Playback D.1.1 Basic Playback
The client MUST additionally be able to do the following: To support on-demand playback of media streams, the client MUST
* Include and parse the Range header, with ``npt'' units. additionally be able to do the following:
* Generate the PAUSE reqeust. * Include and parse the Range header, with NPT units.
* Generate the PAUSE request.
* Implement the REDIRECT method, and the Location header. * 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 D.1.2 Authentication-enabled
The client MUST additionally be able to do the following: In order to access media presentations from RTSP servers that require
authentication, the client MUST additionally be able to do the
following:
* Recognize the 401 status code. * Recognize the 401 status code.
* Parse and include the WWW-Authenicate header * Parse and include the WWW-Authenticate header
* Implement Basic and Digest authentication * Implement Basic Authentication and Digest Authentication
D.2 Server D.2 Server
A minimal server implementation MUST be able to do the following: A minimal server implementation MUST be able to do the following:
* Implement SETUP, TEARDOWN, OPTIONS and one of the PLAY(ie. a * Implement the following methods: SETUP, TEARDOWN, OPTIONS and
minimal playback server) or RECORD(ie. a minimal recording server) either PLAY (for a minimal playback server) or RECORD (for a
methods. If RECORD is implemented, ANNOUNCE should be implemented minimal recording server).
as well. If RECORD is implemented, ANNOUNCE should be implemented as well.
* Include the following headers in responses: Connection, * Include the following headers in responses: Connection,
Content-Length, Content-Type, Content-Language, Content-Encoding, Content-Length, Content-Type, Content-Language, Content-Encoding,
Transport, Public. The capability to include the Location header Transport, Public. The capability to include the Location header
should be implemented if the RECORD method is. RTP-complient should be implemented if the RECORD method is. RTP-compliant
implementations should also implement the RTP-Info field. implementations should also implement the RTP-Info field.
* Parse and respond appropriately to the following headers in * Parse and respond appropriately to the following headers in
requests: Connection, Session, Transport, Require. requests: Connection, Session, Transport, Require.
* Implement RTP transport.
Inclusion of the Server header is recommended. Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial
implementations and/or to be a ``good citizen''.
The following capability sets are defined over and above the minimal * Implement RTP/AVP/UDP as a valid transport.
implementation : * Inclusion of the Server header.
* Implement the DESCRIBE method.
* Generate SDP session descriptions as defined in Appendix C
There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict
requirements.
D.2.1 Basic Playback D.2.1 Basic Playback
The server MUST additionally be able to do the following: To support on-demand playback of media streams, the server MUST
* Include and parse the Range header, with ``npt'' units. additionally be able to do the following:
Implementation of ``smpte'' units is recommended.
* Include and parse the Range header, with NPT units. Implementation
of SMPTE units is recommended.
* Implement the PAUSE method. * Implement the PAUSE method.
* Implement the REDIRECT method, and the Location header.
Implementation of DESCRIBE and generation of SDP descriptions as In addition, in order to support commonly-accepted user interface
defined in Appendix C is highly recommended for this case. features, the following are highly recommended for on-demand media
servers:
* Include the length of the media presentation in the media
initialization information.
* Include mappings from data-specific timestamps to NPT. When RTP is
used, the rtptime portion of the RTP-Info field may be used to map
RTP timestamps to NPT.
Client implementations may use the presence of length information
to determine if the clip is seekable, and visably disable seeking
features for clips for which the length information is unavailable.
A common use of the presentation length is to implement a ``slider
bar'' which serves as both a progress indicator and a timeline
positioning tool.
Mappings from RTP timestamps to NPT are necessary to ensure correct
positioning of the slider bar.
D.2.2 Authentication-enabled D.2.2 Authentication-enabled
The server MUST additionally be able to do the following: In order to correctly handle client authentication, the server MUST
additionally be able to do the following:
* Generate the 401 status code when authentication is required for * Generate the 401 status code when authentication is required for
the resource. the resource.
* Parse and include the WWW-Authenicate header * Parse and include the WWW-Authenticate header
* Implement Basic and Digest authentication * Implement Basic Authentication and Digest Authentication
E Open Issues E Changes
1. Define text/rtsp-parameter MIME type. Since draft04 (September 17, 1997 version) of RTSP, the following
2. Allow byte offsets for Range (Prasoon Tiwari). changes were made:
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.
5. Server issues TEARDOWN and other 'event' notifications to
client? This raises the problem discussed in the previous open
issue, but is useful for the client if the data stream contains
no end indication.
F Changes * Further explanation of container files and how to deal with
``single-stream container files''.
* IANA procedure for registering option tags.
* New response codes (``461 Unsupported Transport'', ``462
Destination Unreachable'', ``551 Option Not Supported'').
* Practical minimum implementations established in Appendix D.
* Removed quasi-specification of ``text/rtsp-parameters'' with the
intent to define this separately.
* Closed out open issues
* Inserted ommisions in ``Since draft03...'' below (``etag''
change).
Since draft03 (July 30, 1997 version) of RTSP, the following changes Since draft03 (July 30, 1997 version) of RTSP, the following changes
were made: were made:
* PEP was removed, ``Require'' header returns * PEP was removed, ``Require'' header returns.
* Usage of SDP within RTSP is specified as an appendix * Usage of SDP within RTSP is specified as an appendix.
* Minimal RTSP implementation specified as an appendix * Minimal RTSP implementation specified as an appendix.
* The RTSP control sequence number was moved off of the request and * The RTSP control sequence number was moved off of the request and
response lines, and put into a new CSeq: header. response lines, and put into a new CSeq: header.
* Interaction with RTP appendix added * Interaction with RTP appendix added.
* Several changes to Transport: and RTP-Info: fields (RTP-Info: was * Several changes to Transport: and RTP-Info: fields ( RTP-Info was
formerly Transport-Info:) formerly Transport-Info:).
* Addition of ``etag'' mechanism in SDP, and corresponding If-Match:
field.
Between draft02 (March, 1997) and draft03 (July, 1997), the following Between draft02 (March, 1997) and draft03 (July, 1997), the following
changes were made: changes were made:
* Definition of RTP behavior. * Definition of RTP behavior.
* Definition of behavior for container files. * Definition of behavior for container files.
* Remove server-to-client DESCRIBE request. * 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,
and the start sequence number that will be used in the RTP and the start sequence number that will be used in the RTP
packets. packets.
* The use of the Session: header has been enhanced. Requests for * The use of the Session: header has been enhanced. Requests for
multiple URLs may be sent in a single session. multiple URLs may be sent in a single session.
* There is a distinction between aggregate(presentation) URLs and * There is a distinction between aggregate(presentation) URLs and
stream URLs. Error codes have been added to reflect the fact that stream URLs. Error codes have been added to reflect the fact that
some methods may be allowed only on a particular type of URL. some methods may be allowed only on a particular type of URL.
* 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.
G Author Addresses F 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 RealNetworks
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
H Acknowledgements G 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, Bruce Butterfield, Steve Casner, Francisco Cortes, Rahul Agarwal, Bruce Butterfield, Steve Casner, Francisco Cortes,
Martin Dunsmuir, Eric Fleischman, V. Guruprasad, Peter Haight, Mark Martin Dunsmuir, Eric Fleischman, V. Guruprasad, Peter Haight, Mark
Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka, Anders Klemets, Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka, Anders Klemets,
Ruth Lang, Stephanie Leif, Eduardo F. Llach, Rob McCool, David Oran, Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Rob
Sujal Patel, Alagu Periyannan, Igor Plotnikov, Pinaki Shah, Jeff McCool, David Oran, Sujal Patel, Alagu Periyannan, Igor Plotnikov,
Smith, Alexander Sokolsky, Dale Stammen, and John Francis Stracke. Pinaki Shah, Jeff Smith, Alexander Sokolsky, Dale Stammen, and John
Francis Stracke.
References References
1 H. Schulzrinne, ``RTP profile for audio and video conferences 1 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.
skipping to change at line 3699 skipping to change at line 4027
1889, Internet Engineering Task Force, Jan. 1996. 1889, Internet Engineering Task Force, Jan. 1996.
21 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.
22 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.
Full Copyright Statement
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