draft-ietf-mmusic-rtsp-05.txt   draft-ietf-mmusic-rtsp-06.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-05.txt Columbia U./Netscape/RealNetworks draft-ietf-mmusic-rtsp-06.txt Columbia U./Netscape/RealNetworks
October 28, 1997 Expires: April 28, 1998 November 21, 1997 Expires: May 21, 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 71 skipping to change at line 71
* 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 + 3.8 Option Tags
o 3.8.1 Registering New Option Tags With IANA 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 195 skipping to change at line 195
+ 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 Definitions + 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
H. Schulzrinne, A. Rao, R. Lanphier Page 4 H. Schulzrinne, A. Rao, R. Lanphier Page 4
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 Range 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 Aggregate Control Not Available + C.2 Aggregate Control Not Available
+ C.3 Aggregate Control Available + C.3 Aggregate Control Available
* 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
skipping to change at line 221 skipping to change at line 221
* 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 10.11). media stream with the control stream is possible (see Section 10.12).
In other words, RTSP acts as a ``network remote control'' for In other words, RTSP acts as a ``network remote control'' for
multimedia servers. multimedia servers.
The set of streams to be controlled is defined by a presentation The set of streams to be controlled is defined by a presentation
description. This memorandum does not define a format for a description. This memorandum does not define a format for a
presentation description. presentation description.
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
skipping to change at line 255 skipping to change at line 255
aspects from HTTP: aspects from HTTP:
* RTSP introduces a number of new methods and has a different * RTSP introduces a number of new methods and has a different
protocol identifier. protocol identifier.
* An RTSP server needs to maintain state by default in almost all * An RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP. cases, as opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests. * Both an RTSP server and client can issue requests.
* Data is carried out-of-band by a different protocol. (There is an * Data is carried out-of-band by a different protocol. (There is an
exception to this.) exception to this.)
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts [3]. consistent with current HTML internationalization efforts [2].
* 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.
The protocol supports the following operations: The protocol supports the following operations:
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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.
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 [3].
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 [4]. Terms not
listed here are defined as in HTTP/1.1. listed here are defined as in HTTP/1.1.
Aggregate control: Aggregate control:
The control of the multiple streams using a single timeline by The control of the multiple streams using a single timeline by
the server. For audio/video feeds, this means that the client the server. For audio/video feeds, this means that the client
may issue a single play or pause message to control both the may issue a single play or pause message to control both the
audio and video feeds. audio and video feeds.
Conference: Conference:
a multiparty, multimedia presentation, where ``multi'' implies a multiparty, multimedia presentation, where ``multi'' implies
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H. Schulzrinne, A. Rao, R. Lanphier Page 7 H. Schulzrinne, A. Rao, R. Lanphier Page 7
Continuous media: Continuous media:
Data where there is a timing relationship between source and Data where there is a timing relationship between source and
sink; that is, the sink must reproduce the timing relationship sink; that is, the sink must reproduce the timing relationship
that existed at the source. The most common examples of that existed at the source. The most common examples of
continuous media are audio and motion video. Continuous media continuous media are audio and motion video. Continuous media
can be real-time (interactive), where there is a ``tight'' can be real-time (interactive), where there is a ``tight''
timing relationship between source and sink, or streaming timing relationship between source and sink, or streaming
(playback), where the relationship is less strict. (playback), where the relationship is less strict.
Entity:
The information transferred as the payload of a request or
response. An entity consists of metainformation in the form of
entity-header fields and content in the form of an entity-body,
as described in Section 8.
Media initialization: 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.
Media parameter: Media parameter:
Parameter specific to a media type that may be changed before Parameter specific to a media type that may be changed before
or during stream playback. or during stream playback.
Media server: Media server:
The network entity providing playback or recording services for The server providing playback or recording services for one or
one or more media streams. Different media streams within a more media streams. Different media streams within a
presentation may originate from different media servers. A presentation may originate from different media servers. A
media server may reside on the same or a different host as the media server may reside on the same or a different host as the
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.
(Media) stream: (Media) stream:
A single media instance, e.g., an audio stream or a video A single media instance, e.g., an audio stream or a video
stream as well as a single whiteboard or shared application stream as well as a single whiteboard or shared application
group. When using RTP, a stream consists of all RTP and RTCP group. When using RTP, a stream consists of all RTP and RTCP
packets created by a source within an RTP session. This is packets created by a source within an RTP session. This is
equivalent to the definition of a DSM-CC stream([19]). equivalent to the definition of a DSM-CC stream([5]).
H. Schulzrinne, A. Rao, R. Lanphier Page 8
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:
Member of a conference. A participant may be a machine, e.g., a Member of a conference. A participant may be a machine, e.g., a
media record or playback server. media record or playback server.
H. Schulzrinne, A. Rao, R. Lanphier Page 8
Presentation: Presentation:
A set of one or more streams presented to the client as a A set of one or more streams presented to the client as a
complete media feed, using a presentation description as complete media feed, using a presentation description as
defined below. In most cases in the RTSP context, this implies defined below. In most cases in the RTSP context, this implies
aggregate control of those streams, but doesn't have to. aggregate control of those streams, but does not have to.
Presentation description: Presentation description:
A presentation description contains information about one or A presentation description contains information about one or
more media streams within a presentation, such as the set of more media streams within a presentation, such as the set of
encodings, network addresses and information about the content. encodings, network addresses and information about the content.
Other IETF protocols such as SDP [6] use the term ``session'' Other IETF protocols such as SDP (RFC XXXX) use the term
for a live presentation. The presentation description may take ``session'' for a live presentation. The presentation
several different formats, including but not limited to the description may take several different formats, including but
session description format SDP. not limited to the session description format SDP.
Response: Response:
An RTSP response. If an HTTP response is meant, that is An RTSP response. If an HTTP response is meant, that is
indicated explicitly. indicated explicitly.
Request: Request:
An RTSP request. If an HTTP request is meant, that is indicated An RTSP request. If an HTTP request is meant, that is 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 (e.g., port numbers, The negotiation of transport information (e.g., port numbers,
transport protocols) between the client and the server. transport protocols) between the client and the server.
H. Schulzrinne, A. Rao, R. Lanphier Page 9
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, RFC XXXX) or within the protocol itself. All HTTP
authentication mechanisms such as basic [5, Section 11.1] and authentication mechanisms such as basic [4, Section 11.1] and
digest authentication [8] are directly applicable. digest authentication [6] 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) [7],
a reliable datagram protocol (RDP, not widely used [10]) or a a reliable datagram protocol (RDP, not widely used [8]) or a
reliable stream protocol such as TCP [11] as it implements reliable stream protocol such as TCP [9] 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.
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 [10] or H.323
may be used to invite a server to a conference. may be used to invite a server to a conference.
Suitable for professional applications: Suitable for professional applications:
RTSP supports frame-level accuracy through SMPTE time stamps to RTSP supports frame-level accuracy through SMPTE time stamps to
allow remote digital editing. allow remote digital editing.
Presentation description neutral: Presentation description neutral:
The protocol does not impose a particular presentation The protocol does not impose a particular presentation
description or metafile format and can convey the type of description or metafile format and can convey the type of
format to be used. However, the presentation description must format to be used. However, the presentation description must
contain at least one RTSP URI. contain at least one RTSP URI.
Proxy and firewall friendly: Proxy and firewall friendly:
The protocol should be readily handled by both application and The protocol should be readily handled by both application and
transport-layer (SOCKS [13]) firewalls. A firewall may need to transport-layer (SOCKS [11]) 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 reuses HTTP concepts, so that the existing Where sensible, RTSP reuses HTTP concepts, so that the existing
infrastructure can be reused. This infrastructure includes PICS infrastructure can be reused. This infrastructure includes PICS
(Platform for Internet Content Selection [21]) for associating (Platform for Internet Content Selection [12,13]) for
labels with content. However, RTSP does not just add methods to associating labels with content. However, RTSP does not just
HTTP since the controlling continuous media requires server add methods to HTTP since the controlling continuous media
state in most cases. requires server state in most cases.
Appropriate server control: Appropriate server control:
If a client can start a stream, it must be able to stop a If a client can start a stream, it must be able to stop a
stream. Servers should not start streaming to clients in such a stream. Servers should not start streaming to clients in such a
way that clients cannot stop the stream. way that clients cannot stop the stream.
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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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, which reduces roundtrips in can be retrieved using HTTP or RTSP, which reduces roundtrips in
web-browser-based scenarios, yet also allows for standalone RTSP web-browser-based scenarios, yet also allows for standalone RTSP
servers and clients which don't rely on HTTP at all. servers and clients which do not rely on HTTP at all.
However, RTSP differs fundamentally from HTTP in that data delivery However, RTSP differs fundamentally from HTTP in that data delivery
takes place out-of-band in a different protocol. HTTP is an asymmetric takes place out-of-band in a different protocol. HTTP is an asymmetric
protocol where the client issues requests and the server responds. In protocol where the client issues requests and the server responds. In
RTSP, both the media client and media server can issue requests. RTSP RTSP, both the media client and media server can issue requests. RTSP
requests are also not stateless; they may set parameters and continue requests are also not stateless; they may set parameters and continue
to control a media stream long after the request has been to control a media stream long after the 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,
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2 Notational Conventions 2 Notational Conventions
Since many of the definitions and syntax are identical to HTTP/1.1, Since many of the definitions and syntax are identical to HTTP/1.1,
this specification only points to the section where they are defined this specification only points to the section where they are defined
rather than copying it. For brevity, [HX.Y] is to be taken to refer to rather than copying it. For brevity, [HX.Y] is to be taken to refer to
Section X.Y of the current HTTP/1.1 specification (RFC 2068). Section X.Y of the current HTTP/1.1 specification (RFC 2068).
All the mechanisms specified in this document are described in both All the mechanisms specified in this document are described in both
prose and an augmented Backus-Naur form (BNF) similar to that used in prose and an augmented Backus-Naur form (BNF) similar to that used in
RFC 2068 [H2.1]. It is described in detail in [14]. RFC 2068 [H2.1]. It is described in detail in [14], with the
difference that this RTSP specification maintains the ``1#'' notation
for comma-separated lists.
In this draft, we use indented and smaller-type paragraphs to provide In this draft, we use indented and smaller-type paragraphs to provide
background and motivation. Some of these paragraphs are marked with background and motivation. This is intended to give readers who were
HS, AR and RL, designating opinions and comments by the individual not involved with the formulation of the specification an
authors which may not be shared by the co-authors and require understanding of why things are the way that they are in RTSP.
resolution.
3 Protocol Parameters 3 Protocol Parameters
3.1 RTSP Version 3.1 RTSP Version
[H3.1] applies, with HTTP replaced by RTSP. [H3.1] applies, with HTTP replaced by RTSP.
3.2 RTSP URL 3.2 RTSP URL
The ``rtsp'', ``rtspu'' and ``rtsps'' schemes are used to refer to The ``rtsp'', ``rtspu'' and ``rtsps'' schemes are used to refer to
skipping to change at line 693 skipping to change at line 701
abs_path is defined in [H3.2.1]. abs_path is defined in [H3.2.1].
Note that fragment and query identifiers do not have a well-defined Note that fragment and query identifiers do not have a well-defined
meaning at this time, with the interpretation left to the RTSP meaning at this time, with the interpretation left to the RTSP
server. server.
The scheme rtsp requires that commands are issued via a reliable The scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtspu identifies protocol (within the Internet, TCP), while the scheme rtspu identifies
an unreliable protocol (within the Internet, UDP). The scheme rtsps an unreliable protocol (within the Internet, UDP). The scheme rtsps
indicates that a TCP connection secured by TLS [7] must be used. indicates that a TCP connection secured by TLS (RFC XXXX) must be
used.
If the port is empty or not given, port 554 is assumed. The semantics If the port is empty or not given, port 554 is assumed. The semantics
are that the identified resource can be controlled by RTSP at the are that the identified resource can be controlled by RTSP at the
server listening for TCP (scheme ``rtsp'') connections or UDP (scheme server listening for TCP (scheme ``rtsp'') connections or UDP (scheme
``rtspu'') packets on that port of host, and the Request-URI for the ``rtspu'') packets on that port of host, and the Request-URI for the
resource is rtsp_URL. resource is rtsp_URL.
The use of IP addresses in URLs SHOULD be avoided whenever possible The use of IP addresses in URLs SHOULD be avoided whenever possible
(see RFC 1924 [15]). (see RFC 1924 [15]).
A presentation or a stream is identified by a textual media A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions [H3.2] of identifier, using the character set and escape conventions [H3.2] of
URLs [17]. URLs may refer to a stream or an aggregate of streams, URLs [16]. URLs may refer to a stream or an aggregate of streams,
i.e., a presentation. Accordingly, requests described in Section 10 i.e., a presentation. Accordingly, requests described in Section 10
can apply to either the whole presentation or an individual stream can apply to either the whole presentation or an individual stream
within the presentation. Note that some request methods can only be within the presentation. Note that some request methods can only be
applied to 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
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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*xchar
Conference identifiers are used to allow RTSP sessions to obtain Conference identifiers are used to allow RTSP sessions to obtain
parameters from multimedia conferences the media server is parameters from multimedia conferences the media server is
participating in. These conferences are created by protocols participating in. These conferences are created by protocols
outside the scope of this specification, e.g., H.323 [18] or SIP outside the scope of this specification, e.g., H.323 [17] or SIP
[12]. Instead of the RTSP client explicitly providing transport [10]. 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.
participant inviting the media server would only supply a
conference identifier which is unique for that inviting party, the
media server could add an internal identifier for that party, e.g.,
its Internet address. However, this would prevent the conference
participant and the initiator of the RTSP commands from being two
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*( ALPHA | DIGIT | safe )
3.5 SMPTE Relative Timestamps 3.5 SMPTE Relative Timestamps
A SMPTE relative timestamp expresses time relative to the start of A SMPTE relative timestamp expresses time relative to the start of
the clip. Relative timestamps are expressed as SMPTE time codes for the clip. Relative timestamps are expressed as SMPTE time codes for
frame-level access accuracy. The time code has the format frame-level access accuracy. The time code has the format
hours:minutes:seconds:frames.subframes, with the origin at the start hours:minutes:seconds:frames.subframes, with the origin at the start
of the clip. RTSP uses the ``SMPTE 30 drop'' format. The frame rate is of the clip. The default smpte format is``SMPTE 30 drop'' format, with
29.97 frames per second. The ``frames'' field in the time value can frame rate is 29.97 frames per second. Other SMPTE codes MAY be
assume the values 0 through 29. The difference between 30 and 29.97 supported (such as "SMPTE 25") through the use of alternative use of
frames per second is handled by dropping the first two frame indices "smpte time". For the ``frames'' field in the time value can assume
(values 00 and 01) of every minute, except every tenth minute. If the the values 0 through 29. The difference between 30 and 29.97 frames
frame value is zero, it may be omitted. Subframes are measured in per second is handled by dropping the first two frame indices (values
00 and 01) of every minute, except every tenth minute. If the frame
value is zero, it may be omitted. Subframes are measured in
one-hundredth of a frame. one-hundredth of a frame.
smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ] smpte-range = smpte-type "=" smpte-time "-" [ smpte-time ]
smpte-time = 2DIGIT ":" 2DIGIT ":" 2DIGIT [ ":" 2DIGIT ] [ "." 2DIGIT] smpte-type = "smpte" | "smpte-30-drop" | "smpte-25"
; Other timecodes may be added
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ]
[ "." 1*2DIGIT ]
Examples: Examples:
smpte=10:12:33:20- smpte=10:12:33:20-
smpte=10:07:33- smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01 smpte=10:07:00-10:07:33:05.01
smpte-25=10:07:00-10:07:33:05.01
3.6 Normal Play Time 3.6 Normal Play Time
Normal play time (NPT) indicates the stream absolute position Normal play time (NPT) indicates the stream absolute position
relative to the beginning of the presentation. The timestamp consists relative to the beginning of the presentation. The timestamp consists
of a decimal fraction. The part left of the decimal may be expressed of a decimal fraction. The part left of the decimal may be expressed
in either seconds or hours, minutes, and seconds. The part right of in either seconds or hours, minutes, and seconds. The part right of
the decimal point measures fractions of a second. the decimal point measures fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative The beginning of a presentation corresponds to 0.0 seconds. Negative
values are not defined. The special constant now is defined as the values are not defined. The special constant now is defined as the
current instant of a live event. It may be used only for live events. current instant of a live event. It may be used only for live events.
NPT is defined as in DSM-CC: ``Intuitively, NPT is the clock the NPT is defined as in DSM-CC: ``Intuitively, NPT is the clock the
viewer associates with a program. It is often digitally displayed on a viewer associates with a program. It is often digitally displayed on 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
ratio) and is fixed in pause mode. NPT is (logically) equivalent to ratio) and is fixed in pause mode. NPT is (logically) equivalent to
SMPTE time codes.'' [19] SMPTE time codes.'' [5]
npt-range = ( npt-time "-" [ npt-time ] ) | ( "-" npt-time )
npt-time = "now" | npt-sec | npt-hhmmss npt-time = "now" | npt-sec | npt-hhmmss
npt-sec = 1*DIGIT [ "." *DIGIT ] npt-sec = 1*DIGIT [ "." *DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT] npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT]
npt-hh = 1*DIGIT ; any positive number npt-hh = 1*DIGIT ; any positive number
npt-mm = 2DIGIT ; 00-59 npt-mm = 1*2DIGIT ; 0-59
npt-ss = 2DIGIT ; 00-59 npt-ss = 1*2DIGIT ; 0-59
Examples: Examples:
npt=123.45-125 npt=123.45-125
npt=12:05:35.3 npt=12:05:35.3
npt=now npt=now
The syntax conforms to ISO 8601. The npt-sec notation is optimized The syntax conforms to ISO 8601. The npt-sec notation is optimized
for automatic generation, the ntp-hhmmss notation for consumption for automatic generation, the ntp-hhmmss notation for consumption
by human readers. The ``now'' constant allows clients to request to by human readers. The ``now'' constant allows clients to request to
receive the live feed rather than the stored or time-delayed receive the live feed rather than the stored or time-delayed
skipping to change at line 847 skipping to change at line 859
utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction > utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC: UTC:
19961108T143720.25Z 19961108T143720.25Z
3.8 Option Tags 3.8 Option Tags
Option tags are unique identifiers used to designate new options in Option tags are unique identifiers used to designate new options in
RTSP. These tags are used in correspondance with Require RTSP. These tags are used in in Require (Section 12.32) and
(Section 12.32) and Proxy-Require (Section 12.27) fields. Proxy-Require (Section 12.27) header fields.
Syntax: Syntax:
}{ option-tag = 1*xchar
option-tag &=& 1*OCTET &; LWS must be URL-escaped
The creator of a new RTSP option should either prefix the option with The creator of a new RTSP option should either prefix the option with
a reverse domain name (e.g., ``com.foo.mynewfeature'' is apt name for a reverse domain name (e.g., ``com.foo.mynewfeature'' is an apt name
a feature whose inventor can be reached at ``foo.com''), or register for a feature whose inventor can be reached at ``foo.com''), or
the new option with the Internet Assigned Numbers Authority (IANA). register the new option with the Internet Assigned Numbers Authority
(IANA).
3.8.1 Registering New Option Tags With IANA 3.8.1 Registering New Option Tags with IANA
When registering a new RTSP option, the following information should When registering a new RTSP option, the following information should
be provided: be provided:
* Name and description of option. The name may be of any length, but * Name and description of option. The name may be of any length, but
SHOULD be no more than a twenty-character descriptive term. SHOULD be no more than twenty characters long. The name MUST not
contain any spaces, control characters or periods.
* Indication of who has change control over the option (for example, * Indication of who has change control over the option (for example,
IETF, ISO, ITU-T, other international standardization bodies, a IETF, ISO, ITU-T, other international standardization bodies, a
consortium or a particular company or group of companies); consortium or a particular company or group of companies);
* A reference to a further description, if available, for example * A reference to a further description, if available, for example
(in order of preference) an RFC, a published paper, a patent (in order of preference) an RFC, a published paper, a patent
filing, a technical report, documented source code or a computer filing, a technical report, documented source code or a computer
manual; manual;
* For proprietary options, contact information (postal and email * For proprietary options, contact information (postal and email
address); address);
skipping to change at line 924 skipping to change at line 936
4.3 Message Body 4.3 Message Body
See [H4.3] See [H4.3]
4.4 Message Length 4.4 Message Length
When a message body is included with a message, the length of that When a message body is included with a message, the length of that
body is determined by one of the following (in order of precedence): body is determined by one of the following (in order of precedence):
1. 1. Any response message which MUST NOT include a message body
Any response message which MUST NOT include a message body
(such as the 1xx, 204, and 304 responses) is always terminated (such as the 1xx, 204, and 304 responses) is always terminated
by the first empty line after the header fields, regardless of by the first empty line after the header fields, regardless of
the entity-header fields present in the message. (Note: An the entity-header fields present in the message. (Note: An
empty line consists of only CRLF.) empty line consists of only CRLF.)
2.
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 986 skipping to change at line 997
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 10.10 | "RECORD" ; Section 10.11
| "REDIRECT" ; Section | "REDIRECT" ; Section 10.10
| "SETUP" ; Section 10.4 | "SETUP" ; Section 10.4
| "SET_PARAMETER" ; Section 10.9 | "SET_PARAMETER" ; Section 10.9
| "TEARDOWN" ; Section 10.7 | "TEARDOWN" ; Section 10.7
| extension-method | extension-method
extension-method = token extension-method = token
Request-URI = "*" | absolute_URI Request-URI = "*" | absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
skipping to change at line 1083 skipping to change at line 1094
* 4xx: Client Error - The request contains bad syntax or cannot be * 4xx: Client Error - The request contains bad syntax or cannot be
fulfilled fulfilled
* 5xx: Server Error - The server failed to fulfill an apparently * 5xx: Server Error - The server failed to fulfill an apparently
valid request valid request
The individual values of the numeric status codes defined for The individual values of the numeric status codes defined for
RTSP/1.0, and an example set of corresponding Reason-Phrase's, are RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
presented below. The reason phrases listed here are only recommended - presented below. The reason phrases listed here are only recommended -
they may be replaced by local equivalents without affecting the they may be replaced by local equivalents without affecting the
protocol. Note that RTSP adopts most HTTP/1.1 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 starting at x50 to avoid conflicts with
with newly defined HTTP status codes. newly defined HTTP status codes.
Status-Code = "100" ; Continue Status-Code = "100" ; Continue
| "200" ; OK | "200" ; OK
| "201" ; Created | "201" ; Created
| "250" ; Low on Storage Space | "250" ; Low on Storage Space
| "300" ; Multiple Choices | "300" ; Multiple Choices
| "301" ; Moved Permanently | "301" ; Moved Permanently
| "302" ; Moved Temporarily | "302" ; Moved Temporarily
| "303" ; See Other | "303" ; See Other
| "304" ; Not Modified | "304" ; Not Modified
skipping to change at line 1199 skipping to change at line 1209
462 Destination Unreachable 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 551 Option not support all
Status codes and their usage with RTSP methods Table 1: Status codes and their usage with RTSP methods
7.1.2 Response Header Fields 7.1.2 Response Header Fields
The response-header fields allow the request recipient to pass The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the server the Status-Line. These header fields give information about the server
and about further access to the resource identified by the and about further access to the resource identified by the
Request-URI. Request-URI.
response-header = Location ; Section 12.25 response-header = Location ; Section 12.25
skipping to change at line 1292 skipping to change at line 1302
A client that supports persistent connections or connectionless mode A client that supports persistent connections or connectionless mode
MAY ``pipeline'' its requests (i.e., send multiple requests without MAY ``pipeline'' its requests (i.e., send multiple requests without
waiting for each response). A server MUST send its responses to those waiting for each response). A server MUST send its responses to those
requests in the same order that the requests were received. requests in the same order that the requests were received.
9.2 Reliability and Acknowledgements 9.2 Reliability and Acknowledgements
Requests are acknowledged by the receiver unless they are sent to a Requests are acknowledged by the receiver unless they are sent to a
multicast group. If there is no acknowledgement, the sender may resend multicast group. If there is no acknowledgement, the sender may 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 1123), 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 to be aware of whether that the client implementation does not have to be aware of whether
a 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 The Timestamp header (Section 12.38) is used to avoid the
for each request transmitted. If a request is repeated because of lack retransmission ambiguity problem [18, p. 301] and obviates the need
of acknowledgement, the sequence number is incremented. for Karn's algorithm.
This avoids ambiguities when computing round-trip time estimates.
The reliability mechanism described here does not protect against Each request carries a sequence number in the CSeq header
reordering. This may cause problems in some instances. For example, a (Section 12.17), which is incremented by one for each distinct request
TEARDOWN followed by a PLAY has quite a different effect than the transmitted. If a request is repeated because of lack of
reverse. Similarly, if a PLAY request arrives before all parameters acknowledgement, the request MUST carry the original sequence number
are set due to reordering, the media server would have to issue an (i.e. sequence number is not incremented).
error indication. Since sequence numbers for retransmissions are
incremented (to allow easy RTT estimation), the receiver cannot just
ignore out-of-order packets. [TBD: This problem could be fixed by
including both a sequence number that stays the same for
retransmissions and a timestamp for RTT estimation.]
Systems implementing RTSP MUST support carrying RTSP over TCP and MAY Systems implementing RTSP MUST support carrying RTSP over TCP and MAY
support UDP. The default port for the RTSP server is 554 for both UDP support UDP. The default port for the RTSP server is 554 for both UDP
and TCP. and TCP.
A number of RTSP packets destined for the same control end point may A number of RTSP packets destined for the same control end point may
be packed into a single lower-layer PDU or encapsulated into a TCP be packed into a single lower-layer PDU or encapsulated into a TCP
stream. RTSP data MAY be interleaved with RTP and RTCP packets. Unlike stream. RTSP data MAY be interleaved with RTP and RTCP packets. Unlike
HTTP, an RTSP message MUST contain a Content-Length header whenever HTTP, an RTSP message MUST contain a Content-Length header whenever
that message contains a payload. Otherwise, an RTSP packet is that message contains a payload. Otherwise, an RTSP packet is
skipping to change at line 1347 skipping to change at line 1350
The method token indicates the method to be performed on the The method token indicates the method to be performed on the
resource identified by the Request-URI. The method is case-sensitive. resource identified by the Request-URI. The method is case-sensitive.
New methods may be defined in the future. Method names may not start New methods may be defined in the future. Method names may not start
with a $ character (decimal 24) and must be a token. Methods are with a $ character (decimal 24) and must be a token. Methods are
summarized in Table 2. summarized in Table 2.
method direction object requirement method direction object requirement
DESCRIBE C->S P,S recommended DESCRIBE C->S P,S recommended
ANNOUNCE C->S, S->C P,S optional ANNOUNCE C->S, S->C P,S optional
GET_PARAMETER C->S, S->C P,S optional GET_PARAMETER C->S, S->C P,S optional
OPTIONS C->S P,S required OPTIONS C->S, S->C P,S required
(S->C: optional)
PAUSE C->S P,S recommended PAUSE C->S P,S recommended
PLAY C->S P,S required PLAY C->S P,S required
RECORD C->S P,S optional RECORD C->S P,S optional
REDIRECT S->C P,S optional REDIRECT S->C P,S optional
SETUP C->S S required SETUP C->S S required
SET_PARAMETER C->S, S->C P,S optional SET_PARAMETER C->S, S->C P,S optional
TEARDOWN C->S P,S required TEARDOWN C->S P,S required
Overview of RTSP methods, their direction, and what objects (P: Table 2: Overview of RTSP methods, their direction, and what
presentation, S: stream) they operate on objects (P: presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommended, but not required in that a Notes on Table 2: PAUSE is recommended, but not required in that a
fully functional server can be built that does not support this fully functional server can be built that does not support this
method, for example, for live feeds. If a server does not support a method, for example, for live feeds. If a server does not support a
particular method, it MUST return "501 Not Implemented" and a client particular method, it MUST return "501 Not Implemented" and a client
SHOULD not try this method again for this server. SHOULD not try this method again for this server.
10.1 OPTIONS 10.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An OPTIONS The behavior is equivalent to that described in [H9.2]. An OPTIONS
skipping to change at line 1440 skipping to change at line 1444
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.
Media initialization is a requirement for any RTSP-based system, Media initialization is a requirement for any RTSP-based system,
but the RTSP specification doesn't dictate that this must be done but the RTSP specification does not dictate that this must be done
via the DESCRIBE method. There are three ways that an RTSP client via the DESCRIBE method. There are three ways that an RTSP client
may receive initialization information: may receive initialization information:
* Via RTSP's DESCRIBE method * via RTSP's DESCRIBE method;
* Via some other protocol (HTTP, email attachment, etc.) * via some other protocol (HTTP, email attachment, etc.);
* Via the command line or standard input (thus working as a browser * via the command line or standard input (thus working as a browser
helper application launched with an SDP file or other media helper application launched with an SDP file or other media
initialization format) initialization format).
In the interest of practical interoperability, it is highly In the interest of practical interoperability, it is highly
recommended that minimal servers support the DESCRIBE method, and recommended that minimal servers support the DESCRIBE method, and
highly recommended that minimal clients support the ability to act highly recommended that minimal clients support the ability to act
as a ``helper application'' that accepts a media initialization as a ``helper application'' that accepts a media initialization
file from standard input, command line, and/or other means that are file from standard input, command line, and/or other means that are
appropriate to the operating environment of the client. appropriate to the operating environment of the client.
10.3 ANNOUNCE 10.3 ANNOUNCE
skipping to change at line 1543 skipping to change at line 1547
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 requests in
the example below arrive, the server will first play seconds 10 the example below arrive, the server will first play seconds 10
through 15, then, immediately following, seconds 20 to 25, and finally through 15, then, immediately following, seconds 20 to 25, and 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
skipping to change at line 1619 skipping to change at line 1623
10.6 PAUSE 10.6 PAUSE
The PAUSE request causes the stream delivery to be interrupted The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only playback (halted) temporarily. If the request URL names a stream, only 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, though servers MAY close the session and free resources
after being paused for the duration specified with the timeout
parameter of the Session header in the SETUP message.
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834
Session: 1234
S->C: RTSP/1.0 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
The PAUSE request may contain a Range header specifying when the 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 in the first time the server is encountering the time point specified in
any of the currently pending PLAY requests. If the Range header any of the currently pending PLAY requests. If the Range header
specifies a time outside any currently pending PLAY requests, the specifies a time outside any currently pending PLAY requests, the
error ``457 Invalid Range'' is returned. If this header is missing, error ``457 Invalid Range'' is returned. If this header is missing,
stream delivery is interrupted immediately on receipt of the message. stream delivery is interrupted immediately on receipt of the message.
skipping to change at line 1652 skipping to change at line 1668
range, with the second PLAY request effectively being ignored, range, with the second PLAY request effectively being ignored,
assuming the PAUSE request arrives before the server has started assuming the PAUSE request arrives before the server has started
playing the second, overlapping range. Regardless of when the PAUSE playing the second, overlapping range. Regardless of when the PAUSE
request arrives, it sets the NPT to 14. request arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the If the server has already sent data beyond the time specified in the
Range header, a PLAY would still resume at that point in time, as it Range header, a PLAY would still resume at that point in time, as it
is assumed that the client has discarded data after that point. This is assumed that the client has discarded data after that point. This
ensures continuous pause/play cycling without gaps. ensures continuous pause/play cycling without gaps.
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834
Session: 1234
S->C: RTSP/1.0 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
10.7 TEARDOWN 10.7 TEARDOWN
The TEARDOWN request stops the stream delivery for the given URI, The TEARDOWN request stops the stream delivery for the given URI,
freeing the resources associated with it. If the URI is the freeing the resources associated with it. If the URI is the
presentation URI for this presentation, any RTSP session identifier presentation URI for this presentation, any RTSP session identifier
associated with the session is no longer valid. Unless all transport associated with the session is no longer valid. Unless all transport
parameters are defined by the session description, a SETUP request has parameters are defined by the session description, a SETUP request has
to be issued before the session can be played again. to be issued before the session can be played again.
Example: Example:
skipping to change at line 1676 skipping to change at line 1682
presentation URI for this presentation, any RTSP session identifier presentation URI for this presentation, any RTSP session identifier
associated with the session is no longer valid. Unless all transport associated with the session is no longer valid. Unless all transport
parameters are defined by the session description, a SETUP request has parameters are defined by the session description, a SETUP request has
to be issued before the 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
10.8 GET_PARAMETER 10.8 GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a The GET_PARAMETER request retrieves the value of a parameter of a
presentation or stream specified in the URI. presentation or stream specified in the URI. The content of the reply
and response is left to the implementation. GET_PARAMETER with no
The content of the reply and response is left to the implementation. entity body may be used to test client or server liveness (``ping'').
GET_PARAMETER with no entity body may be used to test client or server
liveness (``ping'').
Example: Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 431 CSeq: 431
Content-Type: text/x-rtsp-parameters Content-Type: text/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/parameters
packets_received: 10 packets_received: 10
jitter: 0.3838 jitter: 0.3838
The ``text/x-rtsp-parameters'' section is only an example type for The ``text/parameters'' section is only an example type for
parameter. This method is intentionally loosely defined with the parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be intention that the reply content and response content will be
defined after further experimentation. defined after further experimentation.
10.9 SET_PARAMETER 10.9 SET_PARAMETER
This method requests to set the value of a parameter for a This method requests to set the value of a parameter for a
presentation or stream specified by the URI. presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client A request SHOULD only contain a single parameter to allow the client
to determine why a particular request failed. A server MUST allow a to determine why a particular request failed. If the request contains
several parameters, the server MUST only act on the request if all of
the parameters can be set successfully. A server MUST allow a
parameter to be set repeatedly to the same value, but it MAY disallow parameter to be set repeatedly to the same value, but it MAY disallow
changing parameter values. changing parameter values.
Note: transport parameters for the media stream MUST only be set with Note: transport parameters for the media stream MUST only be set with
the SETUP command. the SETUP command.
Restricting setting transport parameters to SETUP is for the Restricting setting transport parameters to SETUP is for the
benefit of firewalls. benefit of firewalls.
The parameters are split in a fine-grained fashion so that there The parameters are split in a fine-grained fashion so that there
can be more meaningful error indications. However, it may make can be more meaningful error indications. However, it may make
sense to allow the setting of several parameters if an atomic sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client does setting is desirable. Imagine device control where the client does
not want the camera to pan unless it can also tilt to the right not want the camera to pan unless it can also tilt to the right
angle at the same time. angle at the same time.
A SET_PARAMETER request without parameters can be used as a way to
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/x-rtsp-parameters Content-length: 20
Content-type: text/parameters
barparam: barstuff barparam: barstuff
S->C: RTSP/1.0 451 Invalid Parameter S->C: RTSP/1.0 451 Invalid Parameter
CSeq: 421 CSeq: 421
Content-Length: 6 Content-length: 10
Content-type: text/parameters
barparam barparam
The ``text/x-rtsp-parameters'' section is only an example type for The ``text/parameters'' section is only an example type for
parameter. This method is intentionally loosely defined with the parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be intention that the reply content and response content will be
defined after further experimentation. defined after further experimentation.
10.10 REDIRECT 10.10 REDIRECT
A redirect request informs the client that it must connect to A redirect request informs the client that it must connect to
another 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
skipping to change at line 1818 skipping to change at line 1822
Stream data such as RTP packets is encapsulated by an ASCII dollar Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier, followed sign (24 decimal), followed by a one-byte channel identifier, followed
by the length of the encapsulated binary data as a binary, two-byte by the length of the encapsulated binary data as a binary, two-byte
integer in network byte order. The stream data follows immediately integer in network byte order. The stream data follows immediately
afterwards, without a CRLF, but including the upper-layer protocol afterwards, without a CRLF, but including the upper-layer protocol
headers. Each $ block contains exactly one upper-layer protocol data headers. Each $ block contains exactly one upper-layer protocol data
unit, e.g., one RTP packet. unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header with the The channel identifier is defined in the Transport header with the
interleaved parameter 12.39. interleaved parameter(Section 12.39).
When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. As a default, RTCP packets are
sent on the first available channel higher than the RTP channel. The
client MAY explicitly request RTCP packets on another channel. This is
done by specifying two channels in the interleaved parameter of the
Transport header(Section 12.39).
RTCP is needed for synchronization when two or more streams are
interleaved in such a fashion. Also, this provides a convenient way
to tunnel RTP/RTCP packets through the TCP control connection when
required by the network configuration and transfer them onto UDP
when possible.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
CSeq: 2 CSeq: 2
Transport: RTP/AVP/TCP;interleaved=0 Transport: RTP/AVP/TCP;interleaved=0-1
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 2 CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;interleaved=0 Transport: RTP/AVP/TCP;interleaved=0-1
Session: 12345 Session: 12345
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0 C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
CSeq: 3 CSeq: 3
Session: 12345 Session: 12345
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
RTP-Info: url=rtsp://foo.com/bar.file;seq=232433;rtptime=972948234
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header} S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header} S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\001{2 byte length}{"length" bytes RTCP packet}
11 Status Code Definitions 11 Status Code Definitions
Where applicable, HTTP status [H10] codes are reused. Status codes Where applicable, HTTP status [H10] codes are reused. Status codes
that have the same meaning are not repeated here. See Table 1 for a that have the same meaning are not repeated here. See Table 1 for a
listing of which status codes may be returned by which requests. listing of which status codes may be returned by which requests.
11.1 Success 2xx 11.1 Success 2xx
11.1.1 250 Low on Storage Space 11.1.1 250 Low on Storage Space
skipping to change at line 1903 skipping to change at line 1922
may, for example, be the result of a resource reservation failure. may, for example, be the result of a resource reservation failure.
11.3.5 454 Session Not Found 11.3.5 454 Session Not Found
The RTSP session identifier in the Session header is missing, invalid, The RTSP session identifier in the Session header is missing, invalid,
or has timed out. or has timed out.
11.3.6 455 Method Not Valid in This State 11.3.6 455 Method Not Valid in This State
The client or server cannot process this request in its current state. The client or server cannot process this request in its current state.
The response SHOULD contain an Allow header to make error recovery
easier.
11.3.7 456 Header Field Not Valid for Resource 11.3.7 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example, if The server could not act on a required request header. For example, 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.
11.3.8 457 Invalid Range 11.3.8 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the The Range value given is out of bounds, e.g., beyond the end of the
skipping to change at line 1946 skipping to change at line 1967
11.3.13 462 Destination Unreachable 11.3.13 462 Destination Unreachable
The data transmission channel could not be established because the The data transmission channel could not be established because the
client address could not be reached. This error will most likely be client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid Destination the result of a client attempt to place an invalid Destination
parameter in the Transport field. parameter in the Transport field.
11.3.14 551 Option not supported 11.3.14 551 Option not supported
An option given in the Require or the Proxy-Require fields was not An option given in the Require or the Proxy-Require fields was not
supported. supported. The Unsupported header should be returned stating the
option for which there is no support.
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.
Table 3 summarizes the header fields used by RTSP. Type ``g'' Table 3 summarizes the header fields used by RTSP. Type ``g''
designates general request headers to be found in both requests and designates general request headers to be found in both requests and
responses, type ``R'' designates request headers, type ``r'' responses, type ``R'' designates request headers, type ``r''
designates response headers, and type ``e'' designates entity header designates response headers, and type ``e'' designates entity header
fields. Fields marked with ``req.'' in the column labeled ``support'' fields. Fields marked with ``req.'' in the column labeled ``support''
MUST be implemented by the recipient for a particular method, while MUST be implemented by the recipient for a particular method, while
fields marked ``opt.'' are optional. Note that not all fields marked fields marked ``opt.'' are optional. Note that not all fields marked
'r' will be sent in every request of this type. The ``r'' means only ``req.'' will be sent in every request of this type. The ``req.''
that client (for response headers) and server (for request headers) means only that client (for response headers) and server (for request
MUST implement the fields. The last column lists the method for which headers) MUST implement the fields. The last column lists the method
this header field is meaningful; the designation ``entity'' refers to for which this header field is meaningful; the designation ``entity''
all methods that return a message body. Within this specification, refers to all methods that return a message body. Within this
DESCRIBE and GET_PARAMETER fall into this class. specification, DESCRIBE and GET_PARAMETER fall into this class.
If the field content does not apply to the particular resource, the
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
Allow r opt. all
Authorization R opt. all Authorization R opt. all
Bandwidth R opt. all Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control g opt. SETUP Cache-Control g opt. SETUP
Conference R opt. SETUP Conference R opt. SETUP
Connection g req. all Connection g req. all
Content-Base e opt. entity Content-Base e opt. entity
Content-Encoding e req. SET_PARAMETER Content-Encoding e req. SET_PARAMETER
Content-Encoding e req. DESCRIBE, ANNOUNCE Content-Encoding e req. DESCRIBE, ANNOUNCE
Content-Language e req. DESCRIBE, ANNOUNCE Content-Language e req. DESCRIBE, ANNOUNCE
skipping to change at line 2199 skipping to change at line 2219
Conference = "Conference" ":" conference-id Conference = "Conference" ":" conference-id
Example: Example:
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
A response code of 452 (452 Conference Not Found) is returned if the A response code of 452 (452 Conference Not Found) is returned if the
conference-id is not valid. conference-id is not valid.
12.10 Connection 12.10 Connection
See [H14.10]. See [H14.10]
TBD: Connection: timeout=<secs>
12.11 Content-Base 12.11 Content-Base
See [H14.11] See [H14.11]
12.12 Content-Encoding 12.12 Content-Encoding
See [H14.12] See [H14.12]
12.13 Content-Language 12.13 Content-Language
skipping to change at line 2239 skipping to change at line 2257
See [H14.18]. Note that the content types suitable for RTSP are See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and likely to be restricted in practice to presentation descriptions and
parameter-value types. parameter-value types.
12.17 CSeq 12.17 CSeq
The CSeq field specifies the sequence number for an RTSP The CSeq field specifies the sequence number for an RTSP
request-response pair. This field MUST be present in all requests and request-response pair. This field MUST be present in all requests and
responses. For every RTSP request containing the given sequence responses. For every RTSP request containing the given sequence
number, there will be a corresponding response having the same number. number, there will be a corresponding response having the same number.
Any retransmitted request must contain the same sequence number as the
original (i.e. the sequence number is not incremented for
retransmissions of the same request).
12.18 Date 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 the description or media-stream should be considered stale. The
interpretation depends on the method: interpretation depends on the method:
skipping to change at line 2328 skipping to change at line 2349
If-Modified-Since = "If-Modified-Since" ":" HTTP-date If-Modified-Since = "If-Modified-Since" ":" HTTP-date
An example of the field is: An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
12.24 Last-Modified 12.24 Last-Modified
The Last-Modified entity-header field indicates the date and time at The Last-Modified entity-header field indicates the date and time at
which the origin server believes the entity (presentation description which the origin server believes the presentation description or media
or media stream) was last modified. See [H14.29]. For the methods stream was last modified. See [H14.29]. For the methods DESCRIBE or
DESCRIBE or ANNOUNCE, the header field indicates the last modification ANNOUNCE, the header field indicates the last modification date and
date and time of the description, for SETUP that of the media stream. time of the description, for SETUP that of the media stream.
12.25 Location 12.25 Location
See [H14.30]. See [H14.30].
12.26 Proxy-Authenticate 12.26 Proxy-Authenticate
See [H14.33]. See [H14.33].
12.27 Proxy-Require 12.27 Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive The Proxy-Require header is used to indicate proxy-sensitive
features that MUST be supported by the proxy. Any Proxy-Require header features that MUST be supported by the proxy. Any Proxy-Require header
features that are not supported by the proxy MUST be negatively features that are not supported by the proxy MUST be negatively
acknowledged by the proxy to the client if not supported. Servers acknowledged by the proxy to the client if not supported. Servers
should treat this field identically to the Require field. should treat this field identically to the Require field.
See Section 12.32 for more details on the mechanics of this message See Section 12.32 for more details on the mechanics of this message
and a usage example. and a usage example.
We explored using the W3C's PEP proposal [22] for this
functionality. However, Require, Proxy-Require, and Unsupported
allow the addition of extensions with far less complexity.
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 (Section 3.5), npt (Section 3.6), and clock
units. Within RTSP, byte ranges [H14.36.1] are not meaningful and MUST (Section 3.7) range units. Within RTSP, byte ranges [H14.36.1] are not
NOT be used. The header may also contain a time parameter in UTC, meaningful and MUST NOT be used. The header may also contain a time
specifying the time at which the operation is to be made effective. parameter in UTC, specifying the time at which the operation is to be
Servers supporting the Range header MUST understand the NPT range made effective. Servers supporting the Range header MUST understand
format and SHOULD understand the SMPTE range format. The Range the NPT range format and SHOULD understand the SMPTE range format. The
response header indicates what range of time is actually being played Range response header indicates what range of time is actually being
or recorded. played or recorded. If the Range header is given in a time format that
is not understood, the recipient should return ``501 Not
Implemented''.
Range = "Range" ":" 1#ranges-specifier Range = "Range" ":" 1#ranges-specifier
[ ";" "time" "=" utc-time ] [ ";" "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 byterange The notation is similar to that used for the HTTP/1.1 byterange
header. It allows clients to select an excerpt from the media header. It allows clients to select an excerpt from the media
skipping to change at line 2405 skipping to change at line 2422
See [H14.38]. See [H14.38].
12.32 Require 12.32 Require
The Require header is used by clients to query the server about The Require header is used by clients to query the server about
options that it may or may not support. The server MUST respond to options that it may or may not support. The server MUST respond to
this header by using the Unsupported header to negatively acknowledge this header by using the Unsupported header to negatively acknowledge
those options which are NOT supported. those options which are NOT supported.
This is to make sure that the client-server interaction will
proceed without delay when all options are understood by both
sides, and only slow down if options are not understood (as in the
case above). For a well-matched client-server pair, the interaction
proceeds quickly, saving a round-trip often required by negotiation
mechanisms. In addition, it also removes state ambiguity when the
client requires features that the server does not understand.
Require = "Require" ":" 1#option-tag Require = "Require" ":" 1#option-tag
Example: 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/1.0 551 Option not supported S->C: RTSP/1.0 551 Option not supported
CSeq: 302 CSeq: 302
Unsupported: funky-feature Unsupported: funky-feature
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0 C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 303 CSeq: 303
S->C: RTSP/1.0 200 OK S->C: RTSP/1.0 200 OK
CSeq: 303 CSeq: 303
This is to make sure that the client-server interaction will proceed In this example, ``funky-feature'' is the feature tag which indicates
optimally when all options are understood by both sides, and only slow to the client that the fictional Funky-Parameter field is required.
down if options aren't understood (as in the case above). For a The relationship between ``funky-feature'' and Funky-Parameter is not
well-matched client-server pair, the interaction proceeds quickly, communicated via the RTSP exchange, since that relationship is an
saving a round-trip often required by negotiation mechanisms. In immutable property of ``funky-feature'' and thus should not be
addition, it also removes state ambiguity when the client requires transmitted with every exchange.
features that the server doesn't understand.
We explored using the W3C's PEP proposal [22] for this
functionality. However, Require, Proxy-Require, and Unsupported
allow the addition of extensions with far less complexity.
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
This field is used to set RTP-specific parameters in the PLAY This field is used to set RTP-specific parameters in the PLAY
response. response.
skipping to change at line 2478 skipping to change at line 2496
delivered reliably, this mapping is placed in the RTSP control delivered reliably, this mapping is placed in the RTSP control
channel. channel.
In order to compensate for drift for long, uninterrupted In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP, presentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping. reports to check drift against the mapping.
Syntax: Syntax:
RTP-Info = "RTP-Info" ":" 1#stream-url ";" *parameter RTP-Info = "RTP-Info" ":" 1#stream-url 1*parameter
stream-url = "url" "=" url stream-url = "url" "=" url
parameter = ";" "seq" "=" sequence-number parameter = ";" "seq" "=" 1*DIGIT
sequence-number = 1*(DIGIT) | ";" "rtptime" "=" 1*DIGIT
Example: Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=43754027,
url=rtsp://foo.com/bar.avi/streamid=1;seq=34834738 RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
12.34 Scale 12.34 Scale
A scale value of 1 indicates normal play or record at the normal A scale value of 1 indicates normal play or record at the normal
forward viewing rate. If not 1, the value corresponds to the rate with forward viewing rate. If not 1, the value corresponds to the rate with
respect to normal viewing rate. For example, a ratio of 2 indicates respect to normal viewing rate. For example, a ratio of 2 indicates
twice the normal viewing rate (``fast forward'') and a ratio of 0.5 twice the normal viewing rate (``fast forward'') and a ratio of 0.5
indicates half the normal viewing rate. In other words, a ratio of 2 indicates half the normal viewing rate. In other words, a ratio of 2
has normal play time increase at twice the wallclock rate. For every has normal play time increase at twice the wallclock rate. For every
second of elapsed (wallclock) time, 2 seconds of content will be second of elapsed (wallclock) time, 2 seconds of content will be
skipping to change at line 2650 skipping to change at line 2668
remote-controlled denial-of-service attack, a server SHOULD remote-controlled denial-of-service attack, a server SHOULD
authenticate the client and SHOULD log such attempts before authenticate the client and SHOULD log such attempts before
allowing the client to direct a media stream to an address not allowing the client to direct a media stream to an address not
chosen by the server. This is particularly important if RTSP chosen by the server. This is particularly important if RTSP
commands are issued via UDP, but implementations cannot rely on commands are issued via UDP, but implementations cannot rely on
TCP as reliable means of client identification by itself. A TCP as reliable means of client identification by itself. A
server SHOULD not allow a client to direct media streams to an server SHOULD not allow a client to direct media streams to an
address that differs from the address commands are coming from. address that differs from the address commands are coming from.
source: source:
Unicast only. If the source address for the stream is different If the source address for the stream is different than can be
than can be derived from the RTSP endpoint address (the server derived from the RTSP endpoint address (the server in playback
in playback or the client in recording), the source MAY be 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.
append:
If the mode parameter includes RECORD, the append parameter
indicates that the media data should append to the existing indicates that the media data should append to the existing
resource rather than overwrite it. If appending is requested resource rather than overwrite it. If appending is requested
and the server does not support this, it MUST refuse the and the server does not support this, it MUST refuse the
request rather than overwrite the resource identified by the request rather than overwrite the resource identified by the
URI. The append parameter is ignored if the mode parameter does URI. The append parameter is ignored if the mode parameter 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 10.11. control stream, using the mechanism defined in Section 10.12.
The argument provides the channel number to be used in the $ The argument provides the channel number to be used in the $
statement. statement. This parameter may be specified as a range, e.g.,
interleaved=4-5 in cases where the transport choice for the
media stream requires it.
This allows RTP/RTCP to be handled similarly to the way that it is
done with UDP, i.e., one channel for RTP and the other for RTCP.
Multicast specific: Multicast specific:
ttl: ttl:
multicast time-to-live multicast time-to-live
RTP Specific: RTP Specific:
compressed:
Boolean parameter indicating compressed RTP according to RFC
XXXX.
port: port:
This parameter provides the RTP/RTCP port pair for a multicast This parameter provides the RTP/RTCP port pair for a multicast
session. It is specified as a range, e.g., port=3456-3457. session. It is specified as a range, e.g., port=3456-3457.
client_port: client_port:
This parameter provides the unicast RTP/RTCP port pair on the This parameter provides the unicast RTP/RTCP port pair on the
client where media data and control information is to be sent. client where media data and control information is to be sent.
It is specified as a range, e.g., port=3456-3457. It is specified as a range, e.g., port=3456-3457.
server_port: server_port:
This parameter provides the unicast RTP/RTCP port pair on the This parameter provides the unicast RTP/RTCP port pair on the
server where media data and control information is to be sent. server where media data and control information is to be sent.
It is specified as a range, e.g., port=3456-3457. It is specified as a range, e.g., port=3456-3457.
ssrc: ssrc:
The ssrc parameter indicates the RTP SSRC [20, Sec. 3] value The ssrc parameter indicates the RTP SSRC [19, Sec. 3] value
that should be (request) or will be (response) used by the that should be (request) or will be (response) used by the
media server. This parameter is only valid for unicast media server. This parameter is only valid for unicast
transmission. It identifies the synchronization source to be transmission. It identifies the synchronization source to be
associated with the media stream. associated with the media stream.
Transport &=& "Transport" ":" Transport = "Transport" ":"
& & 1#transport-spec 1\#transport-spec
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" | ";" "interleaved" "=" channel [ "-" 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 = <"> *Method <"> | Method
Example: Example:
Transport: RTP/AVP;multicast;compressed;ttl=127;mode="PLAY", Transport: RTP/AVP;multicast;ttl=127;mode="PLAY",
RTP/AVP;unicast;compressed;client_port=3456-3457;mode="PLAY" RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"
The Transport header is restricted to describing a single RTP The Transport header is restricted to describing a single RTP
stream. (RTSP can also control multiple streams as a single stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of of session description formats greatly simplifies designs of
firewalls. firewalls.
12.40 Unsupported 12.40 Unsupported
Negative acknowledgement of features not supported by the server. In The Unsupported response header lists the features not supported by
the case where the feature was specified via the Proxy-Require field the server. In the case where the feature was specified via the
(Section 12.32), if there is a proxy on the path between the client Proxy-Require field (Section 12.32), if there is a proxy on the path
and the server, the proxy MUST insert a message reply with an error between the client and the server, the proxy MUST insert a message
message ``551 Option Not Supported''. reply with an error message ``551 Option Not Supported''.
We explored using the W3C's PEP proposal [22] for this
functionality. However, Require, Proxy-Require, and Unsupported
allow the addition of extensions with far less complexity.
This field roughly corresponds to the PEP-Info and C-PEP-Info in
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]
12.42 Vary 12.42 Vary
See [H14.43] See [H14.43]
skipping to change at line 2850 skipping to change at line 2864
available, dynamic RTP payload types, the protocol stack, and content available, dynamic RTP payload types, the protocol stack, and content
information such as language or copyright restrictions. It may also information such as language or copyright restrictions. It may also
give an indication about the timeline of the movie. give an indication about the timeline of the movie.
In this example, the client is only interested in the last part of the In this example, the client is only interested in the last part of the
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=control:rtsp://audio.example.com/twister/audio.en a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31 m=video 0 RTP/AVP 31
a=control:rtsp://audio.example.com/twister/video a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1 CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057 Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
A->C: RTSP/1.0 200 OK A->C: RTSP/1.0 200 OK
CSeq: 1 CSeq: 1
Session: 1234 Session: 1234
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057; Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
server_port=5000-5001 server_port=5000-5001
skipping to change at line 2890 skipping to change at line 2904
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 Session: 1235
Range: smpte=0:10:00-0:20:00 Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://video.example.com/twister/video;
seq=12312232;rtptime=78712811
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 2 CSeq: 2
Session: 1234 Session: 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 Session: 1234
Range: smpte=0:10:00-0:20:00 Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
seq=876655;rtptime=1032181
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0 C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 3 CSeq: 3
Session: 1234 Session: 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: 4 CSeq: 3
Session: 1235 Session: 1235
V->C: RTSP/1.0 200 OK V->C: RTSP/1.0 200 OK
CSeq: 4 CSeq: 3
Even though the audio and video track are on two different servers, Even though the audio and video track are on two different servers,
and may start at slightly different times and may drift with respect and may start at slightly different times and may drift with respect
to each other, the client can synchronize the two using standard RTP to each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender methods, in particular the time scale contained in the RTCP sender
reports. reports.
14.2 Streaming of a Container file 14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in For purposes of this example, a container file is a storage entity in
which multiple continuous media types pertaining to the same end-user which multiple continuous media types pertaining to the same end-user
skipping to change at line 2988 skipping to change at line 3005
C->M: SETUP rtsp://foo/twister/video RTSP/1.0 C->M: SETUP rtsp://foo/twister/video RTSP/1.0
CSeq: 3 CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003 Transport: RTP/AVP;unicast;client_port=8002-8003
Session: 1234 Session: 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; Transport: RTP/AVP;unicast;client_port=8002-8003;
server_port=9004-9005 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
RTP-Info: url=rtsp://foo/twister/video;
seq=9810092;rtptime=3450012
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0 C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
CSeq: 5 CSeq: 5
Session: 1234 Session: 1234
M->C: RTSP/1.0 460 Only aggregate operation allowed M->C: RTSP/1.0 460 Only aggregate operation allowed
CSeq: 5 CSeq: 5
C->M: PAUSE rtsp://foo/twister RTSP/1.0 C->M: PAUSE rtsp://foo/twister RTSP/1.0
CSeq: 6 CSeq: 6
Session: 1234 Session: 1234
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 6 CSeq: 6
Session: 1234 Session: 1234
skipping to change at line 3039 skipping to change at line 3058
Some RTSP servers may treat all files as though they are ``container Some RTSP servers may treat all files as though they are ``container
files'', yet other servers may not support such a concept. Because of files'', yet other servers may not support such a concept. Because of
this, clients SHOULD use the rules set forth in the session this, clients SHOULD use the rules set forth in the session
description for request URLs, rather than assuming that a consistant description for request URLs, rather than assuming that a consistant
URL may always be used throughout. Here's an example of how a URL may always be used throughout. Here's an example of how a
multi-stream server might expect a single-stream file to be served: multi-stream server might expect a single-stream file to be served:
C->S DESCRIBE rtsp://foo.com/test.wav RTSP/1.0 C->S DESCRIBE rtsp://foo.com/test.wav RTSP/1.0
Accept: application/x-rtsp-mh, application/sdp Accept: application/x-rtsp-mh, application/sdp
CSeq: 2 CSeq: 1
S->C RTSP/1.0 200 OK S->C RTSP/1.0 200 OK
CSeq: 2 CSeq: 1
Content-base: rtsp://foo.com/test.wav/ Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp Content-type: application/sdp
Content-length: 48 Content-length: 48
v=0 v=0
o=- 872653257 872653257 IN IP4 172.16.2.187 o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file s=mu-law wave file
i=audio test i=audio test
t=0 0 t=0 0
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=control:streamid=0 a=control:streamid=0
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0 C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;mode=play Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;mode=play
skipping to change at line 3057 skipping to change at line 3075
v=0 v=0
o=- 872653257 872653257 IN IP4 172.16.2.187 o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file s=mu-law wave file
i=audio test i=audio test
t=0 0 t=0 0
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
a=control:streamid=0 a=control:streamid=0
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0 C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;mode=play Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;mode=play
CSeq: 3 CSeq: 2
S->C RTSP/1.0 200 OK S->C RTSP/1.0 200 OK
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971; Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
server_port=6970-6971;mode=play server_port=6970-6971;mode=play
CSeq: 3 CSeq: 2
Session: 2034820394 Session: 2034820394
C->S PLAY rtsp://foo.com/test.wav RTSP/1.0 C->S PLAY rtsp://foo.com/test.wav RTSP/1.0
CSeq: 4 CSeq: 3
Session: 2034820394 Session: 2034820394
S->C RTSP/1.0 200 OK S->C RTSP/1.0 200 OK
CSeq: 4 CSeq: 3
Session: 2034820394
RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
seq=981888;rtptime=3781123
Note the different URL in the SETUP command, and then the switch back Note the different URL in the SETUP command, and then the switch back
to the aggregate URL in the PLAY command. This makes complete sense to the aggregate URL in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but is less when there are multiple streams with aggregate control, but is less
than intuitive in the special case where the number of streams is one. than intuitive in the special case where the number of streams is one.
In this special case, it is recommended that servers be forgiving of In this special case, it is recommended that servers be forgiving of
implementations that send: implementations that send:
C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0 C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
CSeq: 4 CSeq: 3
In the worst case, servers should send back: In the worst case, servers should send back:
S->C RTSP/1.0 460 Only aggregate operation allowed S->C RTSP/1.0 460 Only aggregate operation allowed
CSeq: 4 CSeq: 3
One would also hope that server implementations are also forgiving of One would also hope that server implementations are also forgiving of
the following: the following:
C->S SETUP rtsp://foo.com/test.wav RTSP/1.0 C->S SETUP rtsp://foo.com/test.wav RTSP/1.0
Transport: rtp/avp/udp;client_port=6970-6971;mode=play Transport: rtp/avp/udp;client_port=6970-6971;mode=play
CSeq: 3 CSeq: 2
Since there is only a single stream in this file, it's not ambiguous Since there is only a single stream in this file, it's not ambiguous
what this means. what this means.
14.4 Live Media Presentation Using Multicast 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.
skipping to change at line 3130 skipping to change at line 3153
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 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: RTP/AVP;multicast 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 Transport: RTP/AVP;multicast;destination=224.2.0.1;port=3456-3457;
ttl=16
Session: 0456804596 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 Session: 0456804596
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
Session: 0456804596 Session: 0456804596
skipping to change at line 3189 skipping to change at line 3212
C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0 C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 3 CSeq: 3
Session: 91389234234 Session: 91389234234
M->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 3 CSeq: 3
14.6 Recording 14.6 Recording
The conference participant client C asks the media server M to record The conference participant client C asks the media server M to record
the audio portion of a meeting. The client uses the ANNOUNCE method to the audio and video portions of a meeting. The client uses the
provide meta-information about the recorded session to the server. ANNOUNCE method to provide meta-information about the recorded session
to the server.
C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0 C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
CSeq: 90 CSeq: 90
Content-Type: application/sdp Content-Type: application/sdp
Content-Length: 121 Content-Length: 121
v=0 v=0
o=camera1 3080117314 3080118787 IN IP4 195.27.192.36 o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
s=IETF Meeting, Munich - 1 s=IETF Meeting, Munich - 1
i=The thirty-ninth IETF meeting will be held in Munich, Germany i=The thirty-ninth IETF meeting will be held in Munich, Germany
skipping to change at line 3216 skipping to change at line 3240
a=tool:sdr v2.4a6 a=tool:sdr v2.4a6
a=type:test a=type:test
m=audio 21010 RTP/AVP 5 m=audio 21010 RTP/AVP 5
c=IN IP4 224.0.1.11/127 c=IN IP4 224.0.1.11/127
a=ptime:40 a=ptime:40
m=video 61010 RTP/AVP 31 m=video 61010 RTP/AVP 31
c=IN IP4 224.0.1.12/127 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->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
C->S: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
CSeq: 91 CSeq: 91
Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=21010-21011; Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=21010-21011;
mode=record;ttl=127 mode=record;ttl=127
S->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 91 CSeq: 91
Session: 508876 Session: 508876
Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=21010-21011; Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=21010-21011;
mode=record;ttl=127 mode=record;ttl=127
C->S: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0 C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
CSeq: 92 CSeq: 92
Session: 508876 Session: 508876
Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=61010-61011; Transport: RTP/AVP;mulicast;destination=224.0.1.12;port=61010-61011;
mode=record;ttl=127 mode=record;ttl=127
S->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 92 CSeq: 92
Transport: RTP/AVP;mulicast;destination=224.0.1.11;port=61010-61011; Transport: RTP/AVP;mulicast;destination=224.0.1.12;port=61010-61011;
mode=record;ttl=127 mode=record;ttl=127
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
CSeq: 93 CSeq: 93
Session: 508876 Session: 508876
Range: clock 19961110T1925-19961110T2015 Range: clock=19961110T1925-19961110T2015
S->C: RTSP/1.0 200 OK M->C: RTSP/1.0 200 OK
CSeq: 93 CSeq: 93
15 Syntax 15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF) The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 (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)>
HT = <US-ASCII HT, horizontal-tab (9)> HT = <US-ASCII HT, horizontal-tab (9)>
<"> = <US-ASCII double-quote mark (34)> <"> = <US-ASCII double-quote mark (34)>
CRLF = CR LF CRLF = CR LF
LWS = [CRLF] 1*( SP | HT ) LWS = [CRLF] 1*( SP | HT )
TEXT = <any OCTET except CTLs> TEXT = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@" tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | "\" | <"> | "," | ";" | ":" | "\" | <">
| "/" | "[" | "]" | "?" | "=" | "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT | "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials> token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> ) quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">> qdtext = <any TEXT except <">>
quoted-pair = "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 field-content = <the OCTETs making up the field-value and
consisting of either *TEXT or consisting of either *TEXT or
combinations of token, tspecials, and combinations of token, tspecials, and
quoted-string> quoted-string>
safe = "\$" | "-" | "_" | "." | "+"
extra = "!" | "*" | "$'$" | "(" | ")" | ","
hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
"a" | "b" | "c" | "d" | "e" | "f"
escape = "\%" hex hex
reserved = ";" | "/" | "?" | ":" | "@" | "&" | "="
unreserved = alpha | digit | safe | extra
xchar = unreserved | reserved | escape
16 Security Considerations 16 Security Considerations
The protocol offers the opportunity for a remote-controlled Because of the similarity in syntax and usage between RTSP servers
denial-of-service attack. and HTTP servers, the security considerations outlined in [H15] apply.
Specifically, please note the following:
The attacker may initiate traffic flows to one or more IP addresses by Authentication Mechanisms:
specifying them as the destination in SETUP requests. While the RTSP and HTTP share common authentication schemes, and thus
attacker's IP address may be known in this case, this is not always should follow the same prescriptions with regards to
useful in prevention of more attacks or ascertaining the attackers authentication. See [H15.1] for client authentication issues,
identity. Thus, an RTSP server SHOULD only allow client-specified and [H15.2] for issues regarding support for multiple
destinations for RTSP-initiated traffic flows if the server has authentication mechanisms.
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 Abuse of Server Log Information:
RTSP session, it is possible for a malicious client to issue requests RTSP and HTTP servers will presumably have similar logging
with random session identifiers which would affect unsuspecting mechanisms, and thus should be equally guarded in protecting
clients. The server SHOULD use a large, random and non-sequential the contents of those logs, thus protecting the privacy of the
session identifier to minimize the possibility of this kind of attack. users of the servers. See [H15.3] for HTTP server
recommendations regarding server logs.
Servers SHOULD implement both basic and digest [8] authentication. In Transfer of Sensitive Information:
environments requring tighter security for the control messages, There is no reason to believe that information transferred via
transport layer mechanims such as [7] SHOULD be used. RTSP may be any less sensitive than that normally transmitted
via HTTP. Therefore, all of the precautions regarding the
protection of data privacy and user privacy apply to
implementors of RTSP clients, servers, and proxies. See [H15.4]
for further details.
In addition, the security considerations outlined in [H15] apply. Attacks Based On File and Path Names:
Though RTSP URLs are opaque handles that do not necessarily
have file system semantics, it is anticipated that many
implementations will translate portions of the request URLs
directly to file system calls. In such cases, file systems
SHOULD follow the precautions outlined in [H15.5], such as
checking for ``..'' in path components.
Personal Information:
RTSP clients are often privy to the same information that HTTP
clients are (user name, location, etc.) and thus should be
equally. See [H15.6] for further recommendations.
Privacy Issues Connected to Accept Headers:
Since may of the same ``Accept'' headers exist in RTSP as in
HTTP, the same caveats outlined in [H15.7] with regards to
their use should be followed.
DNS Spoofing:
Presumably, given the longer connection times typically
associated to RTSP sessions relative to HTTP sessions, RTSP
client DNS optimizations should be less prevalent. Nonetheless,
the recommendations provided in [H15.8] are still relevant to
any implementation which attempts to rely on a DNS-to-IP
mapping to hold beyond a single use of the mapping.
Location Headers and Spoofing:
If a single server supports multiple organizations that do not
trust one another, then it must check the values of Location
and Content-Location headers in responses that are generated
under control of said organizations to make sure that they do
not attempt to invalidate resources over which they have no
authority. ([H15.9])
In addition to the recommendations in the current HTTP specification
(RFC 2068, as of this writing), future HTTP specifications may provide
additional guidance on security issues.
The following are added considerations for RTSP implementations.
Concentrated Denial-Of-Service:
The protocol offers the opportunity for a remote-controlled
denial-of-service attack. The attacker may initiate traffic
flows to one or more IP addresses by specifying them as the
destination in SETUP requests. While the attacker's IP address
may be known in this case, this is not always useful in
prevention of more attacks or ascertaining the attackers
identity. Thus, an RTSP server SHOULD only allow
client-specified destinations for RTSP-initiated traffic flows
if the server has verified the client's identity, either
against a database of known users using RTSP authentication
mechanisms (preferrably digest authentication or stronger), or
other secure means.
Session Hijacking:
Since there is no relation between a transport layer connection
and an RTSP session, it is possible for a malicious client to
issue requests with random session identifiers which would
affect unsuspecting clients. The server SHOULD use a large,
random and non-sequential session identifier to minimize the
possibility of this kind of attack.
Authentication:
Servers SHOULD implement both basic and digest [6]
authentication. In environments requiring tighter security for
the control messages, transport layer mechanisms such as TLS
(RFC XXXX) SHOULD be used.
Stream issues:
RTSP only provides for stream control. Stream delivery issues
are not covered in this section, nor in the rest of this draft.
RTSP implementations will most likely rely on other protocols
such as RTP, IP Multicast, RSVP, and IGMP, and should address
considerations brought up in these specifications (even when
non-standard equivalents are used in place of said protocols).
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
skipping to change at line 3452 skipping to change at line 3568
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 media clients to control selected, non-contiguous RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP sections of media presentations, rendering those streams with an RTP
media layer[20]. The media layer rendering the RTP stream should not media layer[19]. 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.
skipping to change at line 3492 skipping to change at line 3608
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. first sequence number of the next 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, RFC XXXX) may be used to
streams or presentations in RTSP. Such usage is limited to specifying describe streams or presentations in RTSP. Such usage is limited to
means of access and encoding(s) for: specifying means of access and encoding(s) for:
aggregate control: aggregate control:
A presentation composed of streams from one or more servers A presentation composed of streams from one or more servers
that are not available for aggregate control. Such a that are not available for aggregate control. Such a
description is typically retrieved by HTTP or other non-RTSP description is typically retrieved by HTTP or other non-RTSP
means. However, they may be received with ANNOUNCE methods. means. However, they may be received with ANNOUNCE methods.
non-aggregate control: non-aggregate control:
A presentation composed of multiple streams from a single A presentation composed of multiple streams from a single
server that are available for aggregate control. Such a server that are available for aggregate control. Such a
skipping to change at line 3524 skipping to change at line 3640
describes how a client should interpret SDP content returned in reply describes how a client should interpret SDP content returned in reply
to a DESCRIBE request. SDP provides no mechanism by which a client can to a DESCRIBE request. SDP provides no mechanism by which a client can
distinguish, without human guidance, between several media streams to distinguish, without human guidance, between several media streams to
be rendered simultaneously and a set of alternatives (e.g., two audio be rendered simultaneously and a set of alternatives (e.g., two audio
streams spoken in different languages). streams spoken in different languages).
C.1 Definitions C.1 Definitions
The terms ``session-level'', ``media-level'' and other key/attribute The terms ``session-level'', ``media-level'' and other key/attribute
names and values used in this appendix are to be used as defined in names and values used in this appendix are to be used as defined in
[6]. RFC XXXX (SDP):
C.1.1 Control URL C.1.1 Control URL
The ``a=control:'' attribute is used to convey the control URL. This The ``a=control:'' attribute is used to convey the control URL. This
attribute is used both for the session and media descriptions. If used attribute is used both for the session and media descriptions. If used
for individual media, it indicates the URL to be used for controlling for individual media, it indicates the URL to be used for controlling
that particular media stream. If found at the session level, the that particular media stream. If found at the session level, the
attribute indicates the URL for aggregate control. attribute indicates the URL for aggregate control.
Example: Example:
a=control:rtsp://example.com/foo a=control:rtsp://example.com/foo
This attribute may contain either relative and absolute URLs, This attribute may contain either relative and absolute URLs,
following the rules and conventions set out in RFC 1808 ([16]). following the rules and conventions set out in RFC 1808 ([20]).
Implementations should look for a base URL in the following order: Implementations should look for a base URL in the following order:
1. The RTSP Content-Base field 1. The RTSP Content-Base field
2. The RTSP Content-Location field 2. The RTSP Content-Location field
3. The RTSP request URL 3. The RTSP request URL
If this attribute contains only an asterisk (*), then the URL is If this attribute contains only an asterisk (*), then the URL is
treated as if it were an empty embedded URL, and thus inherits the treated as if it were an empty embedded URL, and thus inherits the
entire base URL. entire base URL.
skipping to change at line 3569 skipping to change at line 3685
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 RFC XXXX. Codec-specific
are not specified in this field, but rather in the ``fmtp'' attribute parameters are not specified in this field, but rather in the ``fmtp''
described below. Implementors seeking to register new encodings should attribute described below. Implementors seeking to register new
follow the procedure in RFC 1890. If the media type is not suited to encodings should follow the procedure in RFC 1890. If the media type
the RTP AV profile, then it is recommended that a new profile be is not suited to the RTP AV profile, then it is recommended that a new
created and the appropriate profile name be used in lieu of profile be created and the appropriate profile name be used in lieu of
``RTP/AVP'' in the ``m='' field. ``RTP/AVP'' in the ``m='' field.
C.1.4 Format-specific parameters C.1.4 Format-specific parameters
Format-specific parameters are conveyed using the ``fmtp'' media Format-specific parameters are conveyed using the ``fmtp'' media
attribute. The syntax of the ``fmtp'' attribute is specific to the attribute. The syntax of the ``fmtp'' attribute is specific to the
encoding(s) that the attribute refers to. Note that the packetization encoding(s) that the attribute refers to. Note that the packetization
interval is conveyed using the ``ptime'' attribute. interval is conveyed using the ``ptime'' attribute.
C.1.5 Length of presentation C.1.5 Range of presentation
The ``a=length'' attribute defines the total length of stored The ``a=range'' attribute defines the total time range of the stored
sessions. (The length of live sessions can be deduced from the ``t'' session. (The length of live sessions can be deduced from the ``t''
and ``r'' parameters.) Unless the presentation contains media streams and ``r'' parameters.) Unless the presentation contains media streams
of different durations, the length attribute is a session-level of different durations, the length attribute is a session-level
attribute. The unit is specified first, followed by the value. The attribute. The unit is specified first, followed by the value range.
units and their values are as defined in Section 3.5, 3.6 and 3.7. The units and their values are as defined in Section 3.5, 3.6 and 3.7.
Example: Examples:
a=length:npt=34.4368 a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203
C.1.6 Time of availability C.1.6 Time of availability
The ``t='' field MUST contain suitable values for the start and stop The ``t='' field MUST contain suitable values for the start and stop
times for both aggregate and non-aggregate stream control. With times for both aggregate and non-aggregate stream control. With
aggregate control, the server SHOULD indicate a stop time value for aggregate control, the server SHOULD indicate a stop time value for
which it guarantees the description to be valid, and a start time that which it guarantees the description to be valid, and a start time that
is equal to or before the time at which the DESCRIBE request was is equal to or before the time at which the DESCRIBE request was
received. It MAY also indicate start and stop times of 0, meaning that received. It MAY also indicate start and stop times of 0, meaning that
the session is always available. With non-aggregate control, the the session is always available. With non-aggregate control, the
values should reflect the actual period for which the session is values should reflect the actual period for which the session is
skipping to change at line 3748 skipping to change at line 3865
There may be RTSP applications different from those initially There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for which envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict recommendations above serve only as guidelines instead of strict
requirements. requirements.
D.1.1 Basic Playback D.1.1 Basic Playback
To support on-demand playback of media streams, the client MUST To support on-demand playback of media streams, the client MUST
additionally be able to do the following: additionally be able to do the following:
* Include and parse the Range header, with NPT units. * generate the PAUSE request;
* Generate the PAUSE request. * implement the REDIRECT method, and the Location header.
* Implement the REDIRECT method, and the Location header.
D.1.2 Authentication-enabled D.1.2 Authentication-enabled
In order to access media presentations from RTSP servers that require In order to access media presentations from RTSP servers that require
authentication, the client MUST additionally be able to do the authentication, the client MUST additionally be able to do the
following: following:
* Recognize the 401 status code. * recognize the 401 status code;
* Parse and include the WWW-Authenticate header * parse and include the WWW-Authenticate header;
* Implement Basic Authentication and Digest Authentication * implement Basic Authentication and Digest Authentication.
D.2 Server D.2 Server
A minimal server implementation MUST be able to do the following: A minimal server implementation MUST be able to do the following:
* Implement the following methods: SETUP, TEARDOWN, OPTIONS and * Implement the following methods: SETUP, TEARDOWN, OPTIONS and
either PLAY (for a minimal playback server) or RECORD (for a either PLAY (for a minimal playback server) or RECORD (for a
minimal recording server). minimal recording server).
If RECORD is implemented, ANNOUNCE should be implemented 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,
skipping to change at line 3796 skipping to change at line 3913
envisioned by the contributors to the RTSP specification for which envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict recommendations above serve only as guidelines instead of strict
requirements. requirements.
D.2.1 Basic Playback D.2.1 Basic Playback
To support on-demand playback of media streams, the server MUST To support on-demand playback of media streams, the server MUST
additionally be able to do the following: additionally be able to do the following:
* Include and parse the Range header, with NPT units. Implementation * Recognize the Range header, and return an error if seeking is not
of SMPTE units is recommended. supported.
* Implement the PAUSE method. * Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface In addition, in order to support commonly-accepted user interface
features, the following are highly recommended for on-demand media features, the following are highly recommended for on-demand media
servers: servers:
* Include and parse the Range header, with NPT units. Implementation
of SMPTE units is recommended.
* Include the length of the media presentation in the media * Include the length of the media presentation in the media
initialization information. initialization information.
* Include mappings from data-specific timestamps to NPT. When RTP is * Include mappings from data-specific timestamps to NPT. When RTP is
used, the rtptime portion of the RTP-Info field may be used to map used, the rtptime portion of the RTP-Info field may be used to map
RTP timestamps to NPT. RTP timestamps to NPT.
Client implementations may use the presence of length information Client implementations may use the presence of length information
to determine if the clip is seekable, and visably disable seeking to determine if the clip is seekable, and visably disable seeking
features for clips for which the length information is unavailable. features for clips for which the length information is unavailable.
A common use of the presentation length is to implement a ``slider A common use of the presentation length is to implement a ``slider
skipping to change at line 3832 skipping to change at line 3951
In order to correctly handle client authentication, the server MUST In order to correctly handle client authentication, the server MUST
additionally be able to do the following: additionally be able to do the following:
* Generate the 401 status code when authentication is required for * Generate the 401 status code when authentication is required for
the resource. the resource.
* Parse and include the WWW-Authenticate header * Parse and include the WWW-Authenticate header
* Implement Basic Authentication and Digest Authentication * Implement Basic Authentication and Digest Authentication
E Changes E Changes
Since draft 05 (October 28, 1997 version) of RTSP, the following
changes were made:
* Added reference to Timestamp: header
* Added some RTP-Info headers to PLAY responses in example code.
* Added atomicity wording to SET_PARAMETER.
* Added support for smpte-25
* Added Allow header to header table.
* Changed smpte and npt to allow 1*2DIGIT.
* Changed RTP-Info from providing the last sequence number of the
previous segment to first sequence number of the next segment.
* Changed SDP a=length to a=range.
* Described ``append'' Transport parameter further.
* Fixed bugs in CSeq wording (was per packet, now per request).
* Fleshed out security section reference to HTTP by explaining why
each of the HTTP recommendations are applicable to RTSP.
* Allow server initiated OPTIONS exchange
* Fixed wording on the Range header support for minimal
implementations.
* Updated section and example to interleave RTCP packets on the TCP
connection well.
Since draft04 (September 17, 1997 version) of RTSP, the following Since draft04 (September 17, 1997 version) of RTSP, the following
changes were made: changes were made:
* Further explanation of container files and how to deal with * Further explanation of container files and how to deal with
``single-stream container files''. ``single-stream container files''.
* IANA procedure for registering option tags. * IANA procedure for registering option tags.
* New response codes (``461 Unsupported Transport'', ``462 * New response codes (``461 Unsupported Transport'', ``462
Destination Unreachable'', ``551 Option Not Supported''). Destination Unreachable'', ``551 Option Not Supported'').
* Practical minimum implementations established in Appendix D. * Practical minimum implementations established in Appendix D.
* Removed quasi-specification of ``text/rtsp-parameters'' with the * Removed quasi-specification of ``text/rtsp-parameters'' with the
intent to define this separately. intent to define this separately.
* Closed out open issues * Closed out open issues
* Inserted ommisions in ``Since draft03...'' below (``etag'' * Inserted ommisions in ``Since draft03...'' below (``etag''
change). change).
* Addition of ``etag'' mechanism in SDP, and corresponding If-Match
field.
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. Motivation: We explored
using the W3C's PEP proposal for this functionality. However,
Require, Proxy-Require, and Unsupported allow the addition of
extensions with far less complexity. The Proxy-Require field
roughly corresponds to the C-PEP field in the PEP draft. The
Require field roughly corresponds to the PEP field in the PEP
draft. The Unsupported field roughly corresponds to the PEP-Info
and C-PEP-Info in the PEP draft.
* 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 from the request and
response lines, and put into a new CSeq: header. response lines into its own CSeq header.
* Interaction with RTP appendix added. * Appendix detailing interaction with RTP 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: * Addition of etag mechanism in SDP, and corresponding If-Match
field. field.
Between draft02 (March, 1997) and draft03 (July, 1997), the following Between draft 02 (March, 1997) and draft 03 (July, 1997), the
changes were made: following 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
skipping to change at line 3928 skipping to change at line 4078
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, Torsten Braun, Brent Browning, Bruce Butterfield, Ema
Martin Dunsmuir, Eric Fleischman, V. Guruprasad, Peter Haight, Mark Patki, Steve Casner, Francisco Cortes, Kelly Djahandari, Martin
Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka, Anders Klemets, Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad,
Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Rob Peter Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp
McCool, David Oran, Sujal Patel, Alagu Periyannan, Igor Plotnikov, Hoschka, Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif,
Pinaki Shah, Jeff Smith, Alexander Sokolsky, Dale Stammen, and John Jonathan Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria
Francis Stracke. Papadopouli, Sujal Patel, Alagu Periyannan, Igor Plotnikov, Pinaki
Shah, David Singer, 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 F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
mechanism,'' RFC 2109, Internet Engineering Task Force, Feb.
1997.
3 F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
``Internationalization of the hypertext markup language,'' RFC ``Internationalization of the hypertext markup language,'' RFC
2070, Internet Engineering Task Force, Jan. 1997. 2070, Internet Engineering Task Force, Jan. 1997.
4 S. Bradner, ``Key words for use in RFCs to indicate requirement 3 S. Bradner, ``Key words for use in RFCs to indicate requirement
levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997. levels,'' RFC 2119, Internet Engineering Task Force, Mar. 1997.
5 R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. 4 R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T.
Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC Berners-Lee, ``Hypertext transfer protocol - HTTP/1.1,'' RFC
2068, Internet Engineering Task Force, Jan. 1997. 2068, Internet Engineering Task Force, Jan. 1997.
6 M. Handley, ``SDP: Session description protocol,'' Internet 5 ISO/IEC, ``Information technology - generic coding of moving
Draft, Internet Engineering Task Force, Nov. 1996. pictures and associated audio informaiton - part 6: extension
Work in progress. for digital storage media and control,'' Draft International
Standard ISO 13818-6, International Organization for
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Internet Draft, Internet Engineering Task Force, Dec. 1996. Nov. 1995.
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Jan. 1997.
9 J. Postel, ``User datagram protocol,'' STD 6, RFC 768, Internet 7 J. Postel, ``User datagram protocol,'' RFC STD 6, 768, Internet
Engineering Task Force, Aug. 1980. Engineering Task Force, Aug. 1980.
10 R. Hinden and C. Partridge, ``Version 2 of the reliable data 8 B. Hinden and C. Partridge, ``Version 2 of the reliable data
protocol (RDP),'' RFC 1151, Internet Engineering Task Force, protocol (RDP),'' RFC 1151, Internet Engineering Task Force,
Apr. 1990. Apr. 1990.
11 J. Postel, ``Transmission control protocol,'' STD 7, RFC 793, 9 J. Postel, ``Transmission control protocol,'' RFC STD 7, 793,
Internet Engineering Task Force, Sept. 1981. Internet Engineering Task Force, Sept. 1981.
12 M. Handley, H. Schulzrinne, and E. Schooler, ``SIP: Session 10 H. Schulzrinne, ``A comprehensive multimedia control
initiation protocol,'' Internet Draft, Internet Engineering architecture for the Internet,'' in Proc. International
Task Force, Dec. 1996. Workshop on Network and Operating System Support for Digital
Work in progress. Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997.
13 P. McMahon, ``GSS-API authentication method for SOCKS version 11 P. McMahon, ``GSS-API authentication method for SOCKS version
5,'' RFC 1961, Internet Engineering Task Force, June 1996. 5,'' RFC 1961, Internet Engineering Task Force, June 1996.
14 D. Crocker, ``Augmented BNF for syntax specifications: ABNF,'' 12 J. Miller, P. Resnick, and D. Singer, ``Rating services and
Internet Draft, Internet Engineering Task Force, Oct. 1996. rating systems (and their machine readable descriptions),''
Work in progress. Recommendation REC-PICS-services-961031, W3C (World Wide Web
Consortium), Boston, Massachusetts, Oct. 1996.
13 J. Miller, T. Krauskopf, P. Resnick, and W. Treese, ``PICS
label distribution label syntax and communication protocols,''
Recommendation REC-PICS-labels-961031, W3C (World Wide Web
Consortium), Boston, Massachusetts, Oct. 1996.
14 D. Crocker and P. Overell, ``Augmented BNF for syntax
specifications: ABNF,'' RFC 2234, Internet Engineering Task
Force, Nov. 1997.
15 R. Elz, ``A compact representation of IPv6 addresses,'' RFC 15 R. Elz, ``A compact representation of IPv6 addresses,'' RFC
1924, Internet Engineering Task Force, Apr. 1996. 1924, Internet Engineering Task Force, Apr. 1996.
16 R. Fielding, ``Relative Uniform Resource Locators,'' RFC 1808, 16 T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform
Internet Engineering Task Force, June 1995.
17 T. Berners-Lee, L. Masinter, and M. McCahill, ``Uniform
resource locators (URL),'' RFC 1738, Internet Engineering Task resource locators (URL),'' RFC 1738, Internet Engineering Task
Force, Dec. 1994. Force, Dec. 1994.
18 International Telecommunication Union, ``Visual telephone 17 International Telecommunication Union, ``Visual telephone
systems and equipment for local area networks which provide a systems and equipment for local area networks which provide a
non-guaranteed quality of service,'' Recommendation H.323, non-guaranteed quality of service,'' Recommendation H.323,
Telecommunication Standardization Sector of ITU, Geneva, Telecommunication Standardization Sector of ITU, Geneva,
Switzerland, May 1996. Switzerland, May 1996.
19 ISO/IEC, ``Information technology - generic coding of moving 18 W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
pictures and associated audio informaiton - part 6: extension Reading, Massachusetts: Addison-Wesley, 1994.
for digital storage media and control,'' Draft International
Standard ISO 13818-6, International Organization for
Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
Nov. 1995.
20 H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, 19 H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson,
``RTP: a transport protocol for real-time applications,'' RFC ``RTP: a transport protocol for real-time applications,'' RFC
1889, Internet Engineering Task Force, Jan. 1996. 1889, Internet Engineering Task Force, Jan. 1996.
21 J. Miller, P. Resnick, and D. Singer, ``Rating Services and 20 R. Fielding, ``Relative uniform resource locators,'' RFC 1808,
Rating Systems(and Their Machine Readable Descriptions), '' Internet Engineering Task Force, June 1995.
REC-PICS-services-961031, Worldwide Web Consortium, Oct. 1996.
22 D. Connolly, R. Khare, H.F. Nielsen, ``PEP - an Extension
Mechanism for HTTP", Internet draft, work-in-progress. W3C
Draft WD-http-pep-970714
http://www.w3.org/TR/WD-http-pep-970714, July, 1996.
Full Copyright Statement Full Copyright Statement
Copyright (C) The Internet Society (1997). All Rights Reserved. Copyright (C) The Internet Society (1997). All Rights Reserved.
This document and translations of it may be copied and furnished to This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published and or assist in its implmentation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind, distributed, in whole or in part, without restriction of any kind,
provided that the above copyright notice and this paragraph are provided that the above copyright notice and this paragraph are
 End of changes. 

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