Internet Engineering Task Force MMUSIC WG
Internet Draft H. Schulzrinne
Columbia U.
A. Rao
Cisco
R. Lanphier
RealNetworks
draft-ietf-mmusic-rfc2326bis-00.txt
February 22, 2002
Expires: July 2002
Real Time Streaming Protocol (RTSP)
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
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Abstract
This memorandum is a revision of RFC 2326, which is currently a
Proposed Standard.
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and provide a means for choosing delivery
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mechanisms based upon RTP (RFC 1889).
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1 Introduction
1.1 Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls
either a single or several time-synchronized streams of continuous
media such as audio and video. It does not typically deliver the
continuous streams itself, although interleaving of the continuous
media stream with the control stream is possible (see Section 10.12).
In other words, RTSP acts as a "network remote control" for
multimedia servers.
The set of streams to be controlled is defined by a presentation
description. This memorandum does not define a format for a
presentation description.
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
to a transport-level connection such as a TCP connection. During an
RTSP session, an RTSP client may open and close many reliable
transport connections to the server to issue RTSP requests.
Alternatively, it may use a connectionless transport protocol such as
UDP.
The streams controlled by RTSP may use RTP [1], but the operation of
RTSP does not depend on the transport mechanism used to carry
continuous media.
The protocol is intentionally similar in syntax and operation to
HTTP/1.1 [2] so that extension mechanisms to HTTP can in most cases
also be added to RTSP. However, RTSP differs in a number of important
aspects from HTTP:
o RTSP introduces a number of new methods and has a different
protocol identifier.
o An RTSP server needs to maintain state by default in almost
all cases, as opposed to the stateless nature of HTTP.
o Both an RTSP server and client can issue requests.
o Data is carried out-of-band by a different protocol. (There is
an exception to this.)
o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
8859-1, consistent with current HTML internationalization
efforts [3].
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o The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1 [2]
carries only the absolute path in the request and puts the
host name in a separate header field.
This makes "virtual hosting" easier, where a single
host with one IP address hosts several document trees.
The protocol supports the following operations:
Retrieval of media from media server: The client can request a
presentation description via HTTP or some other method. If
the presentation is being multicast, the presentation
description contains the multicast addresses and ports to
be used for the continuous media. If the presentation is
to be sent only to the client via unicast, the client
provides the destination for security reasons.
Invitation of a media server to a conference: A media server can
be "invited" to join an existing conference, either to play
back media into the presentation or to record all or a
subset of the media in a presentation. This mode is useful
for distributed teaching applications. Several parties in
the conference may take turns "pushing the remote control
buttons".
Addition of media to an existing presentation: Particularly for
live presentations, it is useful if the server can tell the
client about additional media becoming available.
RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1 [2].
1.2 Requirements
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [4].
1.3 Terminology
Some of the terminology has been adopted from HTTP/1.1 [2]. Terms
not listed here are defined as in HTTP/1.1.
Aggregate control: The control of the multiple streams using a
single timeline by the server. For audio/video feeds, this
means that the client may issue a single play or pause
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message to control both the audio and video feeds.
Conference: a multiparty, multimedia presentation, where "multi"
implies greater than or equal to one.
Client: The client requests continuous media data from the media
server.
Connection: A transport layer virtual circuit established
between two programs for the purpose of communication.
Container file: A file which may contain multiple media streams
which often comprise a presentation when played together.
RTSP servers may offer aggregate control on these files,
though the concept of a container file is not embedded in
the protocol.
Continuous media: Data where there is a timing relationship
between source and sink; that is, the sink must reproduce
the timing relationship that existed at the source. The
most common examples of continuous media are audio and
motion video. Continuous media can be real-time
(interactive) , where there is a "tight" timing
relationship between source and sink, or streaming
(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: Datatype/codec specific initialization.
This includes such things as clockrates, color tables, etc.
Any transport-independent information which is required by
a client for playback of a media stream occurs in the media
initialization phase of stream setup.
Media parameter: Parameter specific to a media type that may be
changed before or during stream playback.
Media server: The server providing playback or recording
services for one or more media streams. Different media
streams within a presentation may originate from different
media servers. A media server may reside on the same or a
different host as the web server the presentation is
invoked from.
Media server indirection: Redirection of a media client to a
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different media server.
(Media) stream: A single media instance, e.g., an audio stream
or a video stream as well as a single whiteboard or shared
application group. When using RTP, a stream consists of all
RTP and RTCP packets created by a source within an RTP
session. This is equivalent to the definition of a DSM-CC
stream([5]).
Message: The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined
in Section 15 and transmitted via a connection or a
connectionless protocol.
Participant: Member of a conference. A participant may be a
machine, e.g., a media record or playback server.
Presentation: A set of one or more streams presented to the
client as a complete media feed, using a presentation
description as defined below. In most cases in the RTSP
context, this implies aggregate control of those streams,
but does not have to.
Presentation description: A presentation description contains
information about one or more media streams within a
presentation, such as the set of encodings, network
addresses and information about the content. Other IETF
protocols such as SDP (RFC 2327 [6]) use the term "session"
for a live presentation. The presentation description may
take several different formats, including but not limited
to the session description format SDP.
Response: An RTSP response. If an HTTP response is meant, that
is indicated explicitly.
Request: An RTSP request. If an HTTP request is meant, that is
indicated explicitly.
RTSP session: A complete RTSP "transaction", e.g., the viewing
of a movie. A session typically consists of a client
setting up a transport mechanism for the continuous media
stream ( SETUP), starting the stream with PLAY or RECORD,
and closing the stream with TEARDOWN.
Transport initialization: The negotiation of transport
information (e.g., port numbers, transport protocols)
between the client and the server.
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1.4 Protocol Properties
RTSP has the following properties:
Extendable: New methods and parameters can be easily added to
RTSP.
Easy to parse: RTSP can be parsed by standard HTTP or MIME
parsers.
Secure: RTSP re-uses web security mechanisms, either at the
transport level (TLS, RFC 2246 [7]) or within the protocol
itself. All HTTP authentication mechanisms such as basic
(RFC 2068 [2]) and digest authentication (RFC 2069 [8]) are
directly applicable.
Transport-independent: RTSP may use either an unreliable
datagram protocol (UDP) (RFC 768 [9]), a reliable datagram
protocol (RDP, RFC 1151, not widely used [10]) or a
reliable stream protocol such as TCP (RFC 793 [11]) as it
implements application-level reliability.
Multi-server capable: Each media stream within a presentation
can reside on a different server. The client automatically
establishes several concurrent control sessions with the
different media servers. Media synchronization is
performed at the transport level.
Control of recording devices: The protocol can control both
recording and playback devices, as well as devices that can
alternate between the two modes ("VCR").
Separation of stream control and conference initiation: Stream
control is divorced from inviting a media server to a
conference. The only requirement is that the conference
initiation protocol either provides or can be used to
create a unique conference identifier. In particular, SIP
[12] or H.323 [13] may be used to invite a server to a
conference.
Suitable for professional applications: RTSP supports frame-
level accuracy through SMPTE time stamps to allow remote
digital editing.
Presentation description neutral: The protocol does not impose a
particular presentation description or metafile format and
can convey the type of format to be used. However, the
presentation description must contain at least one RTSP
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URI.
Proxy and firewall friendly: The protocol should be readily
handled by both application and transport-layer (SOCKS
[14]) firewalls. A firewall may need to understand the
SETUP method to open a "hole" for the UDP media stream.
HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so
that the existing infrastructure can be reused. This
infrastructure includes PICS (Platform for Internet Content
Selection [15,16]) for associating labels with content.
However, RTSP does not just add methods to HTTP since the
controlling continuous media requires server state in most
cases.
Appropriate server control: 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 way that clients cannot stop
the stream.
Transport negotiation: The client can negotiate the transport
method prior to actually needing to process a continuous
media stream.
Capability negotiation: If basic features are disabled, there
must be some clean mechanism for the client to determine
which methods are not going to be implemented. This allows
clients to present the appropriate user interface. For
example, if seeking is not allowed, the user interface must
be able to disallow moving a sliding position indicator.
An earlier requirement in RTSP was multi-client capability.
However, it was determined that a better approach was to
make sure that the protocol is easily extensible to the
multi-client scenario. Stream identifiers can be used by
several control streams, so that "passing the remote" would
be possible. The protocol would not address how several
clients negotiate access; this is left to either a "social
protocol" or some other floor control mechanism.
1.5 Extending RTSP
Since not all media servers have the same functionality, media
servers by necessity will support different sets of requests. For
example:
o A server may only be capable of playback thus has no need to
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support the RECORD request.
o A server may not be capable of seeking (absolute positioning)
if it is to support live events only.
o Some servers may not support setting stream parameters and
thus not support GET_PARAMETER and SET_PARAMETER.
A server SHOULD implement all header fields described in Section 12.
It is up to the creators of presentation descriptions not to ask the
impossible of a server. This situation is similar in HTTP/1.1 [2],
where the methods described in [H19.6] are not likely to be supported
across all servers.
RTSP can be extended in three ways, listed here in order of the
magnitude of changes supported:
o Existing methods can be extended with new parameters, as long
as these parameters can be safely ignored by the recipient.
(This is equivalent to adding new parameters to an HTML tag.)
If the client needs negative acknowledgement when a method
extension is not supported, a tag corresponding to the
extension may be added in the Require: field (see Section
12.33).
o New methods can be added. If the recipient of the message does
not understand the request, it responds with error code 501
(Not Implemented) and the sender should not attempt to use
this method again. A client may also use the OPTIONS method
to inquire about methods supported by the server. The server
SHOULD list the methods it supports using the Public response
header.
o A new version of the protocol can be defined, allowing almost
all aspects (except the position of the protocol version
number) to change.
1.6 Overall Operation
Each presentation and media stream may be identified by an RTSP URL.
The overall presentation and the properties of the media the
presentation is made up of are defined by a presentation description
file, the format of which is outside the scope of this specification.
The presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored
on the media server.
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For the purposes of this specification, a presentation description is
assumed to describe one or more presentations, each of which
maintains a common time axis. For simplicity of exposition and
without loss of generality, it is assumed that the presentation
description contains exactly one such presentation. A presentation
may contain several media streams.
The presentation description file contains a description of the media
streams making up the presentation, including their encodings,
language, and other parameters that enable the client to choose the
most appropriate combination of media. In this presentation
description, each media stream that is individually controllable by
RTSP is identified by an RTSP URL, which points to the media server
handling that particular media stream and names the stream stored on
that server. Several media streams can be located on different
servers; for example, audio and video streams can be split across
servers for load sharing. The description also enumerates which
transport methods the server is capable of.
Besides the media parameters, the network destination address and
port need to be determined. Several modes of operation can be
distinguished:
Unicast: The media is transmitted to the source of the RTSP
request, with the port number chosen by the client.
Alternatively, the media is transmitted on the same
reliable stream as RTSP.
Multicast, server chooses address: The media server picks the
multicast address and port. This is the typical case for a
live or near-media-on-demand transmission.
Multicast, client chooses address: If the server is to
participate in an existing multicast conference, the
multicast address, port and encryption key are given by the
conference description, established by means outside the
scope of this specification.
1.7 RTSP States
RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel. For example, RTSP control may
occur on a TCP connection while the data flows via UDP. Thus, data
delivery continues even if no RTSP requests are received by the media
server. Also, during its lifetime, a single media stream may be
controlled by RTSP requests issued sequentially on different TCP
connections. Therefore, the server needs to maintain "session state"
to be able to correlate RTSP requests with a stream. The state
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transitions are described in Section A.
Many methods in RTSP do not contribute to state. However, the
following play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and
TEARDOWN.
SETUP: Causes the server to allocate resources for a stream and
start an RTSP session.
PLAY and RECORD: Starts data transmission on a stream allocated
via SETUP.
PAUSE: Temporarily halts a stream without freeing server
resources.
TEARDOWN: Frees resources associated with the stream. The RTSP
session ceases to exist on the server.
RTSP methods that contribute to state use the Session
header field (Section 12.38) to identify the RTSP session
whose state is being manipulated. The server generates
session identifiers in response to SETUP requests (Section
10.4).
1.8 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may
interact 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 allow different hand-off points between a web
server and the media server implementing RTSP. For example, the
presentation description can be retrieved using HTTP or RTSP, which
reduces roundtrips in web-browser-based scenarios, yet also allows
for standalone RTSP servers and clients which do not rely on HTTP at
all.
However, RTSP differs fundamentally from HTTP in that data delivery
takes place out-of-band in a different protocol. HTTP is an
asymmetric protocol where the client issues requests and the server
responds. In RTSP, both the media client and media server can issue
requests. RTSP requests are also not stateless; they may set
parameters and continue to control a media stream long after the
request has been acknowledged.
Re-using HTTP functionality has advantages in at least two
areas, namely security and proxies. The requirements are
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very similar, so having the ability to adopt HTTP work on
caches, proxies and authentication is valuable.
While most real-time media will use RTP as a transport protocol, RTSP
is not tied to RTP.
RTSP assumes the existence of a presentation description format that
can express both static and temporal properties of a presentation
containing several media streams.
2 Notational Conventions
Since many of the definitions and syntax are identical to HTTP/1.1,
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 Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]).
All the mechanisms specified in this document are described in both
prose and an augmented Backus-Naur form (BNF) similar to that used in
[H2.1]. It is described in detail in RFC 2234 [17], with the
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
background and motivation. This is intended to give readers who were
not involved with the formulation of the specification an
understanding of why things are the way that they are in RTSP.
3 Protocol Parameters
3.1 RTSP Version
applies, with HTTP replaced by RTSP.
3.2 RTSP URL
The "rtsp" and "rtspu" schemes are used to refer to network resources
via the RTSP protocol. This section defines the scheme-specific
syntax and semantics for RTSP URLs.
rtsp_URL _ ( "rtsp:" | "rtspu:" )
"//" host [ ":" port ] [ abs_path ]
host _ <A legal Internet host domain name of IP address
(in dotted decimal form), as defined by Section 2.1
of RFC 1123 [18]>
port _ *DIGIT
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abs_path is defined in [H3.2.1].
Note that fragment and query identifiers do not have a
well-defined meaning at this time, with the interpretation
left to the RTSP server.
The scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtspu
identifies an unreliable protocol (within the Internet, UDP).
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 server listening for TCP (scheme "rtsp") connections or UDP
(scheme "rtspu") packets on that port of host, and the Request-URI
for the resource is rtsp_URL.
The use of IP addresses in URLs SHOULD be avoided whenever possible
(see RFC 1924 [19]).
A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions [H3.2] of
URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of
streams, i.e., a presentation. Accordingly, requests described in
Section 10 can apply to either the whole presentation or an
individual stream within the presentation. Note that some request
methods can only be applied to streams, not presentations and vice
versa.
For example, the RTSP URL:
rtsp://media.example.com:554/twister/audiotrack
identifies the audio stream within the presentation "twister", which
can be controlled via RTSP requests issued over a TCP connection to
port 554 of host media.example.com
Also, the RTSP URL:
rtsp://media.example.com:554/twister
identifies the presentation "twister", which may be composed of audio
and video streams.
This does not imply a standard way to reference streams in
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URLs. The presentation description defines the hierarchical
relationships in the presentation and the URLs for the
individual streams. A presentation description may name a
stream "a.mov" and the whole presentation "b.mov".
The path components of the RTSP URL are opaque to the client and do
not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be
used with non-RTSP media control protocols simply by
replacing the scheme in the URL.
3.3 Conference Identifiers
Conference identifiers are opaque to RTSP and are encoded using
standard URI encoding methods (i.e., LWS is escaped with %). They can
contain any octet value. The conference identifier MUST be globally
unique. For H.323, the conferenceID value is to be used.
conference-id _ 1*xchar
Conference identifiers are used to allow RTSP sessions to
obtain parameters from multimedia conferences the media
server is participating in. These conferences are created
by protocols outside the scope of this specification, e.g.,
H.323 [13] or SIP [12]. Instead of the RTSP client
explicitly providing transport information, for example, it
asks the media server to use the values in the conference
description instead.
3.4 Session Identifiers
Session identifiers are opaque strings of arbitrary length. Linear
white space must be URL-escaped. A session identifier MUST be chosen
randomly and MUST be at least eight octets long to make guessing it
more difficult. (See Section 16.)
session-id _ 8*( ALPHA | DIGIT | safe )
3.5 SMPTE Relative Timestamps
A SMPTE relative timestamp expresses time relative to the start of
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the clip. Relative timestamps are expressed as SMPTE time codes for
frame-level access accuracy. The time code has the format
hours:minutes:seconds:frames.subframes
,
with the origin at the start of the clip. The default smpte format
is"SMPTE 30 drop" format, with frame rate is 29.97 frames per second.
Other SMPTE codes MAY be supported (such as "SMPTE 25") through the
use of alternative use of "smpte time". For the "frames" field in the
time value can assume the values 0 through 29. The difference between
30 and 29.97 frames 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.
smpte-range _ smpte-type "=" smpte-range-spec
smpte-range-spec _ ( smpte-time "-" [ smpte-time ] ) | ( "-" smpte-time )
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:
smpte=10:12:33:20-
smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01
smpte-25=10:07:00-10:07:33:05.01
3.6 Normal Play Time
Normal play time (NPT) indicates the stream absolute position
relative to the beginning of the presentation. The timestamp consists
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
the decimal point measures fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative
values are not defined. The special constant now is defined as the
current instant of a live event. It may be used only for live events.
NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the
viewer associates with a program. It is often digitally displayed on
a VCR. NPT advances normally when in normal play mode (scale = 1),
advances at a faster rate when in fast scan forward (high positive
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scale ratio), decrements when in scan reverse (high negative scale
ratio) and is fixed in pause mode. NPT is (logically) equivalent to
SMPTE time codes." [5]
npt-range _ ["npt" "="] npt-range-spec
; implementations SHOULD use npt= prefix, but SHOULD
; be prepared to interoperate with RFC 2326
; implementations which don't use it
npt-range-spec _ ( npt-time "-" [ npt-time ] ) | ( "-" npt-time )
npt-time _ "now" | npt-sec | npt-hhmmss
npt-sec _ 1*DIGIT [ "." *DIGIT ]
npt-hhmmss _ npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
npt-hh _ 1*DIGIT ; any positive number
npt-mm _ 1*2DIGIT ; 0-59
npt-ss _ 1*2DIGIT ; 0-59
Examples:
npt=123.45-125
npt=12:05:35.3-
npt=now-
The syntax conforms to ISO 8601. The npt-sec notation is
optimized for automatic generation, the ntp-hhmmss notation
for consumption by human readers. The "now" constant allows
clients to request to receive the live feed rather than the
stored or time-delayed version. This is needed since
neither absolute time nor zero time are appropriate for
this case.
3.7 Absolute Time
Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
Fractions of a second may be indicated.
utc-range _ "clock" "=" utc-time "-" [ utc-time ]
utc-time _ utc-date "T" utc-time "Z"
utc-date _ 8DIGIT ; < YYYYMMDD >
utc-time _ 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC:
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19961108T143720.25Z
3.8 Option Tags
Option tags are unique identifiers used to designate new options in
RTSP. These tags are used in in Require (Section 12.33) and Proxy-
Require (Section 12.28) header fields.
Syntax:
option-tag _ token
The creator of a new RTSP option should either prefix the option with
a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name
for a feature whose inventor can be reached at "foo.com"), or
register the new option with the Internet Assigned Numbers Authority
(IANA).
3.8.1 Registering New Option Tags with IANA
When registering a new RTSP option, the following information should
be provided:
o Name and description of option. The name may be of any length,
but SHOULD be no more than twenty characters long. The name
MUST not contain any spaces, control characters or periods.
o Indication of who has change control over the option (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium or a particular company or group of
companies);
o A reference to a further description, if available, for
example (in order of preference) an RFC, a published paper, a
patent filing, a technical report, documented source code or a
computer manual;
o For proprietary options, contact information (postal and email
address);
4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [22]). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by
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themselves as line terminators.
Text-based protocols make it easier to add optional
parameters in a self-describing manner. Since the number of
parameters and the frequency of commands is low, processing
efficiency is not a concern. Text-based protocols, if done
carefully, also allow easy implementation of research
prototypes in scripting languages such as Tcl, Visual Basic
and Perl.
The 10646 character set avoids tricky character set switching, but is
invisible to the application as long as US-ASCII is being used. This
is also the encoding used for RTCP. ISO 8859-1 translates directly
into Unicode with a high-order octet of zero. ISO 8859-1 characters
with the most-significant bit set are represented as 1100001x
10xxxxxx. (See RFC 2279 [22])
RTSP messages can be carried over any lower-layer transport protocol
that is 8-bit clean.
Requests contain methods, the object the method is operating upon and
parameters to further describe the method. Methods are idempotent,
unless otherwise noted. Methods are also designed to require little
or no state maintenance at the media server.
4.1 Message Types
See [H4.1]
4.2 Message Headers
See [H4.2]
4.3 Message Body
See [H4.3]
4.4 Message Length
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):
1. Any response message which MUST NOT include a message body
(such as the 1xx, 204, and 304 responses) is always
terminated by the first empty line after the header fields,
regardless of the entity-header fields present in the
message. (Note: An empty line consists of only CRLF.)
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2. If a Content-Length header field (section 12.15) is
present, its value in bytes represents the length of the
message-body. If this header field is not present, a value
of zero is assumed.
Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
transfer coding(see [H3.6]) and requires the presence of the
Content-Length header field.
Given the moderate length of presentation descriptions
returned, the server should always be able to determine its
length, even if it is generated dynamically, making the
chunked transfer encoding unnecessary.
5 General Header Fields
See [H4.5], except that Pragma, Transfer-Encoding and Upgrade
headers are not defined:
general-header = Cache-Control ; Section 12.9
| Connection ; Section 12.11
| CSeq ; Section 12.18
| Date ; Section 12.19
| Via ; Section 12.44
6 Request
A request message from a client to a server or vice versa includes,
within the first line of that message, the method to be applied to
the resource, the identifier of the resource, and the protocol
version in use.
Request = Request-Line ; Section 6.1
*( general-header ; Section 5
| request-header ; Section 6.2
| entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
6.1 Request Line
Request-Line _ Method SP Request-URI SP RTSP-Version CRLF
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Method = "DESCRIBE" ; Section 10.2
| "ANNOUNCE" ; Section 10.3
| "GET_PARAMETER" ; Section 10.8
| "OPTIONS" ; Section 10.1
| "PAUSE" ; Section 10.6
| "PLAY" ; Section 10.5
| "RECORD" ; Section 10.11
| "REDIRECT" ; Section 10.10
| "SETUP" ; Section 10.4
| "SET_PARAMETER" ; Section 10.9
| "TEARDOWN" ; Section 10.7
| extension-method
extension-method _ token
Request-URI _ "*" | absolute_URI
RTSP-Version _ "RTSP" "/" 1*DIGIT "." 1*DIGIT
6.2 Request Header Fields
request-header = Accept ; Section 12.1
| Accept-Encoding ; Section 12.2
| Accept-Language ; Section 12.3
| Authorization ; Section 12.6
| Bandwidth ; Section 12.7
| Blocksize ; Section 12.8
| Conference ; Section 12.10
| From ; Section 12.21
| If-Modified-Since ; Section 12.24
| Proxy-Require ; Section 12.28
| Range ; Section 12.30
| Referer ; Section 12.31
| Require ; Section 12.33
| Scale ; Section 12.35
| Session ; Section 12.38
| Speed ; Section 12.36
| Transport ; Section 12.40
| User-Agent ; Section 12.42
Note that in contrast to HTTP/1.1 [2], RTSP requests always contain
the absolute URL (that is, including the scheme, host and port)
rather than just the absolute path.
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HTTP/1.1 requires servers to understand the absolute URL,
but clients are supposed to use the Host request header.
This is purely needed for backward-compatibility with
HTTP/1.0 servers, a consideration that does not apply to
RTSP.
The asterisk "*" in the Request-URI means that the request does not
apply to a particular resource, but to the server itself, and is only
allowed when the method used does not necessarily apply to a
resource. One example would be:
OPTIONS * RTSP/1.0
7 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version.
Also, RTSP defines additional status codes and does not define some
HTTP codes. The valid response codes and the methods they can be used
with are defined in Table 1.
After receiving and interpreting a request message, the recipient
responds with an RTSP response message.
Response = Status-Line ; Section 7.1
*( general-header ; Section 5
| response-header ; Section 7.1.2
| entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
7.1 Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version followed by a numeric status code, and the
textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
Status-Line _ RTSP-Version SP Status-Code SP Reason-Phrase CRLF
7.1.1 Status Code and Reason Phrase
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The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully
defined in Section 11. The Reason-Phrase is intended to give a short
textual description of the Status-Code. The Status-Code is intended
for use by automata and the Reason-Phrase is intended for the human
user. The client is not required to examine or display the Reason-
Phrase.
The first digit of the Status-Code defines the class of response.
The last two digits do not have any categorization role. There are 5
values for the first digit:
o 1xx: Informational - Request received, continuing process
o 2xx: Success - The action was successfully received,
understood, and accepted
o 3xx: Redirection - Further action must be taken in order to
complete the request
o 4xx: Client Error - The request contains bad syntax or cannot
be fulfilled
o 5xx: Server Error - The server failed to fulfill an apparently
valid request
The individual values of the numeric status codes defined for
RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
presented below. The reason phrases listed here are only recommended
-- they may be replaced by local equivalents without affecting the
protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and
adds RTSP-specific status codes starting at x50 to avoid conflicts
with newly defined HTTP status codes.
Status-Code = "100" ; Continue
| "200" ; OK
| "201" ; Created
| "250" ; Low on Storage Space
| "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "304" ; Not Modified
| "305" ; Use Proxy
| "400" ; Bad Request
| "401" ; Unauthorized
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| "402" ; Payment Required
| "403" ; Forbidden
| "404" ; Not Found
| "405" ; Method Not Allowed
| "406" ; Not Acceptable
| "407" ; Proxy Authentication Required
| "408" ; Request Time-out
| "410" ; Gone
| "411" ; Length Required
| "412" ; Precondition Failed
| "413" ; Request Entity Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "451" ; Parameter Not Understood
| "452" ; Conference Not Found
| "453" ; Not Enough Bandwidth
| "454" ; Session Not Found
| "455" ; Method Not Valid in This State
| "456" ; Header Field Not Valid for Resource
| "457" ; Invalid Range
| "458" ; Parameter Is Read-Only
| "459" ; Aggregate operation not allowed
| "460" ; Only aggregate operation allowed
| "461" ; Unsupported transport
| "462" ; Destination unreachable
| "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Time-out
| "505" ; RTSP Version not supported
| "551" ; Option not supported
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
RTSP status codes are extensible. RTSP applications are not required
to understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if an
unrecognized status code of 431 is received by the client, it can
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safely assume that there was something wrong with its request and
treat the response as if it had received a 400 status code. In such
cases, user agents SHOULD present to the user the entity returned
with the response, since that entity is likely to include human-
readable information which will explain the unusual status.
7.1.2 Response Header Fields
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the
server and about further access to the resource identified by the
Request-URI.
response-header = Location ; Section 12.26
| Proxy-Authenticate ; Section 12.27
| Public ; Section 12.29
| Range ; Section 12.30
| Retry-After ; Section 12.32
| RTP-Info ; Section 12.34
| Scale ; Section 12.35
| Session ; Section 12.38
| Server ; Section 12.37
| Speed ; Section 12.36
| Transport ; Section 12.40
| Unsupported ; Section 12.41
| Vary ; Section 12.43
| WWW-Authenticate ; Section 12.45
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as
entity-header fields.
8 Entity
Request and Response messages MAY transfer an entity if not otherwise
restricted by the request method or response status code. An entity
consists of entity-header fields and an entity-body, although some
responses will only include the entity-headers.
In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity.
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8.1 Entity Header Fields
Entity-header fields define optional metainformation about the
entity-body or, if no body is present, about the resource identified
by the request.
entity-header = Allow ; Section 12.5
| Content-Base ; Section 12.12
| Content-Encoding ; Section 12.13
| Content-Language ; Section 12.14
| Content-Length ; Section 12.15
| Content-Location ; Section 12.16
| Content-Type ; Section 12.17
| Expires ; Section 12.20
| Last-Modified ; Section 12.25
| extension-header
extension-header = message-header
The extension-header mechanism allows additional entity-header fields
to be defined without changing the protocol, but these fields cannot
be assumed to be recognizable by the recipient. Unrecognized header
fields SHOULD be ignored by the recipient and forwarded by proxies.
8.2 Entity Body
See [H7.2]
9 Connections
RTSP requests can be transmitted in several different ways:
o persistent transport connections used for several request-
response transactions;
o one connection per request/response transaction;
o connectionless mode.
The type of transport connection is defined by the RTSP URI (Section
3.2). For the scheme "rtsp", a persistent connection is assumed,
while the scheme "rtspu" calls for RTSP requests to be sent without
setting up a connection.
Unlike HTTP, RTSP allows the media server to send requests to the
media client. However, this is only supported for persistent
connections, as the media server otherwise has no reliable way of
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Code reason
_______________________________________________________
100 Continue all
_______________________________________________________
200 OK all
201 Created RECORD
250 Low on Storage Space RECORD
_______________________________________________________
300 Multiple Choices all
301 Moved Permanently all
302 Moved Temporarily all
303 See Other all
305 Use Proxy all
_______________________________________________________
400 Bad Request all
401 Unauthorized all
402 Payment Required all
403 Forbidden all
404 Not Found all
405 Method Not Allowed all
406 Not Acceptable all
407 Proxy Authentication Required all
408 Request Timeout all
410 Gone all
411 Length Required all
412 Precondition Failed DESCRIBE, SETUP
413 Request Entity Too Large all
414 Request-URI Too Long all
415 Unsupported Media Type all
451 Parameter Not Understood SETUP
452 Illegal Conference Identifier SETUP
453 Not Enough Bandwidth SETUP
454 Session Not Found all
455 Method Not Valid In This State all
456 Header Field Not Valid all
457 Invalid Range PLAY
458 Parameter Is Read-Only SET_PARAMETER
459 Aggregate Operation Not Allowed all
460 Only Aggregate Operation Allowed all
461 Unsupported Transport all
462 Destination Unreachable all
_______________________________________________________
500 Internal Server Error all
501 Not Implemented all
502 Bad Gateway all
503 Service Unavailable all
504 Gateway Timeout all
505 RTSP Version Not Supported all
551 Option not support all
Table 1: Status codes and their usage with RTSP methods
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reaching the client. Also, this is the only way that requests from
media server to client are likely to traverse firewalls.
9.1 Pipelining
A client that supports persistent connections or connectionless mode
MAY "pipeline" its requests (i.e., send multiple requests without
waiting for each response). A server MUST send its responses to those
requests in the same order that the requests were received.
9.2 Reliability and Acknowledgements
Requests are acknowledged by the receiver unless they are sent to a
multicast group. If there is no acknowledgement, the sender may
resend the same message after a timeout of one round-trip time (RTT).
The round-trip time is estimated as in TCP (RFC 1123) [18], with an
initial round-trip value of 500 ms. An implementation MAY cache the
last RTT measurement as the initial value for future connections.
If a reliable transport protocol is used to carry RTSP, requests MUST
NOT be retransmitted; the RTSP application MUST instead rely on the
underlying transport to provide reliability.
If both the underlying reliable transport such as TCP and
the RTSP application retransmit requests, it is possible
that each packet loss results in two retransmissions. The
receiver cannot typically take advantage of the
application-layer retransmission since the transport stack
will not deliver the application-layer retransmission
before the first attempt has reached the receiver. If the
packet loss is caused by congestion, multiple
retransmissions at different layers will exacerbate the
congestion.
If RTSP is used over a small-RTT LAN, standard procedures for
optimizing initial TCP round trip estimates, such as those used in
T/TCP (RFC 1644) [23], can be beneficial.
The Timestamp header (Section 12.39) is used to avoid the
retransmission ambiguity problem [24] and obviates the need for
Karn's algorithm.
Each request carries a sequence number in the CSeq header (Section
12.18), which is incremented by one for each distinct request
transmitted. If a request is repeated because of lack of
acknowledgement, the request MUST carry the original sequence number
(i.e., the sequence number is not incremented).
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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
and TCP.
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
stream. RTSP data MAY be interleaved with RTP and RTCP packets.
Unlike HTTP, an RTSP message MUST contain a Content-Length header
field whenever that message contains a payload. Otherwise, an RTSP
packet is terminated with an empty line immediately following the
last message header.
10 Method Definitions
The method token indicates the method to be performed on the resource
identified by the Request-URI case-sensitive. 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 summarized in Table 2.
method direction object requirement
__________________________________________________________________
DESCRIBE C -> S P,S recommended
ANNOUNCE C -> S, S -> C P,S optional
GET_PARAMETER C -> S, S -> C P,S optional
OPTIONS C -> S, S -> C P,S required (S -> C: optional)
PAUSE C -> S P,S recommended
PLAY C -> S P,S required
RECORD C -> S P,S optional
REDIRECT S -> C P,S optional
SETUP C -> S S required
SET_PARAMETER C -> S, S -> C P,S optional
TEARDOWN C -> S P,S required
Table 2: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommended, but not required in that a
fully functional server can be built that does not support this
method, for example, for live feeds. If a server does not support a
particular method, it MUST return 501 (Not Implemented) and a client
SHOULD not try this method again for this server.
10.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An OPTIONS
request may be issued at any time, e.g., if the client is about to
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try a nonstandard request. It does not influence server state.
Example:
C->S: OPTIONS * RTSP/1.0
CSeq: 1
Require: implicit-play
Proxy-Require: gzipped-messages
S->C: RTSP/1.0 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are necessarily fictional features (one would hope
that we would not purposefully overlook a truly useful feature just
so that we could have a strong example in this section).
10.2 DESCRIBE
The DESCRIBE method retrieves the description of a presentation or
media object identified by the request URL from a server. It may use
the Accept header to specify the description formats that the client
understands. The server responds with a description of the requested
resource. The DESCRIBE reply-response pair constitutes the media
initialization phase of RTSP.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 OK
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 376
v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
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c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
m=whiteboard 32416 UDP WB
a=orient:portrait
The DESCRIBE response MUST contain all media initialization
information for the resource(s) that it describes. If a media client
obtains a presentation description from a source other than DESCRIBE
and that description contains a complete set of media initialization
parameters, the client SHOULD use those parameters and not then
request a description for the same media via RTSP.
Additionally, servers SHOULD NOT use the DESCRIBE response as a means
of media indirection.
By forcing a DESCRIBE response to contain all media
initialization for the set of streams that it describes,
and discouraging use of DESCRIBE for media indirection, we
avoid looping problems that might result from other
approaches.
Media initialization is a requirement for any RTSP-based system, but
the RTSP specification does not dictate that this must be done via
the DESCRIBE method. There are three ways that an RTSP client may
receive initialization information:
o via RTSP's DESCRIBE method;
o via some other protocol (HTTP, email attachment, etc.);
o via the command line or standard input (thus working as a
browser helper application launched with an SDP file or other
media initialization format).
It is RECOMMENDED that minimal servers support the DESCRIBE method,
and highly recommended that minimal clients support the ability to
act as a "helper application" that accepts a media initialization
file from standard input, command line, and/or other means that are
appropriate to the operating environment of the client.
10.3 ANNOUNCE
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The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a
presentation or media object identified by the request URL to a
server. When sent from server to client, ANNOUNCE updates the
session description in real-time.
If a new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent
again, rather than just the additional components, so that components
can be deleted.
Example:
C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Content-Type: application/sdp
Content-Length: 332
v=0
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
S->C: RTSP/1.0 200 OK
CSeq: 312
10.4 SETUP
The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. A client can issue a SETUP request for a
stream that is already playing to change transport parameters, which
a server MAY allow. If it does not allow this, it MUST respond with
error 455 (Method Not Valid In This State). For the benefit of any
intervening firewalls, a client must indicate the transport
parameters even if it has no influence over these parameters, for
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example, where the server advertises a fixed multicast address.
Since SETUP includes all transport initialization
information, firewalls and other intermediate network
devices (which need this information) are spared the more
arduous task of parsing the DESCRIBE response, which has
been reserved for media initialization.
The Transport header specifies the transport parameters acceptable
to the client for data transmission; the response will contain the
transport parameters selected by the server.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Transport: RTP/AVP;unicast;client_port=4588-4589
S->C: RTSP/1.0 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;
client_port=4588-4589;server_port=6256-6257
The server generates session identifiers in response to SETUP
requests. If a SETUP request to a server includes a session
identifier, the server MUST bundle this setup request into the
existing session or return error 459 (Aggregate Operation Not
Allowed) (see Section 11.4.10).
10.5 PLAY
The PLAY method tells the server to start sending data via the
mechanism specified in SETUP. A client MUST NOT issue a PLAY request
until any outstanding SETUP requests have been acknowledged as
successful.
The PLAY request positions the normal play time to the beginning of
the range specified and delivers stream data until the end of the
range is reached. PLAY requests may be pipelined (queued); a server
MUST queue PLAY requests to be executed in order. That is, a PLAY
request arriving while a previous PLAY request is still active is
delayed until the first has been completed.
This allows precise editing. For example, regardless of
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how closely spaced the two PLAY requests in the example
below arrive, the server will first play seconds 10 through
15, then, immediately following, seconds 20 to 25, and
finally seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 835
Session: 12345678
Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 836
Session: 12345678
Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 837
Session: 12345678
Range: npt=30-
See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It starts playing a
stream from the beginning unless the stream has been paused. If a
stream has been paused via PAUSE, stream delivery resumes at the
pause point. If a stream is playing, such a PLAY request causes no
further action and can be used by the client to test server liveness.
The Range header may also contain a time parameter. This parameter
specifies a time in UTC at which the playback should start. If the
message is received after the specified time, playback is started
immediately. The time parameter may be used to aid in
synchronization of streams obtained from different sources.
For a on-demand stream, the server replies with the actual range that
will be played back. This may differ from the requested range if
alignment of the requested range to valid frame boundaries is
required for the media source. If no range is specified in the
request, the current position is returned in the reply. The unit of
the range in the reply is the same as that in the request.
After playing the desired range, the presentation is automatically
paused, as if a PAUSE request had been issued.
The following example plays the whole presentation starting at SMPTE
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time code 0:10:20 until the end of the clip. The playback is to start
at 15:36 on 23 Jan 1997.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
CSeq: 833
Session: 12345678
Range: smpte=0:10:20-;time=19970123T153600Z
S->C: RTSP/1.0 200 OK
CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT
Range: smpte=0:10:22-;time=19970123T153600Z
For playing back a recording of a live presentation, it may be
desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
CSeq: 835
Session: 12345678
Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats.
10.6 PAUSE
The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only
playback and recording of that stream is halted. For example, for
audio, this is equivalent to muting. If the request URL names a
presentation or group of streams, delivery of all currently active
streams within the presentation or group is halted. After resuming
playback or recording, synchronization of the tracks MUST be
maintained. Any server resources 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.
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Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834
Session: 12345678
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
stream or presentation is to be halted. We refer to this point as the
"pause point". The header must contain exactly one value rather than
a time range. The normal play time for the stream is set to the pause
point. The pause request becomes effective the first time the server
is encountering the time point specified in any of the currently
pending PLAY requests. If the Range header specifies a time outside
any currently pending PLAY requests, the error 457 (Invalid Range) is
returned. If a media unit (such as an audio or video frame) starts
presentation at exactly the pause point, it is not played or
recorded. If the Range header is missing, stream delivery is
interrupted immediately on receipt of the message and the pause point
is set to the current normal play time.
A PAUSE request discards all queued PLAY requests. However, the pause
point in the media stream MUST be maintained. A subsequent PLAY
request without Range header resumes from the pause point.
For example, if the server has play requests for ranges 10 to 15 and
20 to 29 pending and then receives a pause request for NPT 21, it
would start playing the second range and stop at NPT 21. If the pause
request is for NPT 12 and the server is playing at NPT 13 serving the
first play request, the server stops immediately. If the pause
request is for NPT 16, the server stops after completing the first
play request and discards the second play request.
As another example, if a server has received requests to play ranges
10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
request for NPT=14 would take effect while the server plays the first
range, with the second PLAY request effectively being ignored,
assuming the PAUSE request arrives before the server has started
playing the second, overlapping range. Regardless of when the PAUSE
request arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the
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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
ensures continuous pause/play cycling without gaps.
10.7 TEARDOWN
The TEARDOWN request stops the stream delivery for the given URI,
freeing the resources associated with it. If the URI is the
presentation URI for this presentation, any RTSP session identifier
associated with the session is no longer valid. Unless all transport
parameters are defined by the session description, a SETUP request
has to be issued before the session can be played again.
Example:
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 892
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 892
10.8 GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a
presentation or stream specified in the URI. The content of the reply
and response is left to the implementation. GET_PARAMETER with no
entity body may be used to test client or server liveness ("ping").
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 431
Content-Type: text/parameters
Session: 12345678
Content-Length: 15
packets_received
jitter
C->S: RTSP/1.0 200 OK
CSeq: 431
Content-Length: 46
Content-Type: text/parameters
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packets_received: 10
jitter: 0.3838
The "text/parameters" section is only an example type for
parameter. This method is intentionally loosely defined
with the intention that the reply content and response
content will be defined after further experimentation.
10.9 SET_PARAMETER
This method requests to set the value of a parameter for a
presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client
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
changing parameter values.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
Restricting setting transport parameters to SETUP is for
the benefit of firewalls.
The parameters are split in a fine-grained fashion so that
there can be more meaningful error indications. However, it
may make sense to allow the setting of several parameters
if an atomic setting is desirable. Imagine device control
where the client does not want the camera to pan unless it
can also tilt to the right angle at the same time.
Example:
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 421
Content-length: 20
Content-type: text/parameters
barparam: barstuff
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S->C: RTSP/1.0 451 Parameter Not Understood
CSeq: 421
Content-length: 10
Content-type: text/parameters
barparam
The "text/parameters" section is only an example type for
parameter. This method is intentionally loosely defined
with the intention that the reply content and response
content will be defined after further experimentation.
10.10 REDIRECT
A redirect request informs the client that it must connect to another
server location. It contains the mandatory header Location, which
indicates that the client should issue requests for that URL. It may
contain the parameter Range, which indicates when the redirection
takes effect. If the client wants to continue to send or receive
media for this URI, the client MUST issue a TEARDOWN request for the
current session and a SETUP for the new session at the designated
host.
This example request redirects traffic for this URI to the new server
at the given play time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 732
Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z-
10.11 RECORD
This method initiates recording a range of media data according to
the presentation description. The timestamp reflects start and end
time (UTC). If no time range is given, use the start or end time
provided in the presentation description. If the session has already
started, commence recording immediately.
The server decides whether to store the recorded data under the
request-URI or another URI. If the server does not use the request-
URI, the response SHOULD be 201 (Created) and contain an entity which
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describes the status of the request and refers to the new resource,
and a Location header.
A media server supporting recording of live presentations MUST
support the clock range format; the smpte format does not make sense.
In this example, the media server was previously invited to the
conference indicated.
C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
CSeq: 954
Session: 12345678
Conference: 128.16.64.19/32492374
10.12 Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server
to interleave RTSP methods and stream data. This interleaving should
generally be avoided unless necessary since it complicates client and
server operation and imposes additional overhead. Interleaved binary
data SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier,
followed by the length of the encapsulated binary data as a binary,
two-byte integer in network byte order. The stream data follows
immediately afterwards, without a CRLF, but including the upper-layer
protocol headers. Each $ block contains exactly one upper-layer
protocol data unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header with the
interleaved parameter(Section 12.40).
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 12.40).
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
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configuration and transfer them onto UDP when possible.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
CSeq: 2
Transport: RTP/AVP/TCP;interleaved=0-1
S->C: RTSP/1.0 200 OK
CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;interleaved=0-1
Session: 12345678
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
CSeq: 3
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 3
Session: 12345678
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: $001{2 byte length}{"length" bytes RTCP packet}
11 Status Code Definitions
Where applicable, HTTP status [H10] codes are reused. Status codes
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.
11.1 Success 2xx
11.1.1 250 Low on Storage Space
The server returns this warning after receiving a RECORD request
that it may not be able to fulfill completely due to insufficient
storage space. If possible, the server should use the Range header to
indicate what time period it may still be able to record. Since other
processes on the server may be consuming storage space
simultaneously, a client should take this only as an estimate.
11.2 Redirection 3xx
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See [H10.3].
Within RTSP, redirection may be used for load balancing or
redirecting stream requests to a server topologically closer to the
client. Mechanisms to determine topological proximity are beyond the
scope of this specification.
11.3 Client Error 4xx
11.4 400 Bad Request
The request could not be understood by the server due to malformed
syntax. The client SHOULD NOT repeat the request without
modifications [H10.4.1]. If the request does not have a CSeq header,
the server MUST not include a CSeq in the response.
11.4.1 405 Method Not Allowed
The method specified in the request is not allowed for the resource
identified by the request URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource.
This status code is also to be used if a request attempts to use a
method not indicated during SETUP, e.g., if a RECORD request is
issued even though the mode parameter in the Transport header only
specified PLAY.
11.4.2 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
contained in the request.
11.4.3 452 Conference Not Found
The conference indicated by a Conference header field is unknown to
the media server.
11.4.4 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.
11.4.5 454 Session Not Found
The RTSP session identifier in the Session header is missing,
invalid, or has timed out.
11.4.6 455 Method Not Valid in This State
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The client or server cannot process this request in its current
state. The response SHOULD contain an Allow header to make error
recovery easier.
11.4.7 456 Header Field Not Valid for Resource
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 seeking.
11.4.8 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
11.4.9 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not
modified.
11.4.10 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URL in question since
it is an aggregate (presentation) URL. The method may be applied on a
stream URL.
11.4.11 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URL in question since
it is not an aggregate (presentation) URL. The method may be applied
on the presentation URL.
11.4.12 461 Unsupported Transport
The Transport field did not contain a supported transport
specification.
11.4.13 462 Destination Unreachable
The data transmission channel could not be established because the
client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid Destination
parameter in the Transport field.
11.5 Server Error 5xx
11.5.1 551 Option not supported
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An option given in the Require or the Proxy-Require fields was not
supported. The Unsupported header should be returned stating the
option for which there is no support.
12 Header Field Definitions
HTTP/1.1 [2] or other, non-standard header fields not listed here
currently have no well-defined meaning and SHOULD be ignored by the
recipient.
Table 3 summarizes the header fields used by RTSP. Type "g"
designates general request headers to be found in both requests and
responses, type "R" designates request headers, type "r" designates
response headers, and type "e" designates entity header fields.
Fields marked with "req." in the column labeled "support" MUST be
implemented by the recipient for a particular method, while fields
marked "opt." are optional. Note that not all fields marked "req."
will be sent in every request of this type. The "req." means only
that client (for response headers) and server (for request headers)
MUST implement the fields. The last column lists the method for which
this header field is meaningful; the designation "entity" refers to
all methods that return a message body. Within this specification,
DESCRIBE and GET_PARAMETER fall into this class.
12.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
The "level" parameter for presentation descriptions is
properly defined as part of the MIME type registration, not
here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl, application/sdp;level=2
12.2 Accept-Encoding
See [H14.3]
12.3 Accept-Language
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See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media
content.
12.4 Accept-Ranges
12.5 Allow
The Allow entity-header field lists the methods supported by the
resource identified by the request-URI. The purpose of this field is
to strictly inform the recipient of valid methods associated with the
resource. An Allow header field must be present in a 405 (Method Not
Allowed) response.
Example of use:
Allow: SETUP, PLAY, RECORD, SET_PARAMETER
12.6 Authorization
See [H14.8]
12.7 Bandwidth
The Bandwidth request-header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured
in bits per second. The bandwidth available to the client may change
during an RTSP session, e.g., due to modem retraining.
Bandwidth _ "Bandwidth" ":" 1*DIGIT
Example:
Bandwidth: 4000
12.8 Blocksize
The Blocksize request-header field is sent from the client to the
media server asking the server for a particular media packet size.
This packet size does not include lower-layer headers such as IP,
UDP, or RTP. The server is free to use a blocksize which is lower
than the one requested. The server MAY truncate this packet size to
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Header type support methods
____________________________________________________________
Accept R opt. entity
Accept-Encoding R opt. entity
Accept-Language R opt. all
Accept-Ranges R opt. all
Allow e opt. all
Authorization R opt. all
Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control g opt. SETUP
Conference R opt. SETUP
Connection g req. all
Content-Base e opt. entity
Content-Encoding e req. SET_PARAMETER
Content-Encoding e req. DESCRIBE, ANNOUNCE
Content-Language e req. DESCRIBE, ANNOUNCE
Content-Length e req. SET_PARAMETER, ANNOUNCE
Content-Length e req. entity
Content-Location e opt. entity
Content-Type e req. SET_PARAMETER, ANNOUNCE
CSeq g req. all
Date g opt. all
Expires e opt. DESCRIBE, ANNOUNCE
From R opt. all
If-Match R opt. SETUP
If-Modified-Since R opt. DESCRIBE, SETUP
Last-Modified e opt. entity
Location r opt. 201, 30x
Proxy-Authenticate r req. 407
Proxy-Require R req. all
Public r opt. all
Range R opt. PLAY, PAUSE, RECORD
Range r opt. PLAY, PAUSE, RECORD
Referer R opt. all
Require R req. all
Retry-After r opt. all
RTP-Info r req. PLAY
Scale g opt. PLAY, RECORD
Session g req. all but OPTIONS
Server r opt. all
Speed g opt. PLAY
Transport g req. SETUP
Unsupported r req. all
User-Agent R opt. all
Vary r opt. all
Via g opt. all
WWW-Authenticate r opt. all
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Table 3: Overview of RTSP header fields
the closest multiple of the minimum, media-specific block size, or
override it with the media-specific size if necessary. The block size
MUST be a positive decimal number, measured in octets. The server
only returns an error (416) if the value is syntactically invalid.
Blocksize _ "Blocksize" ":" 1*DIGIT
12.9 Cache-Control
The Cache-Control general-header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the
request/response chain.
Cache directives must be passed through by a proxy or gateway
application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a cache-
directive for a specific cache.
Cache-Control should only be specified in a SETUP request and its
response. Note: Cache-Control does not govern the caching of
responses as for HTTP, but rather of the stream identified by the
SETUP request. Responses to RTSP requests are not cacheable, except
for responses to DESCRIBE.
Cache-Control = "Cache-Control" ":" 1#cache-directive
cache-directive = cache-request-directive
| cache-response-directive
cache-request-directive = "no-cache"
| "max-stale"
| "min-fresh"
| "only-if-cached"
| cache-extension
cache-response-directive = "public"
| "private"
| "no-cache"
| "no-transform"
| "must-revalidate"
| "proxy-revalidate"
| "max-age" "=" delta-seconds
| cache-extension
cache-extension = token [ "=" ( token | quoted-string ) ]
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no-cache: Indicates that the media stream MUST NOT be cached
anywhere. This allows an origin server to prevent caching
even by caches that have been configured to return stale
responses to client requests.
public: Indicates that the media stream is cacheable by any
cache.
private: Indicates that the media stream is intended for a
single user and MUST NOT be cached by a shared cache. A
private (non-shared) cache may cache the media stream.
no-transform: An intermediate cache (proxy) may find it useful
to convert the media type of a certain stream. A proxy
might, for example, convert between video formats to save
cache space or to reduce the amount of traffic on a slow
link. Serious operational problems may occur, however, when
these transformations have been applied to streams intended
for certain kinds of applications. For example,
applications for medical imaging, scientific data analysis
and those using end-to-end authentication all depend on
receiving a stream that is bit-for-bit identical to the
original entity-body. Therefore, if a response includes the
no-transform directive, an intermediate cache or proxy MUST
NOT change the encoding of the stream. Unlike HTTP, RTSP
does not provide for partial transformation at this point,
e.g., allowing translation into a different language.
only-if-cached: In some cases, such as times of extremely poor
network connectivity, a client may want a cache to return
only those media streams that it currently has stored, and
not to receive these from the origin server. To do this,
the client may include the only-if-cached directive in a
request. If it receives this directive, a cache SHOULD
either respond using a cached media stream that is
consistent with the other constraints of the request, or
respond with a 504 (Gateway Timeout) status. However, if a
group of caches is being operated as a unified system with
good internal connectivity, such a request MAY be forwarded
within that group of caches.
max-stale: Indicates that the client is willing to accept a
media stream that has exceeded its expiration time. If
max-stale is assigned a value, then the client is willing
to accept a response that has exceeded its expiration time
by no more than the specified number of seconds. If no
value is assigned to max-stale, then the client is willing
to accept a stale response of any age.
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min-fresh: Indicates that the client is willing to accept a
media stream whose freshness lifetime is no less than its
current age plus the specified time in seconds. That is,
the client wants a response that will still be fresh for at
least the specified number of seconds.
must-revalidate: When the must-revalidate directive is present
in a SETUP response received by a cache, that cache MUST
NOT use the entry after it becomes stale to respond to a
subsequent request without first revalidating it with the
origin server. That is, the cache must do an end-to-end
revalidation every time, if, based solely on the origin
server's Expires, the cached response is stale.)
12.10 Conference
The Conference request-header field establishes a logical connection
between a pre-established conference and an RTSP stream. The
conference-id must not be changed for the same RTSP session.
Conference _ "Conference" ":" conference-id
Example:
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
A response code of 452 (Conference Not Found) is returned if the
conference-id is not valid.
12.11 Connection
See [H14.10]
12.12 Content-Base
See [H14.11]
12.13 Content-Encoding
See [H14.12]
12.14 Content-Language
See [H14.13]
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12.15 Content-Length
The Content-Length general-header field contains the length of the
content of the method (i.e. after the double CRLF following the last
header). Unlike HTTP, it MUST be included in all messages that carry
content beyond the header portion of the message. If it is missing, a
default value of zero is assumed. It is interpreted according to
[H14.14].
12.16 Content-Location
See [H14.15]
12.17 Content-Type
See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and
parameter-value types.
12.18 CSeq
The CSeq general-header field specifies the sequence number for an
RTSP request-response pair. This field MUST be present in all
requests and responses. For every RTSP request containing the given
sequence number, the corresponding response will have 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).
CSeq _ "Cseq" ":" 1*DIGIT
12.19 Date
See [H14.19].
12.20 Expires
The Expires entity-header field gives a date and time after which
the description or media-stream should be considered stale. The
interpretation depends on the method:
DESCRIBE response: The Expires header indicates a date and time
after which the description should be considered stale.
A stale cache entry may not normally be returned by a cache (either a
proxy cache or an user agent cache) unless it is first validated with
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the origin server (or with an intermediate cache that has a fresh
copy of the entity). See section 13 for further discussion of the
expiration model.
The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.
The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format:
Expires _ "Expires" ":" HTTP-date
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occurred in the past
(i.e., already expired).
To mark a response as "already expired," an origin server should use
an Expires date that is equal to the Date header value. To mark a
response as "never expires," an origin server should use an Expires
date approximately one year from the time the response is sent.
RTSP/1.0 servers should not send Expires dates more than one year in
the future.
The presence of an Expires header field with a date value of some
time in the future on a media stream that otherwise would by default
be non-cacheable indicates that the media stream is cacheable, unless
indicated otherwise by a Cache-Control header field (Section 12.9).
12.21 From
See [H14.22].
12.22 Host
The Host HTTP request header field is not needed for RTSP. It should
be silently ignored if sent.
12.23 If-Match
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See [H14.25].
The If-Match request-header field is especially useful for ensuring
the integrity of the presentation description, in both the case where
it is fetched via means external to RTSP (such as HTTP), or in the
case where the server implementation is guaranteeing the integrity of
the description between the time of the DESCRIBE message and the
SETUP message.
The identifier is an opaque identifier, and thus is not specific to
any particular session description language.
12.24 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (Not Modified)
response will be returned without any message-body.
If-Modified-Since _ "If-Modified-Since" ":" HTTP-date
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
12.25 Last-Modified
The Last-Modified entity-header field indicates the date and time at
which the origin server believes the presentation description or
media stream was last modified. See [H14.29]. For the methods
DESCRIBE or ANNOUNCE, the header field indicates the last
modification date and time of the description, for SETUP that of the
media stream.
12.26 Location
See [H14.30].
12.27 Proxy-Authenticate
See [H14.33].
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12.28 Proxy-Require
The Proxy-Require request-header field is used to indicate proxy-
sensitive features that MUST be supported by the proxy. Any Proxy-
Require header features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client if not supported.
Servers should treat this field identically to the Require field.
See Section 12.33 for more details on the mechanics of this message
and a usage example.
12.29 Public
See [H14.35].
12.30 Range
The Range request and response header field specifies a range of
time. The range can be specified in a number of units. This
specification defines the smpte (Section 3.5), npt (Section 3.6),
and clock (Section 3.7) range units. Within RTSP, byte ranges
[H14.36.1] are not meaningful and MUST NOT be used. The header may
also contain a time parameter in UTC, specifying the time at which
the operation is to be made effective. Servers supporting the Range
header MUST understand the NPT range format and SHOULD understand the
SMPTE range format. The Range response header indicates what range
of time is actually being played or recorded. If the Range header is
given in a time format that is not understood, the recipient should
return 501 (Not Implemented).
Ranges are half-open intervals, including the lower point, but
excluding the upper point. In other words, a range of a-b starts
exactly at time a, but stops just before b. Only the start time of a
media unit such as a video or audio frame is relevant. As an example,
assume that video frames are generated every 40 ms. A range of
10.0-10.1 would include a video frame starting at 10.0 or later time
and would include a video frame starting at 10.08, even though it
lasted beyond the interval. A range of 10.0-10.08, on the other hand,
would exclude the frame at 10.08.
Range _ "Range" ":" 1#ranges-specifier [ ";" "time" "=" utc-time ]
ranges-specifier _ npt-range | utc-range | smpte-range
Example:
Range: clock=19960213T143205Z-;time=19970123T143720Z
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The notation is similar to that used for the HTTP/1.1 [2]
byte-range header. It allows clients to select an excerpt
from the media object, and to play from a given point to
the end as well as from the current location to a given
point. The start of playback can be scheduled for any time
in the future, although a server may refuse to keep server
resources for extended idle periods.
12.31 Referer
See [H14.37]. The URL refers to that of the presentation description,
typically retrieved via HTTP.
12.32 Retry-After
See [H14.38].
12.33 Require
The Require request-header field is used by clients to query the
server about options that it may or may not support. The server MUST
respond to this header by using the Unsupported header to negatively
acknowledge 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
Example:
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/1.0 551 Option not supported
CSeq: 302
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Unsupported: funky-feature
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 303
S->C: RTSP/1.0 200 OK
CSeq: 303
In this example, "funky-feature" is the feature tag which indicates
to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.
Proxies and other intermediary devices SHOULD ignore features that
are not understood in this field. If a particular extension requires
that intermediate devices support it, the extension should be tagged
in the Proxy-Require field instead (see Section 12.28).
12.34 RTP-Info
The RTP-Info response-header field is used to set RTP-specific
parameters in the PLAY response.
url: Indicates the stream URL which for which the following RTP
parameters correspond.
seq: Indicates the sequence number of the first packet of the
stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that
originated after the seek.
rtptime: Indicates the RTP timestamp corresponding to the time
value in the Range response header. (Note: For aggregate
control, a particular stream may not actually generate a
packet for the Range time value returned or implied. Thus,
there is no guarantee that the packet with the sequence
number indicated by seq actually has the timestamp
indicated by rtptime.) The client uses this value to
calculate the mapping of RTP time to NPT.
A mapping from RTP timestamps to NTP timestamps (wall
clock) is available via RTCP. However, this
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information is not sufficient to generate a mapping
from RTP timestamps to NPT. Furthermore, in order to
ensure that this information is available at the
necessary time (immediately at startup or after a
seek), and that it is delivered reliably, this mapping
is placed in the RTSP control channel.
In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to
NTP, using initial RTCP sender reports to do the mapping,
and later reports to check drift against the mapping.
Syntax:
RTP-Info ______________________________ "RTP-Info" ":" 1#rtsp-info-spec
rtsp-info-spec ______________________________ stream-url 1*parameter
stream-url ______________________________ quoted-url | unquoted-url
unquoted-url ______________________________ "url" "=" safe-url
| ";" "mode" = <"> 1#Method <">
quoted-url ______________________________ "url" "=" <"> needquote-url <">
safe-url ______________________________ url
needquote-url ______________________________ url
url ______________________________ ( absoluteURI | relativeURI )
parameter ______________________________ ";" "seq" "=" 1*DIGIT
| ";" "rtptime" "=" 1*DIGIT
Additional constraint: safe-url MUST NOT contain the semicolon (";")
or comma (",") characters. The quoted-url form SHOULD only be used
when a URL does not meet the safe-url constraint, in order to ensure
compatibility with implementations conformant to RFC 2326 [25].
absoluteURI and relativeURI are defined in RFC 2396 [26].
Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
12.35 Scale
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 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 indicates half the normal viewing rate. In other words, a
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ratio of 2 has normal play time increase at twice the wallclock rate.
For every second of elapsed (wallclock) time, 2 seconds of content
will be delivered. A negative value indicates reverse direction.
Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected key frames. For audio, it may time-scale
the audio while preserving pitch or, less desirably, deliver
fragments of audio.
The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response
MUST contain the actual scale value chosen by the server.
If the request contains a Range parameter, the new scale value will
take effect at that time.
Scale _ "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
12.36 Speed
The Speed request-header field requests the server to deliver data
to the client at a particular speed, contingent on the server's
ability and desire to serve the media stream at the given speed.
Implementation by the server is OPTIONAL. The default is the bit rate
of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of
2.0 indicates that data is to be delivered twice as fast as normal. A
speed of zero is invalid. If the request contains a Range parameter,
the new speed value will take effect at that time.
Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]
Example:
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Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It is
meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. Implementors
should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss
rates.
12.37 Server
See [H14.39]
12.38 Session
The Session request-header and response-header field identifies an
RTSP session started by the media server in a SETUP response and
concluded by TEARDOWN on the presentation URL. The session
identifier is chosen by the media server (see Section 3.4) and MUST
be returned in the SETUP response. Once a client receives a Session
identifier, it MUST return it for any request related to that
session.
Session _ "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
The timeout parameter is only allowed in a response header. The
server uses it to indicate to the client how long the server is
prepared to wait between RTSP commands before closing the session due
to lack of activity (see Section A). The timeout is measured in
seconds, with a default of 60 seconds (1 minute).
Note that a session identifier identifies an RTSP session across
transport sessions or connections. Control messages for more than one
RTSP URL may be sent within a single RTSP session. Hence, it is
possible that clients use the same session for controlling many
streams constituting a presentation, as long as all the streams come
from the same server. (See example in Section 14). However, multiple
"user" sessions for the same URL from the same client MUST use
different session identifiers.
The session identifier is needed to distinguish several
delivery requests for the same URL coming from the same
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client.
The response 454 (Session Not Found) is returned if the session
identifier is invalid.
12.39 Timestamp
The Timestamp general-header field describes when the client sent
the request to the server. The value of the timestamp is of
significance only to the client and may use any timescale. The server
MUST echo the exact same value and MAY, if it has accurate
information about this, add a floating point number indicating the
number of seconds that has elapsed since it has received the request.
The timestamp is used by the client to compute the round-trip time to
the server so that it can adjust the timeout value for
retransmissions.
Timestamp _ "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay _ *(DIGIT) [ "." *(DIGIT) ]
12.40 Transport
The Transport request-header field indicates which transport
protocol is to be used and configures its parameters such as
destination address, compression, multicast time-to-live and
destination port for a single stream. It sets those values not
already determined by a presentation description.
Transports are comma separated, listed in order of preference.
Parameters may be added to each transport, separated by a semicolon.
The Transport header field MAY also be used to change certain
transport parameters. A server MAY refuse to change parameters of an
existing stream.
The server MAY return a Transport response-header field in the
response to indicate the values actually chosen.
A Transport request header field may contain a list of transport
options acceptable to the client, in the form of multiple transport-
spec entries. In that case, the server MUST return a single option (
transport-spec) which was actually chosen.
A transport-spec transport option may only contain one of any given
parameter within it. Parameters may be given in any order.
Additionally, it may only contain the unicast or multicast transport
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parameter.
The Transport header field is restricted to describing a
single RTP stream. (RTSP can also control multiple streams
as a single entity.) Making it part of RTSP rather than
relying on a multitude of session description formats
greatly simplifies designs of firewalls.
The syntax for the transport specifier is
transport
/
profile
/
lower-transport
The default value for the "lower-transport" parameters is specific to
the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport:
General parameters:
unicast | multicast: This parameter is a mutually exclusive
indication of whether unicast or multicast delivery will be
attempted. One of the two values MUST be specified. Clients
that are capable of handling both unicast and multicast
transmission MUST indicate such capability by including two
full transport-specs with separate parameters for each.
destination: The address to which a stream will be sent. The
client may specify the destination address with the
destination parameter. To avoid becoming the unwitting
perpetrator of a remote-controlled denial-of-service
attack, a server SHOULD authenticate the client and SHOULD
log such attempts before allowing the client to direct a
media stream to an address not chosen by the server. This
is particularly important if RTSP commands are issued via
UDP, but implementations cannot rely on TCP as reliable
means of client identification by itself.
source: If the source address for the stream is different than
can be derived from the RTSP endpoint address (the server
in playback or the client in recording), the source address
MAY be specified.
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This information may also be available through SDP.
However, since this is more a feature of transport
than media initialization, the authoritative source
for this information should be in the SETUP response.
layers: The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses
starting at the destination address.
mode: The mode parameter indicates the methods to be supported
for this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY.
append: If the mode parameter includes RECORD, the append
parameter indicates that the media data should append to
the existing resource rather than overwrite it. If
appending is requested and the server does not support
this, it MUST refuse the request rather than overwrite the
resource identified by the URI. The append parameter is
ignored if the mode parameter does not contain RECORD.
interleaved: The interleaved parameter implies mixing the media
stream with the control stream in whatever protocol is
being used by the control stream, using the mechanism
defined in Section 10.12. The argument provides the channel
number to be used in the $ 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:
ttl: multicast time-to-live.
RTP-specific:
port: This parameter provides the RTP/RTCP port pair for a
multicast session. It is specified as a range, e.g.,
port=3456-3457
client_port: This parameter provides the unicast RTP/RTCP port
pair on the client where media data and control information
is to be sent. It is specified as a range, e.g.,
port=3456-3457
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server_port: This parameter provides the unicast RTP/RTCP port
pair on the server where media data and control information
is to be sent. It is specified as a range, e.g.,
port=3456-3457
ssrc: The ssrc parameter indicates the RTP SSRC [27] value that
should be (request) or will be (response) used by the media
server. This parameter is only valid for unicast
transmission. It identifies the synchronization source to
be associated with the media stream, and is expressed as an
eight digit hexidecimal value.
Transport ______________________________________________ "Transport" ":" 1#transport-spec
transport-spec = transport-id *parameter
transport-id = transport-protocol "/" profile ["/" lower-transport]
; no LWS is allowed inside transport-id
transport-protocol = "RTP" | token
profile = "AVP" | token
lower-transport = "TCP" | "UDP" | token
parameter = ";" ( "unicast" | "multicast" )
| ";" "source" [ "=" address ]
| ";" "destination" [ "=" address ]
| ";" "interleaved" "=" channel [ "-" channel ]
| ";" "append"
| ";" "ttl" "=" ttl
| ";" "layers" "=" 1*DIGIT
| ";" "port" "=" port [ "-" port ]
| ";" "client_port" "=" port [ "-" port ]
| ";" "server_port" "=" port [ "-" port ]
| ";" "source" "=" address
| ";" "ssrc" "=" ssrc
| ";" "mode" "=" mode-spec
ttl = 1*3(DIGIT)
port = 1*5(DIGIT)
ssrc = 8*8(HEX)
channel = 1*3(DIGIT)
address = host
mode-spec = <"> 1#mode <"> | mode
mode = "PLAY" | "RECORD" | token
Below is a usage example, showing a client advertising the capability
to handle multicast or unicast, preferring multicast. Since this is a
unicast-only stream, the server responds with the proper transport
parameters for unicast.
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C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Transport: RTP/AVP;multicast;mode="PLAY",
RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"
S->C: RTSP/1.0 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;client_port=3456-3457;
server_port=6256-6257;mode="PLAY"
12.41 Unsupported
The Unsupported response-header field lists the features not
supported by the server. In the case where the feature was specified
via the Proxy-Require field (Section 12.33), if there is a proxy on
the path between the client and the server, the proxy MUST insert a
response message with a status code of 551 (Option Not Supported).
See Section 12.33 for a usage example.
Unsupported _ "Unsupported" ":" 1#option-tag
12.42 User-Agent
See [H14.42]
12.43 Vary
See [H14.43]
12.44 Via
See [H14.44].
12.45 WWW-Authenticate
See [H14.46].
13 Caching
In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
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exception of the presentation description returned by DESCRIBE or
included with ANNOUNCE. (Since the responses for anything but
DESCRIBE and GET_PARAMETER do not return any data, caching is not
really an issue for these requests.) However, it is desirable for the
continuous media data, typically delivered out-of-band with respect
to RTSP, to be cached, as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up-to-date by
issuing a SETUP or DESCRIBE request, respectively, and comparing
the Last-Modified header with that of the cached copy. If the copy
is not up-to-date, it modifies the SETUP transport parameters as
appropriate and forwards the request to the origin server. Subsequent
control commands such as PLAY or PAUSE then pass the proxy
unmodified. The proxy delivers the continuous media data to the
client, while possibly making a local copy for later reuse. The exact
behavior allowed to the cache is given by the cache-response
directives described in Section 12.9. A cache MUST answer any
DESCRIBE requests if it is currently serving the stream to the
requestor, as it is possible that low-level details of the stream
description may have changed on the origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
through" variety. Rather than retrieving the whole resource from the
origin server, the cache simply copies the streaming data as it
passes by on its way to the client. Thus, it does not introduce
additional latency.
To the client, an RTSP proxy cache appears like a regular media
server, to the media origin server like a client. Just as an HTTP
cache has to store the content type, content language, and so on for
the objects it caches, a media cache has to store the presentation
description. Typically, a cache eliminates all transport-references
(that is, multicast information) from the presentation description,
since these are independent of the data delivery from the cache to
the client. Information on the encodings remains the same. If the
cache is able to translate the cached media data, it would create a
new presentation description with all the encoding possibilities it
can offer.
14 Examples
The following examples refer to stream description formats that are
not standards, such as RTSL. The following examples are not to be
used as a reference for those formats.
14.1 Media on Demand (Unicast)
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Client C requests a movie from media servers A ( audio.example.com )
and V ( video.example.com ). The media description is stored on a web
server W. The media description contains descriptions of the
presentation and all its streams, including the codecs that are
available, dynamic RTP payload types, the protocol stack, and content
information such as language or copyright restrictions. It may also
give an indication about the timeline of the movie.
In this example, the client is only interested in the last part of
the movie.
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.0 200 OK
Content-Type: application/sdp
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
A->C: RTSP/1.0 200 OK
CSeq: 1
Session: 12345678
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
server_port=5000-5001
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
V->C: RTSP/1.0 200 OK
CSeq: 1
Session: 23456789
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
server_port=5002-5003
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
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CSeq: 2
Session: 23456789
Range: smpte=0:10:00-
V->C: RTSP/1.0 200 OK
CSeq: 2
Session: 23456789
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
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/1.0 200 OK
CSeq: 2
Session: 12345678
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
CSeq: 3
Session: 12345678
A->C: RTSP/1.0 200 OK
CSeq: 3
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 3
Session: 23456789
V->C: RTSP/1.0 200 OK
CSeq: 3
Even though the audio and video track are on two different servers,
and may start at slightly different times and may drift with respect
to each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender
reports.
14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in
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which multiple continuous media types pertaining to the same end-user
presentation are present. In effect, the container file represents an
RTSP presentation, with each of its components being RTSP streams.
Container files are a widely used means to store such presentations.
While the components are transported as independent streams, it is
desirable to maintain a common context for those streams at the
server end.
This enables the server to keep a single storage handle
open easily. It also allows treating all the streams
equally in case of any prioritization of streams by the
server.
It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by the client in order to preserve
the artistic effect of the combined media presentation. Similarly, in
such a tightly bound presentation, it is desirable to be able to
control all the streams via a single control message using an
aggregate URL.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URL to
the container file.
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 164
v=0
o=- 2890844256 2890842807 IN IP4 172.16.2.93
s=RTSP Session
i=An Example of RTSP Session Usage
a=control:rtsp://foo/twister
t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://foo/twister/audio
m=video 0 RTP/AVP 26
a=control:rtsp://foo/twister/video
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C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001;
server_port=9000-9001
Session: 12345678
C->M: SETUP rtsp://foo/twister/video RTSP/1.0
CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003;
server_port=9004-9005
Session: 12345678
C->M: PLAY rtsp://foo/twister RTSP/1.0
CSeq: 4
Range: npt=0-
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 4
Session: 12345678
RTP-Info: url=rtsp://foo/twister/video;
seq=9810092;rtptime=3450012
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
CSeq: 5
Session: 12345678
M->C: RTSP/1.0 460 Only aggregate operation allowed
CSeq: 5
C->M: PAUSE rtsp://foo/twister RTSP/1.0
CSeq: 6
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 6
Session: 12345678
C->M: SETUP rtsp://foo/twister RTSP/1.0
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CSeq: 7
Transport: RTP/AVP;unicast;client_port=10000
M->C: RTSP/1.0 459 Aggregate operation not allowed
CSeq: 7
In the first instance of failure, the client tries to pause one
stream (in this case video) of the presentation. This is disallowed
for that presentation by the server. In the second instance, the
aggregate URL may not be used for SETUP and one control message is
required per stream to set up transport parameters.
This keeps the syntax of the Transport header simple and
allows easy parsing of transport information by firewalls.
14.3 Single Stream Container Files
Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients SHOULD use the rules set forth in the session
description for request URLs, rather than assuming that a consistent
URL may always be used throughout. Here's an example of how a multi-
stream server might expect a single-stream file to be served:
C->S DESCRIBE rtsp://foo.com/test.wav RTSP/1.0
Accept: application/x-rtsp-mh, application/sdp
CSeq: 1
S->C RTSP/1.0 200 OK
CSeq: 1
Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp
Content-length: 48
v=0
o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file
i=audio test
t=0 0
m=audio 0 RTP/AVP 0
a=control:streamid=0
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast;
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client_port=6970-6971;mode="PLAY"
CSeq: 2
S->C RTSP/1.0 200 OK
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
server_port=6970-6971;mode="PLAY"
CSeq: 2
Session: 2034820394
C->S PLAY rtsp://foo.com/test.wav RTSP/1.0
CSeq: 3
Session: 2034820394
S->C RTSP/1.0 200 OK
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 to the aggregate URL in the PLAY command. This makes complete
sense when there are multiple streams with aggregate control, but is
less than intuitive in the special case where the number of streams
is one.
In this special case, it is recommended that servers be forgiving of
implementations that send:
C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
CSeq: 3
In the worst case, servers should send back:
S->C RTSP/1.0 460 Only aggregate operation allowed
CSeq: 3
One would also hope that server implementations are also forgiving of
the following:
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C->S SETUP rtsp://foo.com/test.wav RTSP/1.0
Transport: rtp/avp/udp;client_port=6970-6971;mode="PLAY"
CSeq: 2
Since there is only a single stream in this file, it's not ambiguous
what this means.
14.4 Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, we
assume that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
C->W: GET /concert.sdp HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: application/x-rtsl
<session>
<track src="rtsp://live.example.com/concert/audio">
</session>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 44
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 3456 RTP/AVP 0
c=IN IP4 224.2.0.1/16
a=control:rtsp://live.example.com/concert/audio
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 2
Transport: RTP/AVP;multicast
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=224.2.0.1;
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port=3456-3457;ttl=16
Session: 0456804596
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 3
Session: 0456804596
M->C: RTSP/1.0 200 OK
CSeq: 3
Session: 0456804596
14.5 Playing media into an existing session
A conference participant C wants to have the media server M play back
a demo tape into an existing conference. C indicates to the media
server that the network addresses and encryption keys are already
given by the conference, so they should not be chosen by the server.
The example omits the simple ACK responses.
C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 1
Accept: application/sdp
M->C: RTSP/1.0 200 1 OK
Content-type: application/sdp
Content-Length: 44
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
i=See above
t=0 0
m=audio 0 RTP/AVP 0
C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
port=7000-7001;ttl=127
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
port=7000-7001;ttl=127
Session: 91389234234
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Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 3
Session: 91389234234
M->C: RTSP/1.0 200 OK
CSeq: 3
14.6 Recording
The conference participant client C asks the media server M to record
the audio and video portions of a meeting. The client uses the
ANNOUNCE method to provide meta-information about the recorded
session to the server.
C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
CSeq: 90
Content-Type: application/sdp
Content-Length: 121
v=0
o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
s=IETF Meeting, Munich - 1
i=The thirty-ninth IETF meeting will be held in Munich, Germany
u=http://www.ietf.org/meetings/Munich.html
e=IETF Channel 1 <ietf39-mbone@uni-koeln.de>
p=IETF Channel 1 +49-172-2312 451
c=IN IP4 224.0.1.11/127
t=3080271600 3080703600
a=tool:sdr v2.4a6
a=type:test
m=audio 21010 RTP/AVP 5
c=IN IP4 224.0.1.11/127
a=ptime:40
m=video 61010 RTP/AVP 31
c=IN IP4 224.0.1.12/127
M->C: RTSP/1.0 200 OK
CSeq: 90
C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
CSeq: 91
Transport: RTP/AVP;multicast;destination=224.0.1.11;
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port=21010-21011;mode=record;ttl=127
M->C: RTSP/1.0 200 OK
CSeq: 91
Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.11;
port=21010-21011;mode=record;ttl=127
C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
CSeq: 92
Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127
M->C: RTSP/1.0 200 OK
CSeq: 92
Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
CSeq: 93
Session: 50887676
Range: clock=19961110T1925-19961110T2015
M->C: RTSP/1.0 200 OK
CSeq: 93
15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 [2].
15.1 Base Syntax
OCTET ____________________________ <any 8-bit sequence of data>
CHAR ____________________________ <any US-ASCII character (octets 0 - 127)>
UPALPHA ____________________________ <any US-ASCII uppercase letter "A".."Z">
LOALPHA ____________________________ <any US-ASCII lowercase letter "a".."z">
ALPHA ____________________________ UPALPHA | LOALPHA
DIGIT ____________________________ <any US-ASCII digit "0".."9">
CTL ____________________________ <any US-ASCII control character
(octets 0 - 31) and DEL (127)>
CR ____________________________ <US-ASCII CR, carriage return (13)>
LF ____________________________ <US-ASCII LF, linefeed (10)>
SP ____________________________ <US-ASCII SP, space (32)>
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HT ____________________________ <US-ASCII HT, horizontal-tab (9)>
<"> ____________________________ <US-ASCII double-quote mark (34)>
CRLF ____________________________ CR LF
LWS ____________________________ [CRLF] 1*( SP | HT )
TEXT ____________________________ <any OCTET except CTLs>
tspecials ____________________________ "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | "
\&\h'|\n(40u'\h'|\n(41u'\h'|\n(42u'
" | <">
| "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT
token ____________________________ 1*<any CHAR except CTLs or tspecials>
quoted-string ____________________________ ( <"> *(qdtext) <"> )
qdtext ____________________________ <any TEXT except <">>
quoted-pair ____________________________ "
\&\h'|\n(40u'\h'|\n(41u'\h'|\n(42u'
" CHAR
message-header ____________________________ field-name ":" [ field-value ] CRLF
field-name ____________________________ token
field-value ____________________________ *( field-content | LWS )
field-content ____________________________ <the OCTETs making up the field-value and
consisting
of either *TEXT or combinations of token, tspecials,
and 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
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [H15]
apply. Specifically, please note the following:
Authentication Mechanisms: RTSP and HTTP share common
authentication schemes, and thus should follow the same
prescriptions with regards to authentication. See [H15.1]
for client authentication issues, and [H15.2] for issues
regarding support for multiple authentication mechanisms.
Abuse of Server Log Information: RTSP and HTTP servers will
presumably have similar logging mechanisms, and thus should
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be equally guarded in protecting the contents of those
logs, thus protecting the privacy of the users of the
servers. See [H15.3] for HTTP server recommendations
regarding server logs.
Transfer of Sensitive Information: There is no reason to believe
that information transferred via 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.
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
header fields 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
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(RFC 2068 [2], 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 attack: 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 (preferably
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
[8] authentication. In environments requiring tighter
security for the control messages, transport layer
mechanisms such as TLS (RFC 2246 [7]) 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 security considerations brought up
in those and other applicable specifications.
Persistently suspicious behavior: RTSP servers SHOULD return
error code 403 (Forbidden) upon receiving a single instance
of behavior which is deemed a security risk. RTSP servers
SHOULD also be aware of attempts to probe the server for
weaknesses and entry points and MAY arbitrarily disconnect
and ignore further requests clients which are deemed to be
in violation of local security policy.
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A RTSP Protocol State Machines
The RTSP client and server state machines describe the behavior of
the protocol from RTSP session initialization through RTSP session
termination.
State is defined on a per object basis. An object is uniquely
identified by the stream URL and the RTSP session identifier. Any
request/reply using aggregate URLs denoting RTSP presentations
composed of multiple streams will have an effect on the individual
states of all the streams. For example, if the presentation /movie
contains two streams, /movie/audio and /movie/video , then the
following command:
PLAY rtsp://foo.com/movie RTSP/1.0
CSeq: 559
Session: 12345678
will have an effect on the states of movie/audio and movie/video
This example does not imply a standard way to represent
streams in URLs or a relation to the filesystem. See
Section 3.2.
The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER,
SET_PARAMETER do not have any effect on client or server state and
are therefore not listed in the state tables.
A.1 Client State Machine
The client can assume the following states:
Init : SETUP has been sent, waiting for reply.
Ready : SETUP reply received or PAUSE reply received while in
Playing state.
Playing : PLAY reply received
Recording : RECORD reply received
In general, the client changes state on receipt of replies to
requests. Note that some requests are effective at a future time or
position (such as a PAUSE), and state also changes accordingly. If
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no explicit SETUP is required for the object (for example, it is
available via a multicast group), state begins at Ready are only two
states, Ready and Playing The client also changes state from
Playing/Recording to Ready when the end of the requested range is
reached.
The "next state" column indicates the state assumed after receiving a
success response (2xx). If a request yields a status code of 3xx, the
state becomes Init , and a status code of 4xx yields no change in
state. Messages not listed for each state MUST NOT be issued by the
client in that state, with the exception of messages not affecting
state, as listed above. Receiving a REDIRECT from the server is
equivalent to receiving a 3xx redirect status from the server.
state message sent next state after response
____________________________________________________________
Init SETUP
Ready
TEARDOWN
Init
Ready PLAY
Playing
RECORD
Recording
TEARDOWN
Init
SETUP
Ready
Playing PAUSE
Ready
TEARDOWN
Init
PLAY
Playing
SETUP
Playing
(changed transport)
Recording PAUSE
Ready
TEARDOWN
Init
RECORD
Recording
SETUP
Recording
(changed transport)
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A.2 Server State Machine
The server can assume the following states:
Init : The initial state, no valid SETUP has been received yet.
Ready : Last SETUP received was successful, reply sent or after
playing, last PAUSE received was successful, reply sent.
Playing : Last PLAY received was successful, reply sent. Data
is being sent.
Recording : The server is recording media data.
In general, the server changes state on receiving requests. If the
server is in state Playing or Recording and in unicast mode, it MAY
revert to Init and tear down the RTSP session if it has not received
"wellness" information, such as RTCP reports or RTSP commands, from
the client for a defined interval, with a default of one minute. The
server can declare another timeout value in the Session response
header (Section 12.38). If the server is in state Ready , it MAY
revert to Init if it does not receive an RTSP request for an interval
of more than one minute. Note that some requests (such as PAUSE) may
be effective at a future time or position, and server state changes
at the appropriate time. The server reverts from state Playing or
Recording to state Ready at the end of the range requested by the
client.
The REDIRECT message, when sent, is effective immediately unless it
has a Range header specifying when the redirect is effective. In
such a case, server state will also change at the appropriate time.
If no explicit SETUP is required for the object, the state starts at
Ready and there are only two states, Ready and Playing
The "next state" column indicates the state assumed after sending a
success response (2xx). If a request results in a status code of 3xx,
the state becomes Init change.
state message received next state
_______________________________________
Init
SETUP
Ready
TEARDOWN
Init
Ready
PLAY
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Playing
SETUP
Ready
TEARDOWN
Init
RECORD
Recording
Playing
PLAY
Playing
PAUSE
Ready
TEARDOWN
Init
SETUP
Playing
Recording
RECORD
Recording
PAUSE
Ready
TEARDOWN
Init
SETUP
Recording
B Interaction with RTP
RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP
media layer[27]. The media layer rendering the RTP stream should not
be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
timestamps MUST be continuous and monotonic across jumps of NPT.
As an example, assume a clock frequency of 8000 Hz, a packetization
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 through 20. The first segment is presented as RTP packets with
sequence numbers 0 through 49 and timestamp 0 through 39,200. The
second segment consists of RTP packets with sequence number 50
through 69, with timestamps 40,000 through 55,200.
We cannot assume that the RTSP client can communicate with
the RTP media agent, as the two may be independent
processes. If the RTP timestamp shows the same gap as the
NPT, the media agent will assume that there is a pause in
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the presentation. If the jump in NPT is large enough, the
RTP timestamp may roll over and the media agent may believe
later packets to be duplicates of packets just played out.
For certain datatypes, tight integration between the RTSP layer and
the RTP layer will be necessary. This by no means precludes the above
restriction. Combined RTSP/RTP media clients should use the RTP-Info
field to determine whether incoming RTP packets were sent before or
after a seek.
For continuous audio, the server SHOULD set the RTP marker bit at the
beginning of serving a new PLAY request. This allows the client to
perform playout delay adaptation.
For scaling (see Section 12.35), RTP timestamps should correspond to
the playback timing. For example, when playing video recorded at 30
frames/second at a scale of two and speed (Section 12.36) of one, the
server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame.
The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning. The
sequence parameter of the RTP-Info (Section 12.34) header provides
the first sequence number of the next segment.
C Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
describe streams or presentations in RTSP. Such usage is limited to
specifying means of access and encoding(s) for:
aggregate control: A presentation composed of streams from one
or more servers that are available for aggregate control.
Such a description is typically retrieved by HTTP or other
non-RTSP means. However, they may be received with
ANNOUNCE methods.
non-aggregate control: A presentation composed of multiple
streams from a single server that are not available for
aggregate control. Such a description is typically
returned in reply to a DESCRIBE request on a URL, or
received in an ANNOUNCE method.
This appendix describes how an SDP file, retrieved, for example,
through HTTP, determines the operation of an RTSP session. It also
describes how a client should interpret SDP content returned in reply
to a DESCRIBE request. SDP provides no mechanism by which a client
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can distinguish, without human guidance, between several media
streams to be rendered simultaneously and a set of alternatives
(e.g., two audio streams spoken in different languages).
C.1 Definitions
The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in
SDP (RFC 2327 [6]):
C.1.1 Control URL
The "a=control:" attribute is used to convey the control URL. This
attribute is used both for the session and media descriptions. If
used for individual media, it indicates the URL to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URL for aggregate control.
Example:
a=control:rtsp://example.com/foo
This attribute may contain either relative and absolute URLs,
following the rules and conventions set out in RFC 1808 [28].
Implementations should look for a base URL in the following order:
1. the RTSP Content-Base field;
2. the RTSP Content-Location field;
3. the RTSP request URL.
If this attribute contains only an asterisk (*), then the URL is
treated as if it were an empty embedded URL, and thus inherits the
entire base URL.
C.1.2 Media Streams
The "m=" field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate
synchronization. If the session is unicast, the port number serves as
a recommendation from the server to the client; the client still has
to include it in its SETUP request and may ignore this
recommendation. If the server has no preference, it SHOULD set the
port number value to zero.
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Example:
m=audio 0 RTP/AVP 31
C.1.3 Payload Type(s)
The payload type(s) are specified in the "m=" field. In case the
payload type is a static payload type from RFC 1890 [1], no other
information is required. In case it is a dynamic payload type, the
media attribute "rtpmap" is used to specify what the media is. The
"encoding name" within the "rtpmap" attribute may be one of those
specified in RFC 1890 (Sections 5 and 6), or an experimental encoding
with a "X-" prefix as specified in SDP (RFC 2327 [6]). Codec-specific
parameters are not specified in this field, but rather in the "fmtp"
attribute described below. Implementors seeking to register new
encodings should follow the procedure in RFC 1890 [1]. If the media
type is not suited to the RTP AV profile, then it is recommended that
a new profile be created and the appropriate profile name be used in
lieu of "RTP/AVP" in the "m=" field.
C.1.4 Format-Specific Parameters
Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) that the attribute refers to. Note that the packetization
interval is conveyed using the "ptime" attribute.
C.1.5 Range of Presentation
The "a=range" attribute defines the total time range of the stored
session. (The length of live sessions can be deduced from the "t" and
"r" parameters.) Unless the presentation contains media streams of
different durations, the length attribute is a session-level
attribute. The unit is specified first, followed by the value range.
The units and their values are as defined in Section 3.5, 3.6 and
3.7.
Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203
C.1.6 Time of Availability
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The "t=" field MUST contain suitable values for the start and stop
times for both aggregate and non-aggregate stream control. With
aggregate control, the server SHOULD indicate a stop time value for
which it guarantees the description to be valid, and a start time
that is equal to or before the time at which the DESCRIBE request was
received. It MAY also indicate start and stop times of 0, meaning
that the session is always available. With non-aggregate control, the
values should reflect the actual period for which the session is
available in keeping with SDP semantics, and not depend on other
means (such as the life of the web page containing the description)
for this purpose.
C.1.7 Connection Information
In SDP, the "c=" field contains the destination address for the media
stream. However, for on-demand unicast streams and some multicast
streams, the destination address is specified by the client via the
SETUP request. Unless the media content has a fixed destination
address, the "c=" field is to be set to a suitable null value. For
addresses of type "IP4", this value is "0.0.0.0".
C.1.8 Entity Tag
The optional "a=etag" attribute identifies a version of the session
description. It is opaque to the client. SETUP requests may include
this identifier in the If-Match field (see section 12.23) to only
allow session establishment if this attribute value still corresponds
to that of the current description. The attribute value is opaque
and may contain any character allowed within SDP attribute values.
Example:
a=etag:158bb3e7c7fd62ce67f12b533f06b83a
One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would
put constraints on servers that need to support multiple
session description types other than SDP for the same piece
of media content.
C.2 Aggregate Control Not Available
If a presentation does not support aggregate control and multiple
media sections are specified, each section MUST have the control URL
specified via the "a=control:" attribute.
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Example:
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I came from a web page
c=IN IP4 0.0.0.0
t=0 0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid
Note that the position of the control URL in the description implies
that the client establishes separate RTSP control sessions to the
servers audio.com and video.com
It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media
client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.
C.3 Aggregate Control Available
In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control:" attributes, which are used to specify the stream URLs,
and a session-level "a=control:" attribute which is used as the
request URL for aggregate control. If the media-level URL is
relative, it is resolved to absolute URLs according to Section C.1.1
above.
If the presentation comprises only a single stream, the media-level
"a=control:" attribute may be omitted altogether. However, if the
presentation contains more than one stream, each media stream section
MUST contain its own "a=control" attribute.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I contain
i=<more info>
c=IN IP4 0.0.0.0
t=0 0
a=control:rtsp://example.com/movie/
H. Schulzrinne et. al. [Page 85]
Internet Draft RTSP February 22, 2002
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
In this example, the client is required to establish a single RTSP
session to the server, and uses the URLs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to set up the video and audio
streams, respectively. The URL rtsp://example.com/movie/ controls the
whole movie.
A client is not required to issues SETUP requests for all streams
within an aggregate object. Servers SHOULD allow the client to ask
for only a subset of the streams.
D Minimal RTSP implementation
D.1 Client
A client implementation MUST be able to do the following :
o Generate the following requests: SETUP, TEARDOWN, and one of
PLAY (i.e., a minimal playback client) or RECORD (i.e., a
minimal recording client). If RECORD is implemented, ANNOUNCE
MUST be implemented as well.
o Include the following headers in requests: CSeq, Connection,
Session, Transport. If ANNOUNCE is implemented, the
capability to include headers Content-Language, Content-
Encoding, Content-Length, and Content-Type should be as well.
o Parse and understand the following headers in responses:
CSeq, Connection, Session, Transport, Content-Language,
Content-Encoding, Content-Length, Content-Type. If RECORD is
implemented, the Location header must be understood as well.
RTP-compliant implementations should also implement RTP-Info.
o Understand the class of each error code received and notify
the end-user, if one is present, of error codes in classes 4xx
and 5xx. The notification requirement may be relaxed if the
end-user explicitly does not want it for one or all status
codes.
o Expect and respond to asynchronous requests from the server,
such as ANNOUNCE. This does not necessarily mean that it
H. Schulzrinne et. al. [Page 86]
Internet Draft RTSP February 22, 2002
should implement the ANNOUNCE method, merely that it MUST
respond positively or negatively to any request received from
the server.
Though not required, the following are RECOMMENDED.
o Implement RTP/AVP/UDP as a valid transport.
o Inclusion of the User-Agent header.
o Understand SDP session descriptions as defined in Appendix C
o Accept media initialization formats (such as SDP) from
standard input, command line, or other means appropriate to
the operating environment to act as a "helper application" for
other applications (such as web browsers).
There may be RTSP applications different from those
initially envisioned by the contributors to the RTSP
specification for which the requirements above do not make
sense. Therefore, the recommendations above serve only as
guidelines instead of strict requirements.
D.1.1 Basic Playback
To support on-demand playback of media streams, the client MUST
additionally be able to do the following:
o generate the PAUSE request;
o implement the REDIRECT method, and the Location header.
D.1.2 Authentication-enabled
In order to access media presentations from RTSP servers that require
authentication, the client MUST additionally be able to do the
following:
o recognize the 401 (Unauthorized) status code;
o parse and include the WWW-Authenticate header;
o implement Basic Authentication and Digest Authentication.
D.2 Server
A minimal server implementation MUST be able to do the following:
H. Schulzrinne et. al. [Page 87]
Internet Draft RTSP February 22, 2002
o Implement the following methods: SETUP, TEARDOWN, OPTIONS and
either PLAY (for a minimal playback server) or RECORD (for a
minimal recording server).
If RECORD is implemented, ANNOUNCE SHOULD be implemented as
well.
o Include the following headers in responses: Connection,
Content-Length, Content-Type, Content-Language, Content-
Encoding, Transport, Public. The capability to include the
Location header should be implemented if the RECORD method is.
RTP-compliant implementations should also implement the RTP-
Info field.
o Parse and respond appropriately to the following headers in
requests: Connection, Session, Transport, Require.
Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial
implementations and/or to be a "good citizen".
o Implement RTP/AVP/UDP as a valid transport.
o Inclusion of the Server header.
o Implement the DESCRIBE method.
o Generate SDP session descriptions as defined in Appendix C
There may be RTSP applications different from those
initially envisioned by the contributors to the RTSP
specification for which the requirements above do not make
sense. Therefore, the recommendations above serve only as
guidelines instead of strict requirements.
D.2.1 Basic Playback
To support on-demand playback of media streams, the server MUST
additionally be able to do the following:
o Recognize the Range header, and return an error if seeking is
not supported.
o Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface
features, the following are highly recommended for on-demand media
H. Schulzrinne et. al. [Page 88]
Internet Draft RTSP February 22, 2002
servers:
o Include and parse the Range header, with NPT units.
Implementation of SMPTE units is recommended.
o Include the length of the media presentation in the media
initialization information.
o Include mappings from data-specific timestamps to NPT. When
RTP is used, the rtptime portion of the RTP-Info field may
be used to map RTP timestamps to NPT.
Client implementations may use the presence of length
information to determine if the clip is seekable, and
visably disable seeking features for clips for which the
length information is unavailable. A common use of the
presentation length is to implement a "slider bar" which
serves as both a progress indicator and a timeline
positioning tool.
Mappings from RTP timestamps to NPT are necessary to ensure correct
positioning of the slider bar.
D.2.2 Authentication-enabled
In order to correctly handle client authentication, the server MUST
additionally be able to do the following:
o Generate the 401 (Unauthorized) status code when
authentication is required for the resource.
o Parse and include the WWW-Authenticate header
o Implement Basic Authentication and Digest Authentication
E Changes
Since RFC 2326, the following issues were addressed:
o http://rtsp.org/bug448521 - URLs in Rtp-Info need to be quoted
o http://rtsp.org/bug448525 - Syntax for SSRC should be
clarified
o http://rtsp.org/bug461083 - Body w/o Content-Length
clarification
H. Schulzrinne et. al. [Page 89]
Internet Draft RTSP February 22, 2002
o http://rtsp.org/bug477407 - Transport BNF doesn't properly
deal with semicolon and comma
o http://rtsp.org/bug477413 - Transport BNF: mode parameter
issues
o http://rtsp.org/bug477416 - BNF error section 3.6 NPT
o http://rtsp.org/bug477421 - When to send response
o http://rtsp.org/bug507347 - Removal of destination redirection
Note that this list does not reflect minor changes in wording or
correction of typographical errors.
A word-by-word diff from RFC 2326 can be found at
http://rtsp.org/2002/drafts
F Author Addresses
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Anup Rao
Cisco
USA
electronic mail: anrao@cisco.com
Robert Lanphier
RealNetworks
P.O. Box 91123
Seattle, WA 98111-9223
USA
electronic mail: robla@real.com
G Acknowledgements
This draft is based on the functionality of the original RTSP draft
submitted in October 1996. It also borrows format and descriptions
from HTTP/1.1.
This document has benefited greatly from the comments of all those
participating in the MMUSIC-WG. In addition to those already
H. Schulzrinne et. al. [Page 90]
Internet Draft RTSP February 22, 2002
mentioned, the following individuals have contributed to this
specification:
Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
Bruce Butterfield, Ema Patki, Steve Casner, Francisco Cortes, Kelly
Djahandari, Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy
Grignon, V. Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub,
Volker Hilt, John K. Ho, Philipp Hoschka, Anne Jones, Anders Klemets,
Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
Marshall, Rob McCool, Aravind Narasimhan, David Oran, Joerg Ott,
Maria Papadopouli, Sujal Patel, Alagu Periyannan, Colin Perkins, Igor
Plotnikov, Jonathan Sergent, Pinaki Shah, David Singer, Jeff Smith,
Alexander Sokolsky, Dale Stammen, John Francis Stracke, David Walker,
and Magnus Westerlund.
H Bibliography
[1] H. Schulzrinne, "RTP profile for audio and video conferences with
minimal control," Request for Comments 1890, Internet Engineering
Task Force, Jan. 1996.
[2] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-Lee,
"Hypertext transfer protocol -- HTTP/1.1," Request for Comments 2068,
Internet Engineering Task Force, Jan. 1997.
[3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
"Internationalization of the hypertext markup language," Request for
Comments 2070, Internet Engineering Task Force, Jan. 1997.
[4] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," Request for Comments 2119, Internet Engineering Task Force,
Mar. 1997.
[5] ISO/IEC, "Information technology -- generic coding of moving
pictures and associated audio informaiton -- part 6: extension 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.
[6] M. Handley and V. Jacobson, "SDP: session description protocol,"
Request for Comments 2327, Internet Engineering Task Force, Apr.
1998.
[7] T. Dierks and C. Allen, "The TLS protocol version 1.0," Request
for Comments 2246, Internet Engineering Task Force, Jan. 1999.
[8] J. Franks, P. Hallam-Baker, J. Hostetler, P. Leach, A. Luotonen,
E. Sink, and L. Stewart, "An extension to HTTP : Digest access
H. Schulzrinne et. al. [Page 91]
Internet Draft RTSP February 22, 2002
authentication," Request for Comments 2069, Internet Engineering Task
Force, Jan. 1997.
[9] J. Postel, "User datagram protocol," Request for Comments 768,
Internet Engineering Task Force, Aug. 1980.
[10] C. Partridge and R. M. Hinden, "Version 2 of the reliable data
protocol (RDP)," Request for Comments 1151, Internet Engineering Task
Force, Apr. 1990.
[11] J. Postel, "Transmission control protocol," Request for Comments
793, Internet Engineering Task Force, Sept. 1981.
[12] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
session initiation protocol," Request for Comments 2543, Internet
Engineering Task Force, Mar. 1999.
[13] International Telecommunication Union, "Visual telephone systems
and equipment for local area networks which provide a non-guaranteed
quality of service," Recommendation H.323, Telecommunication
Standardization Sector of ITU, Geneva, Switzerland, May 1996.
[14] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
Request for Comments 1961, Internet Engineering Task Force, June
1996.
[15] J. Miller, P. Resnick, and D. Singer, "Rating services and
rating systems (and their machine readable descriptions),"
Recommendation REC-PICS-services-961031, W3C (World Wide Web
Consortium), Boston, Massachusetts, Oct. 1996.
[16] 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.
[17] D. Crocker, Ed., and P. Overell, "Augmented BNF for syntax
specifications: ABNF," Request for Comments 2234, Internet
Engineering Task Force, Nov. 1997.
[18] R. Braden and Ed, "Requirements for internet hosts - application
and support," Request for Comments 1123, Internet Engineering Task
Force, Oct. 1989.
[19] R. Elz, "A compact representation of IPv6 addresses," Request
for Comments 1924, Internet Engineering Task Force, Apr. 1996.
[20] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
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Internet Draft RTSP February 22, 2002
locators (URL)," Request for Comments 1738, Internet Engineering Task
Force, Dec. 1994.
[21] H. Schulzrinne, "A comprehensive multimedia control architecture
for the Internet," in Proc. International Workshop on Network and
Operating System Support for Digital Audio and Video (NOSSDAV) , (St.
Louis, Missouri), May 1997.
[22] F. Yergeau, "UTF-8, a transformation format of ISO 10646,"
Request for Comments 2279, Internet Engineering Task Force, Jan.
1998.
[23] R. Braden, "T/TCP -- TCP extensions for transactions functional
specification," Request for Comments 1644, Internet Engineering Task
Force, July 1994.
[24] W. R. Stevens, TCP/IP illustrated: the implementation , vol. 2.
Reading, Massachusetts: Addison-Wesley, 1994.
[25] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," Request for Comments 2326, Internet Engineering
Task Force, Apr. 1998.
[26] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
identifiers (URI): generic syntax," Request for Comments 2396,
Internet Engineering Task Force, Aug. 1998.
[27] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
a transport protocol for real-time applications," Request for
Comments 1889, Internet Engineering Task Force, Jan. 1996.
[28] R. Fielding, "Relative uniform resource locators," Request for
Comments 1808, Internet Engineering Task Force, June 1995.
Full Copyright Statement
Copyright (C) The Internet Society (2002). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind,
provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
H. Schulzrinne et. al. [Page 93]
Internet Draft RTSP February 22, 2002
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
H. Schulzrinne et. al. [Page 94]
Table of Contents
1 Introduction ........................................ 3
1.1 Purpose ............................................. 3
1.2 Requirements ........................................ 4
1.3 Terminology ......................................... 4
1.4 Protocol Properties ................................. 7
1.5 Extending RTSP ...................................... 8
1.6 Overall Operation ................................... 9
1.7 RTSP States ......................................... 10
1.8 Relationship with Other Protocols ................... 11
2 Notational Conventions .............................. 12
3 Protocol Parameters ................................. 12
3.1 H3.1 ................................................ 12
3.2 RTSP URL ............................................ 12
3.3 Conference Identifiers .............................. 14
3.4 Session Identifiers ................................. 14
3.5 SMPTE Relative Timestamps ........................... 14
3.6 Normal Play Time .................................... 15
3.7 Absolute Time ....................................... 16
3.8 Option Tags ......................................... 17
3.8.1 Registering New Option Tags with IANA ............... 17
4 RTSP Message ........................................ 17
4.1 Message Types ....................................... 18
4.2 Message Headers ..................................... 18
4.3 Message Body ........................................ 18
4.4 Message Length ...................................... 18
5 General Header Fields ............................... 19
6 Request ............................................. 19
6.1 Request Line ........................................ 19
6.2 Request Header Fields ............................... 20
7 Response ............................................ 21
7.1 Status-Line ......................................... 21
7.1.1 Status Code and Reason Phrase ....................... 21
7.1.2 Response Header Fields .............................. 24
8 Entity .............................................. 24
8.1 Entity Header Fields ................................ 25
8.2 Entity Body ......................................... 25
9 Connections ......................................... 25
9.1 Pipelining .......................................... 27
9.2 Reliability and Acknowledgements .................... 27
10 Method Definitions .................................. 28
10.1 OPTIONS ............................................. 28
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Internet Draft RTSP February 22, 2002
10.2 DESCRIBE ............................................ 29
10.3 ANNOUNCE ............................................ 30
10.4 SETUP ............................................... 31
10.5 PLAY ................................................ 32
10.6 PAUSE ............................................... 34
10.7 TEARDOWN ............................................ 36
10.8 GET_PARAMETER ....................................... 36
10.9 SET_PARAMETER ....................................... 37
10.10 REDIRECT ............................................ 38
10.11 RECORD .............................................. 38
10.12 Embedded (Interleaved) Binary Data .................. 39
11 Status Code Definitions ............................. 40
11.1 Success 2xx ......................................... 40
11.1.1 250 Low on Storage Space ............................ 40
11.2 Redirection 3xx ..................................... 40
11.3 Client Error 4xx .................................... 41
11.4 400 Bad Request ..................................... 41
11.4.1 405 Method Not Allowed .............................. 41
11.4.2 451 Parameter Not Understood ........................ 41
11.4.3 452 Conference Not Found ............................ 41
11.4.4 453 Not Enough Bandwidth ............................ 41
11.4.5 454 Session Not Found ............................... 41
11.4.6 455 Method Not Valid in This State .................. 41
11.4.7 456 Header Field Not Valid for Resource ............. 42
11.4.8 457 Invalid Range ................................... 42
11.4.9 458 Parameter Is Read-Only .......................... 42
11.4.10 459 Aggregate Operation Not Allowed ................. 42
11.4.11 460 Only Aggregate Operation Allowed ................ 42
11.4.12 461 Unsupported Transport ........................... 42
11.4.13 462 Destination Unreachable ......................... 42
11.5 Server Error 5xx .................................... 42
11.5.1 551 Option not supported ............................ 42
12 Header Field Definitions ............................ 43
12.1 Accept .............................................. 43
12.2 Accept-Encoding ..................................... 43
12.3 Accept-Language ..................................... 43
12.4 Accept-Ranges ....................................... 44
12.5 Allow ............................................... 44
12.6 Authorization ....................................... 44
12.7 Bandwidth ........................................... 44
12.8 Blocksize ........................................... 44
12.9 Cache-Control ....................................... 46
12.10 Conference .......................................... 48
12.11 Connection .......................................... 48
12.12 Content-Base ........................................ 48
12.13 Content-Encoding .................................... 48
12.14 Content-Language .................................... 48
12.15 Content-Length ...................................... 49
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Internet Draft RTSP February 22, 2002
12.16 Content-Location .................................... 49
12.17 Content-Type ........................................ 49
12.18 CSeq ................................................ 49
12.19 Date ................................................ 49
12.20 Expires ............................................. 49
12.21 From ................................................ 50
12.22 Host ................................................ 50
12.23 If-Match ............................................ 50
12.24 If-Modified-Since ................................... 51
12.25 Last-Modified ....................................... 51
12.26 Location ............................................ 51
12.27 Proxy-Authenticate .................................. 51
12.28 Proxy-Require ....................................... 52
12.29 Public .............................................. 52
12.30 Range .............................................. 52
12.31 Referer ............................................. 53
12.32 Retry-After ......................................... 53
12.33 Require ............................................. 53
12.34 RTP-Info ............................................ 54
12.35 Scale ............................................... 55
12.36 Speed ............................................... 56
12.37 Server .............................................. 57
12.38 Session ............................................. 57
12.39 Timestamp ........................................... 58
12.40 Transport ........................................... 58
12.41 Unsupported ......................................... 62
12.42 User-Agent .......................................... 62
12.43 Vary ................................................ 62
12.44 Via ................................................. 62
12.45 WWW-Authenticate .................................... 62
13 Caching ............................................. 62
14 Examples ............................................ 63
14.1 Media on Demand (Unicast) ........................... 63
14.2 Streaming of a Container file ....................... 65
14.3 Single Stream Container Files ....................... 68
14.4 Live Media Presentation Using Multicast ............. 70
14.5 Playing media into an existing session .............. 71
14.6 Recording ........................................... 72
15 Syntax .............................................. 73
15.1 Base Syntax ......................................... 73
16 Security Considerations ............................. 74
A RTSP Protocol State Machines ........................ 77
A.1 Client State Machine ................................ 77
A.2 Server State Machine ................................ 79
B Interaction with RTP ................................ 80
C Use of SDP for RTSP Session Descriptions ............ 81
C.1 Definitions ......................................... 82
C.1.1 Control URL ......................................... 82
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C.1.2 Media Streams ....................................... 82
C.1.3 Payload Type(s) ..................................... 83
C.1.4 Format-Specific Parameters .......................... 83
C.1.5 Range of Presentation ............................... 83
C.1.6 Time of Availability ................................ 83
C.1.7 Connection Information .............................. 84
C.1.8 Entity Tag .......................................... 84
C.2 Aggregate Control Not Available ..................... 84
C.3 Aggregate Control Available ......................... 85
D Minimal RTSP implementation ......................... 86
D.1 Client .............................................. 86
D.1.1 Basic Playback ...................................... 87
D.1.2 Authentication-enabled .............................. 87
D.2 Server .............................................. 87
D.2.1 Basic Playback ...................................... 88
D.2.2 Authentication-enabled .............................. 89
E Changes ............................................. 89
F Author Addresses .................................... 90
G Acknowledgements .................................... 90
H Bibliography ........................................ 91
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