rfc2326.txt
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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.
* 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
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HTTP or other means such as email and may not necessarily be stored
on the media server.
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
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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
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 12.37) 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
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request has been acknowledged.
Re-using HTTP functionality has advantages in at least two areas,
namely security and proxies. The requirements are 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 memo, 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
[H3.1] 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
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of RFC 1123 \cite{rfc1123}>
port = *DIGIT
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 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".
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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 = 1*( ALPHA | DIGIT | safe )
3.5 SMPTE Relative Timestamps
A SMPTE relative timestamp expresses time relative to the start of
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.
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