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RFC2326 - Real Time Streaming Protocol (RTSP)

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Network Working Group H. Schulzrinne Request for Comments: 2326 Columbia U. Category: Standards Track A. Rao Netscape R. Lanphier RealNetworks April 1998 Real Time Streaming Protocol (RTSP) Status of this Memo This document specifies an Internet stan
  Network Working Group H. Schulzrinne
Request for Comments: 2326 Columbia U.
Category: Standards Track A. Rao
Netscape
R. Lanphier
RealNetworks
April 1998

Real Time Streaming Protocol (RTSP)

Status of this Memo

This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.

Copyright Notice

Copyright (C) The Internet Society (1998). All Rights Reserved.

Abstract

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
mechanisms based upon RTP (RFC1889).

Table of Contents

* 1 Introduction ................................................. 5
+ 1.1 Purpose ............................................... 5
+ 1.2 Requirements .......................................... 6
+ 1.3 Terminology ........................................... 6
+ 1.4 Protocol Properties ................................... 9
+ 1.5 Extending RTSP ........................................ 11
+ 1.6 Overall Operation ..................................... 11
+ 1.7 RTSP States ........................................... 12
+ 1.8 Relationship with Other Protocols ..................... 13
* 2 Notational Conventions ....................................... 14
* 3 Protocol Parameters .......................................... 14
+ 3.1 RTSP Version .......................................... 14

+ 3.2 RTSP URL .............................................. 14
+ 3.3 Conference Identifiers ................................ 16
+ 3.4 Session Identifiers ................................... 16
+ 3.5 SMPTE Relative Timestamps ............................. 16
+ 3.6 Normal Play Time ...................................... 17
+ 3.7 Absolute Time ......................................... 18
+ 3.8 Option Tags ........................................... 18
o 3.8.1 Registering New Option Tags with IANA .......... 18
* 4 RTSP Message ................................................. 19
+ 4.1 Message Types ......................................... 19
+ 4.2 Message Headers ....................................... 19
+ 4.3 Message Body .......................................... 19
+ 4.4 Message Length ........................................ 20
* 5 General Header Fields ........................................ 20
* 6 Request ...................................................... 20
+ 6.1 Request Line .......................................... 21
+ 6.2 Request Header Fields ................................. 21
* 7 Response ..................................................... 22
+ 7.1 Status-Line ........................................... 22
o 7.1.1 Status Code and Reason Phrase .................. 22
o 7.1.2 Response Header Fields ......................... 26
* 8 Entity ....................................................... 27
+ 8.1 Entity Header Fields .................................. 27
+ 8.2 Entity Body ........................................... 28
* 9 Connections .................................................. 28
+ 9.1 Pipelining ............................................ 28
+ 9.2 Reliability and Acknowledgements ...................... 28
* 10 Method Definitions .......................................... 29
+ 10.1 OPTIONS .............................................. 30
+ 10.2 DESCRIBE ............................................. 31
+ 10.3 ANNOUNCE ............................................. 32
+ 10.4 SETUP ................................................ 33
+ 10.5 PLAY ................................................. 34
+ 10.6 PAUSE ................................................ 36
+ 10.7 TEARDOWN ............................................. 37
+ 10.8 GET_PARAMETER ........................................ 37
+ 10.9 SET_PARAMETER ........................................ 38
+ 10.10 REDIRECT ............................................ 39
+ 10.11 RECORD .............................................. 39
+ 10.12 Embedded (Interleaved) Binary Data .................. 40
* 11 Status Code Definitions ..................................... 41
+ 11.1 Success 2xx .......................................... 41
o 11.1.1 250 Low on Storage Space ...................... 41
+ 11.2 Redirection 3xx ...................................... 41
+ 11.3 Client Error 4xx ..................................... 42
o 11.3.1 405 Method Not Allowed ........................ 42
o 11.3.2 451 Parameter Not Understood .................. 42
o 11.3.3 452 Conference Not Found ...................... 42

o 11.3.4 453 Not Enough Bandwidth ...................... 42
o 11.3.5 454 Session Not Found ......................... 42
o 11.3.6 455 Method Not Valid in This State ............ 42
o 11.3.7 456 Header Field Not Valid for Resource ....... 42
o 11.3.8 457 Invalid Range ............................. 43
o 11.3.9 458 Parameter Is Read-Only .................... 43
o 11.3.10 459 Aggregate Operation Not Allowed .......... 43
o 11.3.11 460 Only Aggregate Operation Allowed ......... 43
o 11.3.12 461 Unsupported Transport .................... 43
o 11.3.13 462 Destination Unreachable .................. 43
o 11.3.14 551 Option not supported ..................... 43
* 12 Header Field Definitions .................................... 44
+ 12.1 Accept ............................................... 46
+ 12.2 Accept-Encoding ...................................... 46
+ 12.3 Accept-Language ...................................... 46
+ 12.4 Allow ................................................ 46
+ 12.5 Authorization ........................................ 46
+ 12.6 Bandwidth ............................................ 46
+ 12.7 Blocksize ............................................ 47
+ 12.8 Cache-Control ........................................ 47
+ 12.9 Conference ........................................... 49
+ 12.10 Connection .......................................... 49
+ 12.11 Content-Base ........................................ 49
+ 12.12 Content-Encoding .................................... 49
+ 12.13 Content-Language .................................... 50
+ 12.14 Content-Length ...................................... 50
+ 12.15 Content-Location .................................... 50
+ 12.16 Content-Type ........................................ 50
+ 12.17 CSeq ................................................ 50
+ 12.18 Date ................................................ 50
+ 12.19 Expires ............................................. 50
+ 12.20 From ................................................ 51
+ 12.21 Host ................................................ 51
+ 12.22 If-Match ............................................ 51
+ 12.23 If-Modified-Since ................................... 52
+ 12.24 Last-Modified........................................ 52
+ 12.25 Location ............................................ 52
+ 12.26 Proxy-Authenticate .................................. 52
+ 12.27 Proxy-Require ....................................... 52
+ 12.28 Public .............................................. 53
+ 12.29 Range ............................................... 53
+ 12.30 Referer ............................................. 54
+ 12.31 Retry-After ......................................... 54
+ 12.32 Require ............................................. 54
+ 12.33 RTP-Info ............................................ 55
+ 12.34 Scale ............................................... 56
+ 12.35 Speed ............................................... 57
+ 12.36 Server .............................................. 57

+ 12.37 Session ............................................. 57
+ 12.38 Timestamp ........................................... 58
+ 12.39 Transport ........................................... 58
+ 12.40 Unsupported ......................................... 62
+ 12.41 User-Agent .......................................... 62
+ 12.42 Vary ................................................ 62
+ 12.43 Via ................................................. 62
+ 12.44 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 ........................ 67
+ 14.4 Live Media Presentation Using Multicast .............. 69
+ 14.5 Playing media into an existing session ............... 70
+ 14.6 Recording ............................................ 71
* 15 Syntax ...................................................... 72
+ 15.1 Base Syntax .......................................... 72
* 16 Security Considerations ..................................... 73
* A RTSP Protocol State Machines ................................. 76
+ A.1 Client State Machine .................................. 76
+ A.2 Server State Machine .................................. 77
* B Interaction with RTP ......................................... 79
* C Use of SDP for RTSP Session Descriptions ..................... 80
+ C.1 Definitions ........................................... 80
o C.1.1 Control URL .................................... 80
o C.1.2 Media streams .................................. 81
o C.1.3 Payload type(s) ................................ 81
o C.1.4 Format-specific parameters ..................... 81
o C.1.5 Range of presentation .......................... 82
o C.1.6 Time of availability ........................... 82
o C.1.7 Connection Information ......................... 82
o C.1.8 Entity Tag ..................................... 82
+ C.2 Aggregate Control Not Available ....................... 83
+ C.3 Aggregate Control Available ........................... 83
* D Minimal RTSP implementation .................................. 85
+ D.1 Client ................................................ 85
o D.1.1 Basic Playback ................................. 86
o D.1.2 Authentication-enabled ......................... 86
+ D.2 Server ................................................ 86
o D.2.1 Basic Playback ................................. 87
o D.2.2 Authentication-enabled ......................... 87
* E Authors' Addresses ........................................... 88
* F Acknowledgements ............................................. 89
* References ..................................................... 90
* Full Copyright Statement ....................................... 92

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:

* RTSP introduces a number of new methods and has a different
protocol identifier.
* An RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP.
* Both an RTSP server and client can issue requests.
* Data is carried out-of-band by a different protocol. (There is an
exception to this.)
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
consistent with current HTML internationalization efforts [3].
* 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 RFC2119 [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 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 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 (RFC2327 [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.

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. All HTTP authentication
mechanisms such as basic (RFC2068 [2, Section 11.1]) and
digest authentication (RFC2069 [8]) are directly applicable.
One may also reuse transport or network layer security
mechanisms.

Transport-independent:
RTSP may use either an unreliable datagram protocol (UDP) (RFC
768 [9]), a reliable datagram protocol (RDP, RFC1151, not
widely used [10]) or a reliable stream protocol such as TCP
(RFC793 [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 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:

* A server may only be capable of playback thus has no need to
support the RECORD request.
* A server may not be capable of seeking (absolute positioning) if
it is to support live events only.
* 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:

* 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 12.32).
* 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.
* 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.

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
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

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 (RFC2068 [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 RFC2234 [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

of RFC1123 \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 RFC1924 [19]).

A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions [H3.2] of
URLs (RFC1738 [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".

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.

smpte-range = smpte-type "=" 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
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-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:

19961108T143720.25Z

3.8 Option Tags

Option tags are unique identifiers used to designate new options in
RTSP. These tags are used in Require (Section 12.32) and Proxy-
Require (Section 12.27) header fields.

Syntax:

option-tag = 1*xchar

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:

* 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.
* Indication of who has change control over the option (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium or a particular company or group of
companies);

* A reference to a further description, if available, for example
(in order of preference) an RFC, a published paper, a patent
filing, a technical report, documented source code or a computer
manual;
* For proprietary options, contact information (postal and email
address);

4 RTSP Message

RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC2279 [21]). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by
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 RFC2279 [21])

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.)

2. If a Content-Length header field (section 12.14) 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.

3. By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a
response.)

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. Even though Content-Length must be present if there is
any entity body, the rules ensure reasonable behavior even if the
length is not given explicitly.

5 General Header Fields

See [H4.5], except that Pragma, Transfer-Encoding and Upgrade headers
are not defined:

general-header = Cache-Control ; Section 12.8
| Connection ; Section 12.10
| Date ; Section 12.18
| Via ; Section 12.43

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

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 12.5
| From ; Section 12.20
| If-Modified-Since ; Section 12.23
| Range ; Section 12.29
| Referer ; Section 12.30
| User-Agent ; Section 12.41

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.

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

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:

* 1xx: Informational - Request received, continuing process
* 2xx: Success - The action was successfully received, understood,
and accepted
* 3xx: Redirection - Further action must be taken in order to
complete the request
* 4xx: Client Error - The request contains bad syntax or cannot be
fulfilled
* 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
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