Internet Engineering Task Force MMUSIC WG
Internet Draft H. Schulzrinne
Columbia U.
A. Rao
Cisco
R. Lanphier
RealNetworks
M. Westerlund
Ericsson
draft-ietf-mmusic-rfc2326bis-01.txt
June 06, 2002
Expires: December, 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.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
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Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
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rial or to cite them other than as "work in progress".
The list of current Internet-Drafts can be accessed at
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Abstract
This memorandum is a revision of RFC 2326, which is currently a Pro-
posed Standard.
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time proper-
ties. RTSP provides an extensible framework to enable controlled, on-
demand delivery of real-time data, such as audio and video. Sources
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Internet Draft RTSP June 06, 2002
of data can include both live data feeds and stored clips. This pro-
tocol 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 (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 con-
tinuous streams itself, although interleaving of the continuous media
stream with the control stream is possible (see Section 10.13). 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 presenta-
tion 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 trans-
port 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 contin-
uous media.
The protocol is intentionally similar in syntax and operation to
HTTP/1.1 [26] 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 pro-
tocol 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 [26]
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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 pre-
sentation 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 dis-
tributed teaching applications. Several parties in the confer-
ence 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 [26].
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 [26]. Terms
not listed here are defined as in HTTP/1.1.
Aggregate control: The control of the multiple streams using a sin-
gle 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.
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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 exam-
ples of continuous media are audio and motion video. Continu-
ous 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 ini-
tialization 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 dif-
ferent 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 applica-
tion group. When using RTP, a stream consists of all RTP and
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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 connection-
less 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 presenta-
tion, such as the set of encodings, network addresses and
information about the content. Other IETF protocols such as
SDP (RFC 2327 [24]) use the term "session" for a live presen-
tation. The presentation description may take several differ-
ent formats, including but not limited to the session descrip-
tion 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.
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Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers.
Secure: RTSP re-uses web security mechanisms, either at the trans-
port level (TLS, RFC 2246 [27]) or within the protocol itself.
All HTTP authentication mechanisms such as basic (RFC 2616
[26]) and digest authentication (RFC 2069 [6]) are directly
applicable.
Transport-independent: RTSP may use either an unreliable datagram
protocol (UDP) (RFC 768 [7]), a reliable datagram protocol
(RDP, RFC 1151, not widely used [8]) or a reliable stream pro-
tocol such as TCP (RFC 793 [9]) as it implements application-
level reliability.
Multi-server capable: Each media stream within a presentation can
reside on a different server. The client automatically estab-
lishes 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 record-
ing and playback devices, as well as devices that can alter-
nate between the two modes ("VCR").
Separation of stream control and conference initiation: Stream con-
trol is divorced from inviting a media server to a conference.
In particular, SIP [10] or H.323 [28] 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 presenta-
tion 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 [11]) fire-
walls. 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
[12,13]) for associating labels with content. However, RTSP
does not just add methods to HTTP since the controlling
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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 possi-
ble. The protocol would not address how several clients nego-
tiate 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 sup-
port 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 [26],
where the methods described in [H19.5] are not likely to be supported
across all servers.
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RTSP can be extended in three ways, listed here in order of the mag-
nitude 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 meth-
ods 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 presen-
tation 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 main-
tains 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, lan-
guage, 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 iden-
tified 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 exam-
ple, audio and video streams can be split across servers for load
sharing. The description also enumerates which transport methods the
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server is capable of.
Besides the media parameters, the network destination address and
port need to be determined. Several modes of operation can be distin-
guished:
Unicast: The media is transmitted to the source of the RTSP
request, with the port number chosen by the client. Alterna-
tively, the media is transmitted on the same reliable stream
as RTSP.
Multicast, server chooses address: The media server picks the mul-
ticast 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 descrip-
tion, established by means outside the scope of this specifi-
cation.
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 con-
trolled by RTSP requests issued sequentially on different TCP connec-
tions. 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 follow-
ing play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and TEAR-
DOWN.
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.
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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 inter-
act with HTTP in that the initial contact with streaming content is
often to be made through a web page. The current protocol specifica-
tion 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 stan-
dalone 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 asymmet-
ric 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 parame-
ters 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 (RFC 2616 [26]).
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 [14], with the differ-
ence that this RTSP specification maintains the "1#" notation for
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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 understand-
ing of why things are the way that they are in RTSP.
3 Protocol Parameters
3.1 RTSP Version
HTTP Specification Section [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 syn-
tax 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 [15]>
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 pro-
tocol (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 [16]).
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A presentation or a stream is identified by a textual media identi-
fier, using the character set and escape conventions [H3.2] of URLs
(RFC 1738 [17]). 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 individ-
ual 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 indi-
vidual 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 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.)
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session-id = 8*( ALPHA | DIGIT | safe )
3.4 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-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
3.5 Normal Play Time
Normal play time (NPT) indicates the stream absolute position rela-
tive 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.
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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" "="] 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 opti-
mized 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.
3.6 Absolute Time
Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
Fractions of a second may be indicated.
utc-range = ["clock" "="] utc-range-spec
utc-range-spec = ( utc-time "-" [ utc-time ] ) | ( "-" utc-time )
utc-time = utc-date "T" utc-time "Z"
utc-date = 8DIGIT ; < YYYYMMDD >
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utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC:
19961108T143720.25Z
3.7 Option Tags
Option tags are unique identifiers used to designate new options in
RTSP. These tags are used in in Require (Section 12.32) and Proxy-
Require (Section 12.27) 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 regis-
ter the new option with the Internet Assigned Numbers Authority
(IANA).
3.7.1 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 exam-
ple, IETF, ISO, ITU-T, other international standardization bod-
ies, 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 fil-
ing, a technical report, documented source code or a computer
manual;
+ For proprietary options, contact information (postal and email
address);
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4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [18]). Lines are terminated by CRLF, but
receivers should be prepared to also interpret CR and LF by them-
selves 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 [18])
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):
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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.
Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
transfer coding(see [H3.6.1]) and requires the presence of the Con-
tent-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 chun-
ked transfer encoding unnecessary.
5 General Header Fields
See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade,
and Warning headers are not defined:
general-header = Cache-Control ; Section 12.9
| Connection ; Section 12.10
| CSeq ; Section 12.17
| Date ; Section 12.18
| 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 ver-
sion in use.
Request = Request-Line ; Section 6.1
*( general-header ; Section 5
| request-header ; Section 6.2
| entity-header ) ; Section 8.1
CRLF
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[ 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 12.6
| Bandwidth ; Section 12.7
| Blocksize ; Section 12.8
| From ; Section 12.20
| If-Modified-Since ; Section 12.23
| Proxy-Require ; Section 12.27
| Range ; Section 12.29
| Referer ; Section 12.30
| Require ; Section 12.32
| Scale ; Section 12.34
| Session ; Section 12.37
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| Speed ; Section 12.35
| Transport ; Section 12.40
| User-Agent ; Section 12.42
Note that in contrast to HTTP/1.1 [26], 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
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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 com-
plete 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 [26] status codes and
adds RTSP-specific status codes starting at x50 to avoid conflicts
with newly defined HTTP status codes.
Status-Code = "100" ; Continue
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| "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
| "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" ; reserved
| "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
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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 unrecog-
nized response MUST NOT be cached. For example, if an unrecognized
status code of 431 is received by the client, it can 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 addi-
tional 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 12.25
| Proxy-Authenticate ; Section 12.26
| Public ; Section 12.28
| Range ; Section 12.29
| Retry-After ; Section 12.31
| RTP-Info ; Section 12.33
| Scale ; Section 12.34
| Session ; Section 12.37
| Server ; Section 12.36
| Speed ; Section 12.35
| 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 combina-
tion with a change in the protocol version. However, new or experi-
mental 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
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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 reserved n/a
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
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505 RTSP Version Not Supported all
551 Option not support all
Table 1: Status codes and their usage with RTSP methods
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.
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 12.11
| Content-Encoding ; Section 12.12
| Content-Language ; Section 12.13
| Content-Length ; Section 12.14
| Content-Location ; Section 12.15
| Content-Type ; Section 12.16
| Expires ; Section 12.19
| Last-Modified ; Section 12.24
| 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]
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9 Connections
RTSP requests can be transmitted in several different ways:
+ persistent transport connections used for several request-
response transactions;
+ one connection per request/response transaction;
+ 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 connec-
tions, as the media server otherwise has no reliable way of 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) [15], 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 can-
not typically take advantage of the application-layer retrans-
mission since the transport stack will not deliver the
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application-layer retransmission before the first attempt has
reached the receiver. If the packet loss is caused by conges-
tion, multiple retransmissions at different layers will exac-
erbate the congestion.
If RTSP is used over a small-RTT LAN, standard procedures for opti-
mizing initial TCP round trip estimates, such as those used in T/TCP
(RFC 1644) [19], can be beneficial.
The Timestamp header (Section 12.39) is used to avoid the retransmis-
sion ambiguity problem [20] and obviates the need for Karn's algo-
rithm.
Each request carries a sequence number in the CSeq header (Section
12.17), which is incremented by one for each distinct request trans-
mitted. 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).
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.
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
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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
PING C->S, S->C P,S optional
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
The behavior is equivalent to that described in [H9.2]. An OPTIONS
request may be issued at any time, e.g., if the client is about to
try 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
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resource. The DESCRIBE reply-response pair constitutes the media ini-
tialization 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)
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 informa-
tion 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 parame-
ters, 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 initial-
ization for the set of streams that it describes, and discour-
aging use of DESCRIBE for media indirection, we avoid looping
problems that might result from other approaches.
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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:
+ via RTSP's DESCRIBE method;
+ via some other protocol (HTTP, email attachment, etc.);
+ via the command line or standard input (thus working as a browser
helper application launched with an SDP file or other media ini-
tialization 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
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 ses-
sion 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
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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 parame-
ters even if it has no influence over these parameters, for 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 ini-
tialization.
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
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The server generates session identifiers in response to SETUP
requests. If a SETUP request to a server includes a session identi-
fier, 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 mecha-
nism 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 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
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pause point.
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
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
RTP-Info:url=rtsp://audio.example.com/twister.en;seq=14783;rtptime=2345962545
For playing back a recording of a live presentation, it may be desir-
able 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
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Range: clock=19961108T142300Z-19961108T143520Z
RTP-Info:url=rtsp://audio.example.com/meeting.en;seq=53745;rtptime=484589019
A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats.
All range specifiers in this specification allow for ranges with |
unspecified begin times (e.g. "npt=-30"). When used in a PLAY |
request, the server treats this as a request to start/resume playback |
from the current pause point, ending at the end time specified in the |
Range header.
10.6 PAUSE
The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only play-
back 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.
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 a single value, expressed as |
the beginning value an open range. For example, the following clip |
will be played from 10 seconds through 21 seconds of the clip's nor- |
mal play time: |
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C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0 |
CSeq: 834 |
Session: 12345678 |
Range: npt=10-30 |
S->C: RTSP/1.0 200 OK |
CSeq: 834 |
Date: 23 Jan 1997 15:35:06 GMT |
Range: npt=10-30 |
RTP-Info:url=rtsp://example.com/fizzle/foo/audiotrack;seq=5712;rtptime=934207921,|
url=rtsp://example.com/fizzle/foo/videotrack;seq=57654;rtptime=2792482193|
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 |
CSeq: 835 |
Session: 12345678 |
Range: npt=21- |
S->C: RTSP/1.0 200 OK |
CSeq: 835 |
Date: 23 Jan 1997 15:35:09 GMT |
Range: npt=21- |
The normal play time for the stream is set to the pause point. The
pause request becomes effective the first time the server is encoun-
tering 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 nor-
mal 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.
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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, assum-
ing 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
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 presenta-
tion URI for this presentation, any RTSP session identifier associ-
ated with the session is no longer valid. Unless all transport param-
eters are defined by the session description, a SETUP request has to
be issued before the session can be played again.
A server that after processing the TEARDOWN still has a valid session
MUST in the response return a session header.
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
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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
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 presenta-
tion 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 parame-
ter 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
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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
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-
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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 pro-
vided 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
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 sup-
port the clock range format; the smpte format does not make sense.
In this example, the media server was previously invited to the con-
ference indicated.
C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
CSeq: 954
Session: 12345678
Conference: 128.16.64.19/32492374
Note: this example needs work, or needs to be removed.
10.12 PING |
This method is a bi-directional mechanism for server or client live- |
ness checking. It has no side effects. |
Prior to using this method, an OPTIONS method MUST be issued in the |
direction which the PING method would be used. This method MUST NOT |
be used if support is not indicated by the Public header. |
When a proxy is in use, PING with a * indicates a single-hop liveness |
check, whereas PING with a URL indicates an end-to-end liveness |
check. |
Example: |
C->S: PING * RTSP/1.0 |
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CSeq: 123 |
S->C: RTSP/1.0 200 OK |
CSeq: 123 |
10.13 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, fol-
lowed by the length of the encapsulated binary data as a binary, two-
byte integer in network byte order. The stream data follows immedi-
ately afterwards, without a CRLF, but including the upper-layer pro-
tocol 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 con-
venient way to tunnel RTP/RTCP packets through the TCP control
connection when required by the network configuration and
transfer them onto UDP when possible.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
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
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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 indi-
cate what time period it may still be able to record. Since other
processes on the server may be consuming storage space simultane-
ously, a client should take this only as an estimate.
11.2 Redirection 3xx
See [H10.3].
Within RTSP, redirection may be used for load balancing or redirect-
ing 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
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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 modifica-
tions [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 reserved
This error code was removed from RFC 2326 [21] and is obsolete.
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 fail-
ure.
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
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.
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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 modi-
fied.
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 specifica-
tion.
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 param-
eter in the Transport field.
11.5 Server Error 5xx
11.5.1 551 Option not supported
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
The general syntax for header fields is covered in Section 4.2 This |
section lists the full set of header fields along with notes on |
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method direction object requirement acronym Body
-----------------------------------------------------------
DESCRIBE C->S P,S recommended DES r
ANNOUNCE C->S, S->C P,S optional ANN R
GET_PARAMETER C->S, S->C P,S optional GPR R,r
OPTIONS C->S P,S required OPT
S->C optional
PAUSE C->S P,S recommended PSE
PING C->S, S->C P,S optional PNG
PLAY C->S P,S required PLY
RECORD C->S P,S optional REC
REDIRECT S->C P,S optional RDR
SETUP C->S S required STP
SET_PARAMETER C->S, S->C P,S optional SPR R,r?
TEARDOWN C->S P,S required TRD
Table 3: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on. Body notes if a method
is allowed to carry body and in which direction, R = Request,
r=response. Note: There has been some usage of the body to convey
more information in error messages for responses containing error
codes. Some error messages seem to mandate such usage.
syntax, meaning, and usage. Throughout this section, we use [HX.Y] |
to refer to Section X.Y of the current HTTP/1.1 specification RFC |
2616 [26]. Examples of each header field are given. |
Information about header fields in relation to methods and proxy pro- |
cessing is summarized in Table 4. |
The "where" column describes the request and response types in which |
the header field can be used. Values in this column are: |
R: header field may only appear in requests; |
r: header field may only appear in responses; |
2xx, 4xx, etc.: A numerical value or range indicates response codes |
with which the header field can be used; |
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c: header field is copied from the request to the response. |
An empty entry in the "where" column indicates that the header field |
may be present in all requests and responses. |
The "proxy" column describes the operations a proxy may perform on a |
header field: |
a: A proxy can add or concatenate the header field if not present. |
m: A proxy can modify an existing header field value. |
d: A proxy can delete a header field value. |
r: A proxy must be able to read the header field, and thus this |
header field cannot be encrypted. |
The rest of the columns relate to the presence of a header field in a |
method. The method names are abbreviated according to table 3: |
c: Conditional; requirements on the header field depend on the con- |
text of the message. |
m: The header field is mandatory. |
m*: The header field SHOULD be sent, but clients/servers need to be |
prepared to receive messages without that header field. |
o: The header field is optional. |
t: The header field SHOULD be sent, but clients/servers need to be |
prepared to receive messages without that header field. If a |
stream-based protocol (such as TCP) is used as a transport, |
then the header field MUST be sent. |
*: The header field is required if the message body is not empty. |
See sections 12.14, 12.16 and 4.3 for details. |
-: The header field is not applicable. |
"Optional" means that a Client/Server MAY include the header field in |
a request or response, and a Client/Server MAY ignore the header |
field if present in the request or response (The exception to this |
rule is the Require header field discussed in 12.32). A "mandatory" |
header field MUST be present in a request, and MUST be understood by |
the Client/Server receiving the request. A mandatory response header |
field MUST be present in the response, and the header field MUST be |
understood by the Client/Server processing the response. "Not |
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applicable" means that the header field MUST NOT be present in a |
request. If one is placed in a request by mistake, it MUST be ignored |
by the Client/Server receiving the request. Similarly, a header field |
labeled "not applicable" for a response means that the Client/Server |
MUST NOT place the header field in the response, and the |
Client/Server MUST ignore the header field in the response. |
A Client/Server SHOULD ignore extension header parameters that are |
not understood. |
The From, Location, and RTP-Info header fields contain a URI. If the |
URI contains a comma, or semicolon, the URI MUST be enclosed in dou- |
ble quotas ("). Any URI parameters are contained within these quotas. |
If the URI is not enclosed in double quotas, any semicolon- delimited |
parameters are header-parameters, not URI parameters. |
12.1 Accept
The Accept request-header field can be used to specify certain pre-
sentation description content types which are acceptable for the
response.
The "level" parameter for presentation descriptions is prop-
erly 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
See [H14.4]. Note that the language specified applies to the presen-
tation description and any reason phrases, not the media content.
12.4 Accept-Ranges |
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Header Where Proxy DES OPT GPR SPR ANN STP PLY REC PSE TRD RDR PNG
---------------------------------------------------------------------------------
Accept R o - - - - - - - - - - -
Accept-Encoding R r o - - - - - - - - - - -
Accept-Language R r o - - - - - - - - - - -
Accept-Ranges r - - - - - - o - - - - -
Allow 405 - - - - m - m m m - - -
Authorization R o o o o o o o o o o o o
Bandwidth R o - - o - o o - - - - -
Blocksize R o - - o - o o - - - - -
Cache-Control r - - - - - o - - - - - -
Connection o o o o o o o o o o o -
Content-Base R - - o o o - - - - - - -
Content-Base r o - o o - - - - - - - -
Content-Base 4xx o o o o o o o o o o o -
Content-Encoding R r - - o o o - - - - - - -
Content-Encoding r r o - o o - - - - - - - -
Content-Encoding 4xx r o o o o o o o o o o o -
Content-Language R r - - o o o - - - - - - -
Content-Language r r o - o o - - - - - - - -
Content-Language 4xx r o o o o o o o o o o o -
Content-Length R r - - * * * - - - - - - -
Content-Length r r * - * * - - - - - - - -
Content-Length 4xx r * * * * * * * * * * * -
Content-Location R - - o o o - - - - - - -
Content-Location r o - o o - - - - - - - -
Content-Location 4xx o o o o o o o o o o o -
Content-Type R - - * * * - - - - - - -
Content-Type r * - * * - - - - - - - -
Content-Type 4xx * * * * * * * * * * * -
CSeq Rc m m m m m m m m m m m m
Date am o o o o o o o o o o o o
Expires r r o - - - - - - - - - - -
From R r o o o o o o o o o o o o
Host o o o o o o o o o o o o
If-Match R r - - - - - o - - - - - -
If-Modified-Since R r o - - - - o - - - - - -
Last-Modified R r - - - - o - - - - - - -
Last-Modified r r o - o - - - - - - - - -
Location R - - - - - - - - - - m -
Location 3xx m - - - - m - - - - - -
Proxy-Authenticate 407 amr m m m m m m m m m m m m
Proxy-Require R ar o o o o o o o o o o o o
Public r admr - m* - - - - - - - - - -
Public 501 admr m* m* m* m* m* m* m* m* m* m* m* m*
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Range R - - - - - - o - o - o -
Range r - - - - - - m* - - - - -
Referer R o o o o o o o o o o o -
Require R o o o o o o o o o o o o
Retry-After 3xx,503 o o o o - o - - - - - -
RTP-Info r - - - - - - m - - - - -
Scale - - - - - - o o - - - -
Session R - o o o m o m m m m m m
Session r - c c c m m m m m o m m
Server R - o o o o - - - - - o o
Server r o o o o o o o o o o - o
Speed - - - - - - o - - - - -
Supported R o o o o o o o o o o o o
Supported r c c c c c c c c c c c c
Timestamp R o o o o o o o o o o o o
Timestamp c m m m m m m m m m m m m
Transport - - - - - m - - - - - -
Unsupported r c c c c c c c c c c c c
User-Agent R m* m* m* m* m* m* m* m* m* m* - -
User-Agent r - - - - - - - - - - m* -
Vary r c c c c c c c c c c - -
Via R amr o o o o o o o o o o o o
Via c dr m m m m m m m m m m m m
WWW-Authenticate 401 m m m m m m m m m m m m
---------------------------------------------------------------------------------
Header Where Proxy DES OPT GPR SPR ANN STP PLY REC PSE TRD RDR PNG
Table 4: Overview of RTSP header fields
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
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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
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 appli-
cation, 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-direc-
tive for a specific cache.
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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 ) ]
no-cache: Indicates that the media stream MUST NOT be cached any-
where. 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 opera-
tional 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 authenti-
cation 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
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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 net-
work 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 con-
straints of the request, or respond with a 504 (Gateway Time-
out) 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.
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 Connection
See [H14.10]
12.11 Content-Base
The Content-Base entity-header field may be used to specify the base
URI for resolving relative URLs within the entity. This header field
is described as Base in RFC 1808, which is expected to be revised.
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Content-Base = "Content-Base" ":" absoluteURI
If no Content-Base field is present, the base URI of an entity is
defined either by its Content-Location (if that Content-Location URI
is an absolute URI) or the URI used to initiate the request, in that
order of precedence. Note, however, that the base URI of the contents
within the entity-body may be redefined within that entity-body.
12.12 Content-Encoding
See [H14.11]
12.13 Content-Language
See [H14.12]
12.14 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
[H14.13].
12.15 Content-Location
See [H14.14]
12.16 Content-Type
See [H14.17]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and
parameter-value types.
12.17 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 num-
ber. 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).
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CSeq = "Cseq" ":" 1*DIGIT
12.18 Date
See [H14.18].
12.19 Expires
The Expires entity-header field gives a date and time after which the
description or media-stream should be considered stale. The interpre-
tation 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
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.
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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).
12.20 From
See [H14.22].
12.21 Host
The Host HTTP request header field [H14.23] is not needed for RTSP.
It should be silently ignored if sent.
12.22 If-Match
See [H14.24].
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.
12.23 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
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12.24 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 modifica-
tion date and time of the description, for SETUP that of the media
stream.
12.25 Location
See [H14.30].
12.26 Proxy-Authenticate
See [H14.33].
12.27 Proxy-Require
The Proxy-Require request-header field is used to indicate proxy-sen-
sitive 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 12.32 for more details on the mechanics of this message
and a usage example.
12.28 Public
The Public response-header field lists the set of methods supported
by the server. The purpose of this field is strictly to inform the
recipient of the capabilities of the server regarding unusual meth-
ods. The methods listed may or may not be applicable to the Request-
URI; the Allow header field (section 14.7) MAY be used to indicate
methods allowed for a particular URI.
Public = "Public" ":" 1#method
Example of use:
Public: OPTIONS, MGET, MHEAD, GET, HEAD
This header field applies only to the server directly connected to
the client (i.e., the nearest neighbor in a chain of connections).
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If the response passes through a proxy, the proxy MUST either remove
the Public header field or replace it with one applicable to its own
capabilities.
12.29 Range
The Range request and response header field specifies a range of
time. The range can be specified in a number of units. This specifi-
cation defines the smpte (Section 3.4), npt (Section 3.5), and clock
(Section 3.6) range units. Within RTSP, byte ranges [H14.35.1] 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 under-
stand the NPT range format and SHOULD understand the SMPTE range for-
mat. The Range response header indicates what range of time is actu-
ally 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
The notation is similar to that used for the HTTP/1.1 [26]
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.
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12.30 Referer
See [H14.36]. The URL refers to that of the presentation description,
typically retrieved via HTTP.
12.31 Retry-After
See [H14.37].
12.32 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
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
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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 trans-
mitted 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 12.27).
12.33 RTP-Info
The RTP-Info response-header field is used to set RTP-specific param-
eters 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 con-
trol, 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 indi-
cated 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 information
is not sufficient to generate a mapping from RTP times-
tamps to NPT. Furthermore, in order to ensure that this
information is available at the necessary time (immedi-
ately at startup or after a seek), and that it is deliv-
ered reliably, this mapping is placed in the RTSP control
channel.
In order to compensate for drift for long, uninterrupted pre-
sentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later
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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 [21].
absoluteURI and relativeURI are defined in RFC 2396 [22].
Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
12.34 Scale
A scale value of 1 indicates normal play or record at the normal for-
ward 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 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 frag-
ments of audio.
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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
12.35 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. Implementa-
tion 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:
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 presen-
tation at a higher or lower rate is necessary. Implementors should
keep in mind that bandwidth for the session may be negotiated before-
hand (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.
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12.36 Server
See [H14.38]
12.37 Session
The Session request-header and response-header field identifies an
RTSP session started by the media server in a SETUP response and con-
cluded by TEARDOWN on the presentation URL. The session identifier is
chosen by the media server (see Section 3.3) 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 pre-
pared 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 pos-
sible 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 deliv-
ery requests for the same URL coming from the same client.
The response 454 (Session Not Found) is returned if the session iden-
tifier is invalid.
12.38 Supported |
The Supported header field enumerates all the extensions supported by |
the client or server. When offered in a request, the receiver MUST |
respond with its cooresponding Supported header. |
The Supported header field contains a list of option tags, described |
in Section 3.7, that are understood by the client or server. |
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Example: |
C->S OPTIONS rtsp://example.com/ RTSP/1.0 ||
Supported: foo, bar, blech ||
SuppoS->C:RTSP/1.0e200 OKz ||
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 trans-
port 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.
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A transport-spec transport option may only contain one of any given |
parameter within it. Parameters may be given in any order. Addition- |
ally, it may only contain the unicast or multicast transport parame- |
ter.
The Transport header field is restricted to describing a sin-
gle 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 indica-
tion 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 trans-
mission 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 destina- |
tion 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 play-
back or the client in recording), the source address MAY be
specified.
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This information may also be available through SDP. How-
ever, 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 pro-
vided, 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 10.13. 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 multi-
cast 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 [23] 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 hexideci-
mal 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.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
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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 sup-
ported by the server. In the case where the feature was specified via
the Proxy-Require field (Section 12.32), if there is a proxy on the
path between the client and the server, the proxy MUST insert a
response message with a status code of 551 (Option Not Supported).
See Section 12.32 for a usage example.
Unsupported = "Unsupported" ":" 1#option-tag
12.42 User-Agent
See [H14.43]
12.43 Vary
See [H14.44]
12.44 Via
See [H14.45].
12.45 WWW-Authenticate
See [H14.47].
13 Caching
In HTTP, response-request pairs are cached. RTSP differs signifi-
cantly in that respect. Responses are not cacheable, with the excep-
tion of the presentation description returned by DESCRIBE or included
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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 issu-
ing 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 possi-
bly 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 cur-
rently 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 addi-
tional 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 presen-
tation and all its streams, including the codecs that are available,
dynamic RTP payload types, the protocol stack, and content informa-
tion 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
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For purposes of this example, a container file is a storage entity in
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 con-
trol 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 descrip-
tion 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 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 ses-
sion 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
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CSeq: 91
Transport: RTP/AVP;multicast;destination=224.0.1.11;
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)>
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CR = <US-ASCII CR, carriage return (13)>
LF = <US-ASCII LF, linefeed (10)>
SP = <US-ASCII SP, space (32)>
HT = <US-ASCII HT, horizontal-tab (9)>
<"> = <US-ASCII double-quote mark (34)>
BACKSLASH = <US-ASCII backslash (92)>
CRLF = CR LF
LWS = [CRLF] 1*( SP | HT )
TEXT = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | BACKSLASH | <">
| "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">>
quoted-pair = BACKSLASH 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 authentica-
tion schemes, and thus should follow the same prescriptions
with regards to authentication. See chapter 15.1 of [2] for
client authentication issues, and chapter 15.2 of [2] for
issues regarding support for multiple authentication mecha-
nisms.
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Abuse of Server Log Information: RTSP and HTTP servers will presum-
ably have similar logging mechanisms, and thus should be
equally guarded in protecting the contents of those logs, thus
protecting the privacy of the users of the servers. See
[H15.1.1] 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 sensi-
tive 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 [H15.1.2] 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 por-
tions of the request URLs directly to file system calls. In
such cases, file systems SHOULD follow the precautions out-
lined in [H15.5], such as checking for ".." in path compo-
nents.
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 [H15.1] for further recommen-
dations.
Privacy Issues Connected to Accept Headers: Since may of the same
"Accept" headers exist in RTSP as in HTTP, the same caveats
outlined in [H15.1.4] with regards to their use should be fol-
lowed.
DNS Spoofing: Presumably, given the longer connection times typi-
cally associated to RTSP sessions relative to HTTP sessions,
RTSP client DNS optimizations should be less prevalent.
Nonetheless, the recommendations provided in [H15.3] 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 invali-
date resources over which they have no authority. ([H15.4])
In addition to the recommendations in the current HTTP specification
(RFC 2616 [26], as of this writing) and also of the previous RFC2068
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[2], 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-ini-
tiated 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 identi-
fiers which would affect unsuspecting clients. The server
SHOULD use a large, random and non-sequential session identi-
fier to minimize the possibility of this kind of attack.
Authentication: Servers SHOULD implement both basic and digest [6]
authentication. In environments requiring tighter security for
the control messages, transport layer mechanisms such as TLS
(RFC 2246 [27]) SHOULD be used.
Stream issues: RTSP only provides for stream control. Stream deliv-
ery 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.
17 IANA Considerations
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This section set up a number of registers for RTSP that should be
maintained by IANA. For each registry there is a description on what
it shall contain, what specification is needed when adding a entry
with IANA, and finally the entries that this document needs to regis-
ter. See also the section 1.5 "Extending RTSP".
The sections describing how to register an item uses some of the
requirements level described in RFC 2434 [29], namely " First Come,
First Served", "Specification Required", and "Standards Action".
A registration request to IANA MUST contain the following informa-
tion:
+ A name of the item to register according to the rules specified
by the intended registry.
+ Indication of who has change control over the option (for exam-
ple, IETF, ISO, ITU-T, other international standardization bod-
ies, 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 fil-
ing, a technical report, documented source code or a computer
manual;
+ For proprietary options, contact information (postal and email
address);
17.1 Option-tags
17.1.1 Description
When a client and server try to determine what part and functionality
of the RTSP specification and any future extensions that its counter
part implements there is need for a namespace. This registry con-
tains named entries representing certain functionality.
The usage of option-tags is explained in section 3.7 and 10.1.
17.1.2 Registering New Option Tags with IANA
The registering of option tags is done on a first come, first served
basis.
The name of the option MUST follow these rules: 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. Any
proprietary option SHOULD have as the first part of the name a vendor
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tag, which identifies the company/person.
17.1.3 Registered entries
The following options tags are in this specification defined and
hereby registered. The change control belongs to the Authors and the
IETF MMUSIC WG.
play-basic: The minimal implementation for playback operations
according to section D.
record-basic: The minimal implementation for record operations
according to section D.
play-setup: The use of teardown and setup in play state.
record-setup: The use of setup and teardown in record state.
17.2 RTSP Methods
17.2.1 Description
What a method is, is described in section 10. Extending the protocol
with new methods allow for totally new functionality.
17.2.2 Registering New Methods with IANA
A new method can only be registered through an IETF standards action.
The reason is that new methods may radically change the protocols
behavior and purpose.
A specification for a new RTSP method MUST consist of the following
items:
+ A method name which follows the BNF rules for methods.
+ A clear specification on what action and response a request with
the method will result in. Which directions the method is used,
C->S or S->C or both. How the use of headers, if any, modifies
the behavior and effect of the method.
+ A list or table specifying which of the registered headers that
are allowed to use with the method in request or/and response.
+ Describe how the method relates to network proxies.
17.2.3 Registered entries
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This specification, RFCXXXX, registers 12 methods: DESCRIBE,
ANNOUNCE, GET_PARAMETER, OPTIONS, PAUSE, PING, PLAY, RECORD, REDI-
RECT, SETUP, SET_PARAMETER, and TEARDOWN.
17.3 RTSP Headers
17.3.1 Description
By specifying new headers a method(s) can be enhanced in many differ-
ent ways. An unknown header will be ignored by the receiving entity.
If the new header is vital for a certain functionality, a option tag
for the functionality can be created and demanded to be used by the
counter-part with the inclusion of a Require header carrying the
option tag.
Unregistered headers SHALL have a name starting with "X-" to signal
that it is a experimental header.
17.3.2 Registering New Headers with IANA
A specification is required to register a header.
The specification MUST contain the following information:
+ The header name following the BNF definition.
+ A BNF specification of how information (if any) is carried in the
header.
+ A list or table specifying when the header may be used, encom-
passing all methods, their request or response, the direction
(C->S or S->C).
+ How the header shall be handled by proxies.
+ A description of the purpose of the header.
17.3.3 Registered entries
All headers specified in section 12 in RFC XXXX are to be registered.
17.4 Parameters
17.4.1 Description
A Parameter allow the counterpart to set something with the owner of
the parameter. Both the client and the server can have parameters.
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17.4.2 Registering New Parameters with IANA
Any Parameter is registered on a first come, first served basis. The
following rules apply for parameters:
+ The parameter name is a BNF token. The name SHOULD not be more
than 20 characters long. Any proprietary parameter should start
the name with a vendor tag, as clearly as possible identify the
company or person.
+ Any non proprietary parameter MUST in the form of BNF specify
what value types that are associated with the parameter.
17.4.3 Registered entries
For the moment no known parameters are defined in RFC XXXX.
A RTSP Protocol State Machine
The RTSP session state machine describe the behavior of the protocol
from RTSP session initialization through RTSP session termination.
State machine is defined on a per session basis which is uniquely
identified by the RTSP session identifier. The session may contain
zero or more media streams depending on state. If a single media
stream is part of the session it is in non-aggregated control. If two
or more is part of the session it is in aggregated control.
This state machine is one possible representation that helps explain
how the protocol works and when different requests are allowed. We
find it a reasonable representation but does not mandate it, and
other representations can be created.
A.1 States
The state machine contains five states, described below. For each
state there exist a table which shows which requests and events that
is allowed and if they will result in a state change.
Init: Initial state no session exist.
Ready-nm: Ready state without any medias.
Ready: Session is ready to start playing or recording.
Play: Session is playing, i.e. sending media stream data in the
direction S->C.
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Record: Session is recording, i.e. sending media stream data in the
direction C->S.
A.2 State variables
This representation of the state machine needs more than its state to
work. A small number of variables are also needed and is explained
below.
NRM: The number of media streams part of this session.
RP: Resume point, the point in the presentation time line at which
a request to continue will resume from. A time format for
variable is not mandated.
A.3 Abbreviations
To make the state tables more compact a number of abbreviations are
used, which are explained below.
PP: Pause Point, the point in the presentation time line at which
the presentation was paused.
Prs: Presentation, the complete multimedia presentation.
IFI: IF Implemented.
RedP: Redirect Point, the point in the presentation time line at
which a REDIRECT was specified to occur.
SES: Session.
A.4 State Tables
This section contains a table for each state. The table contains all
the requests and events that this state is allowed to act on. The
events which is method names are, unless noted, requests with the
given method in the direction client to server (C->S). In some cases
there exist one or more requisite. The response column tells what
type of response actions should be performed. Possible actions that
is requested for an event includes: response codes, e.g. 200, headers
that MUST be included in the response, setting of state variables, or
setting of other session related parameters. The new state column
tells which state the state machine shall change to.
The response to valid request meeting the requisites is normally a
2xx (SUCCESS) unless other noted in the response column. The excep-
tions shall be given a response according to the response column. If
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the request does not meet the requisite, is erroneous or some other
type of error occur the appropriate response code MUST be sent. If
the response code is a 4xx the session state is unchanged. A response
code of 3xx will result in that the session is ended and its state is
changed to Init. However there exist restrictions to when a 3xx
response may be used. A 5xx response SHALL not result in any change
of the session state, except if the error is not possible to recover
from. A unrecoverable error SHOULD result in ending of the session.
The server will timeout the session after the period of time speci-
fied in the SETUP response, if no activity from the client is
detected. Therefore there exist a timeout event for all states
except Init.
In the case that NRM=1 the presentation URL is equal to the media
URL. For NRM>1 the presentation URL MUST be other than any of the
medias that are part of the session. This applies to all states.
Event Prerequisite Response
-----------------------------------------------------------------
DESCRIBE Needs REDIRECT 3xx Redirect
DESCRIBE 200, Session description
OPTIONS Session ID 200, Reset session timeout timer
OPTIONS 200
SET_PARAMETER Valid parameter 200, change value of parameter
GET_PARAMETER Valid parameter 200, return value of parameter
ANNOUNCE C->S, IFI record.
ANNOUNCE S->C, Update SES descr.
Table 5: None state-machine changing events
The methods in Table 5 do not have any effect on the state machine or
the state variables. However some methods do change other session
related parameters, for example SET_PARAMETER which will set the
parameter(s) specified in its body.
The initial state of the state machine, see Table 6 can only be left
by processing a correct SETUP request. As seen in the table the two
state variables are also set by a correct request. This table also
shows that a correct SETUP can in some cases be redirected to another
URL and/or server by a 3xx response.
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Action Requisite New State Response
-------------------------------------------------
SETUP Ready NRM=1, RP=0.0
SETUP Needs Redirect Init 3xx Redirect
Table 6: State: Init
Action Requisite New State Response
--------------------------------------------------------------
SETUP Ready NRM=1,RP=0.0
SETUP Needs Redirect Init 3xx
TEARDOWN URL=* Init No session hdr.
Timeout Init
S->C:REDIRECT Range hdr Play Set RedP
S->C:REDIRECT no range hdr Init Stop Media Playout
RedP reached Init
Table 7: State: Ready-nm
The Ready-nm state has no media streams and therefore can't play or
record. This state exist so that all session related parameters and
resources can be kept while changing media stream(s). As seen in
Table 7 the operations are limited to setting up a new media or tear-
ing down the session. The established session can also be redirected
with the REDIRECT method.
In the Ready state, see Table 8, some of the actions are depending on
the number of media streams in the session, i.e. aggregated or non-
aggregated control. A setup request in the ready state can either add
one more media stream to the session or if the media stream (same
URL) already is part of the session change the transport parameters.
TEARDOWN is depending on both the request URI and the number of media
stream within the session. If the request URI is either * or the pre-
sentations URI the whole session is torn down. If a media URL is used
in the TEARDOWN request the session will remain and a session header
MUST be returned in the response. The number of media streams remain-
ing after tearing down a media stream determines the new state.
The Play state table, see Table 9, is the largest. The table contains
an number of request that has presentation URL as a prerequisite on
the request URL, this is due to the exclusion of non-aggregated
stream control in sessions with more than one media stream.
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Action Requisite New State Response
---------------------------------------------------------------------
SETUP New URL Ready NRM+=1
SETUP Setten up URL Ready Change transport param.
TEARDOWN URL=* Init No session hdr
TEARDOWN Prs URL,NRM>1 Init No session hdr
TEARDOWN md URL,NRM=1 Ready-nm Session hdr, NRM=0
TEARDOWN md URL,NRM>1 Ready Session hdr, NRM-=1
PLAY Prs URL, No range Play Play from RP
PLAY Prs URL, Range Play according to range
RECORD Record
S->C:REDIRECT Range hdr Ready Set RedP
S->C:REDIRECT no range hdr Init
Timeout Init
RedP reached Init
Table 8: State: Ready
Action Requisite New State Response
------------------------------------------------------------------------
PAUSE PrsURL,No range Ready Set RP to present point
PAUSE PrsURL,Range>now Play Set RP & PP to given point
PAUSE PrsURL,Range<=now Ready Set RP to present pos.
PP reached Ready RP = PP
End of media All media Play No action, RP = Invalid
End of media >=1 Media plays Play No action
End of range Play Set RP = End of range
SETUP New URL,IFI Play NRM+=1, 200, RTP-Info
SETUP New URL Play 501
SETUP Setuped URL Play Change transport param.
TEARDOWN URL=* Init No session hdr
TEARDOWN Prs URL,NRM>1 Init No session hdr
TEARDOWN md URL,NRM=1,IFI Ready-nm Session hdr
TEARDOWN md URL,NRM>1,IFI Play Session hdr
TEARDOWN md URL Play 501
S->C:REDIRECT Range hdr Play Set RedP
S->C:REDIRECT no range hdr Init Stop Media Playout
RedP reached Init Stop Media playout
Timeout Init
Table 9: State: Play
To avoid inconsistencies between the client and server, automatic
state transitions are avoided. This can be seen at for example "End
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of media" event when all media has finished playing, the session
still remain in Play state. An explicit PAUSE request must be sent to
change the state to Ready. It may appear that there exist two auto-
matic transitions in "RedP reached" and "PP reached", however they
are requested and acknowledge before they take place. The time at
which the transition will happen is known by looking at the range
header. If the client sends request close in time to these transi-
tions it must be prepared for getting error message as the state may
or may not have changed.
SETUP and TEARDOWN requests with media URLs in aggregated sessions
may not be handled by the server as it is optional functionality. Use
the service discovery mechanism with OPTIONS to find out in before-
hand if the server implements it. If the functionality is not imple-
mented but still tried by the client a "501 Not Implemented" response
SHALL be received.
Action Requisite New State Response
------------------------------------------------------------
PAUSE Ready
Out-of-disc Record Stop recording
TEARDOWN URL=* Init No session hdr
TEARDOWN Prs URL,NRM>1 Init No session hdr
TEARDOWN md URL,NRM=1,IFI Ready-nm Session hdr
TEARDOWN md URL,NRM>1,IFI Record Session hdr
TEARDOWN md URL Record 501
S->C:REDIRECT Range hdr Record Set RedP
S->C:REDIRECT w/o range hdr Init Stop Recording
RedP reached Init Stop Recording
Timeout Init
Table 10: State: Record
The Record state Table 10 has only one event which is unique for this
table, namely the "out-of-disc" event. This event will happen if the
recording server runs out of disc space. The state machine will
remain in the Record state but the server will not be able to perform
the actions related to the state.
Something is needed to signal the client the fact that the
server run out of disc space and not was capable of recording
the data sent by the client.
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B Interaction with RTP
RTSP allows media clients to control selected, non-contiguous sec-
tions of media presentations, rendering those streams with an RTP
media layer[23]. 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 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 12.34), 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 12.35) 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 12.33) header provides
the first sequence number of the next segment.
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C Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, RFC 2327 [24]) may be used to
describe streams or presentations in RTSP. This description is typi-
cally returned in reply to a DESCRIBE request on a URL from a server
to a client, received via HTTP from a server to a client, or sent in
an ANNOUNCE method from the client to the server.
This appendix describes how an SDP file determines the operation of
an RTSP session. SDP provides no mechanism by which a client can
distinguish, without human guidance, between several media streams to
be rendered simultaneously and a set of alternatives (e.g., two audio
streams spoken in different languages).
C.1 Definitions
The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in
SDP (RFC 2327 [24]):
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 con-
trolling 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, follow-
ing the rules and conventions set out in RFC 1808 [25]. Implementa-
tions 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.
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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 synchro-
nization. If the session is unicast, the port number serves as a rec-
ommendation 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.
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 pay-
load type is a static payload type from RFC 1890 [1], no other infor-
mation is required. In case it is a dynamic payload type, the media
attribute "rtpmap" is used to specify what the media is. The "encod-
ing 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 [24]). 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 3.4, 3.5 and
3.6.
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Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203
C.1.6 Time of Availability
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 12.22) 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 func-
tionality. However, it does so in a manner that would put
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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.
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 ini-
tialization 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 con-
trolled 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 rela-
tive, 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.
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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/
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://exam-
ple.com/movie/trackID=1 and rtsp://example.com/movie/trackID=2 to set
up the video and audio streams, respectively. The URL rtsp://exam-
ple.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 :
+ 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.
+ Include the following headers in requests: CSeq, Connection, Ses-
sion, Transport. If ANNOUNCE is implemented, the capability to
include headers Content-Language, Content-Encoding, Content-
Length, and Content-Type should be as well.
+ Parse and understand the following headers in responses: CSeq,
Connection, Session, Transport, Content-Language, Content-Encod-
ing, Content-Length, Content-Type. If RECORD is implemented, the
Location header must be understood as well. RTP-compliant imple-
mentations should also implement RTP-Info.
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+ 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.
+ Expect and respond to asynchronous requests from the server, such
as ANNOUNCE. This does not necessarily mean that it should imple-
ment 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.
+ Implement RTP/AVP/UDP as a valid transport.
+ Inclusion of the User-Agent header.
+ Understand SDP session descriptions as defined in Appendix C
+ Accept media initialization formats (such as SDP) from standard
input, command line, or other means appropriate to the operating
environment to act as a "helper application" for other applica-
tions (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 addi-
tionally be able to do the following:
+ generate the PAUSE request;
+ 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 fol-
lowing:
+ recognize the 401 (Unauthorized) status code;
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+ parse and include the WWW-Authenticate header;
+ implement Basic Authentication and Digest Authentication.
D.2 Server
A minimal server implementation MUST be able to do the following:
+ Implement the following methods: SETUP, TEARDOWN, OPTIONS and
either PLAY (for a minimal playback server) or RECORD (for a min-
imal recording server).
If RECORD is implemented, ANNOUNCE SHOULD be implemented as well.
+ Include the following headers in responses: Connection, Content-
Length, Content-Type, Content-Language, Content-Encoding, Trans-
port, 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.
+ 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 implemen-
tations and/or to be a "good citizen".
+ Implement RTP/AVP/UDP as a valid transport.
+ Inclusion of the Server header.
+ Implement the DESCRIBE method.
+ Generate SDP session descriptions as defined in Appendix C
There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for
which the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of
strict requirements.
D.2.1 Basic Playback
To support on-demand playback of media streams, the server MUST addi-
tionally be able to do the following:
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+ Recognize the Range header, and return an error if seeking is not
supported.
+ Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface
features, the following are highly recommended for on-demand media
servers:
+ Include and parse the Range header, with NPT units. Implementa-
tion of SMPTE units is recommended.
+ Include the length of the media presentation in the media ini-
tialization information.
+ Include mappings from data-specific timestamps to NPT. When RTP
is used, the rtptime portion of the RTP-Info field may be used to
map RTP timestamps to NPT.
Client implementations may use the presence of length informa-
tion 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 indi-
cator 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:
+ Generate the 401 (Unauthorized) status code when authentication
is required for the resource.
+ Parse and include the WWW-Authenticate header
+ Implement Basic Authentication and Digest Authentication
E Changes
Since RFC 2326, the following issues were addressed:
+ http://rtsp.org/bug448521 - URLs in Rtp-Info need to be quoted
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+ http://rtsp.org/bug448525 - Syntax for SSRC should be clarified
+ http://rtsp.org/bug461083 - Body w/o Content-Length clarification
+ http://rtsp.org/bug477407 - Transport BNF doesn't properly deal
with semicolon and comma
+ http://rtsp.org/bug477413 - Transport BNF: mode parameter issues
+ http://rtsp.org/bug477416 - BNF error section 3.6 NPT
+ http://rtsp.org/bug477421 - When to send response
+ http://rtsp.org/bug507347 - Removal of destination redirection
+ http://rtsp.org/bug477404 - Expanded the table to something use-
ful, proxy indications still missing.
+ http://rtsp.org/bug477419 - Started updating to rfc2616 by adding
public header. Section references in header chapter needs update.
+ http://rtsp.org/bug500803 - Rewritten the complete chapter on the
state machine. Needs review.
+ http://rtsp.org/bug513753 - Created a IANA section defining four
registries.
Note that this list does not reflect minor changes in wording or cor-
rection 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
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Robert Lanphier
RealNetworks
P.O. Box 91123
Seattle, WA 98111-9223
USA
electronic mail: robla@real.com
Magnus Westerlund
Ericsson AB, ERA/TVA/A
Torshamsgatan 23
SE-164 80 STOCKHOLM
SWEDEN
electronic mail: magnus.westerlund@ericsson.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 men-
tioned, the following individuals have contributed to this specifica-
tion:
Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
Bruce Butterfield, 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, Ema Patki, Alagu Periyannan, Colin Perkins,
Igor Plotnikov, Jonathan Sergent, Pinaki Shah, David Singer, Jeff
Smith, Alexander Sokolsky, Dale Stammen, John Francis Stracke, and
David Walker.
[1] H. Schulzrinne, "RTP profile for audio and video conferences with
minimal control," RFC 1890, Internet Engineering Task Force, Jan.
1996.
[2] R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. Berners-Lee,
"Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet Engi-
neering Task Force, Jan. 1997.
[3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst, "Internationaliza-
tion of the hypertext markup language," RFC 2070, Internet Engineer-
ing Task Force, Jan. 1997.
H. Schulzrinne et. al. [Page 99]
Internet Draft RTSP June 06, 2002
[4] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[5] ISO/IEC, "Information technology -- generic coding of moving pic-
tures and associated audio informaiton -- part 6: extension for digi-
tal storage media and control," Draft International Standard ISO
13818-6, International Organization for Standardization ISO/IEC
JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.
[6] J. Franks, P. Hallam-Baker, and J. Hostetler, "An extension to
HTTP: digest access authentication," RFC 2069, Internet Engineering
Task Force, Jan. 1997.
[7] J. Postel, "User datagram protocol," RFC STD 6, 768, Internet
Engineering Task Force, Aug. 1980.
[8] B. Hinden and C. Partridge, "Version 2 of the reliable data pro-
tocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr. 1990.
[9] J. Postel, "Transmission control protocol," RFC STD 7, 793,
Internet Engineering Task Force, Sept. 1981.
[10] 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.
[11] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
RFC 1961, Internet Engineering Task Force, June 1996.
[12] J. Miller, P. Resnick, and D. Singer, "Rating services and rat-
ing systems (and their machine readable descriptions)," Recommenda-
tion REC-PICS-services-961031, W3C (World Wide Web Consortium),
Boston, Massachusetts, Oct. 1996.
[13] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
distribution label syntax and communication protocols," Recommenda-
tion REC-PICS-labels-961031, W3C (World Wide Web Consortium), Boston,
Massachusetts, Oct. 1996.
[14] D. Crocker and P. Overell, "Augmented BNF for syntax specifica-
tions: ABNF," RFC 2234, Internet Engineering Task Force, Nov. 1997.
[15] B. Braden, "Requirements for internet hosts - application and
support," RFC STD 3, 1123, Internet Engineering Task Force, Oct.
1989.
H. Schulzrinne et. al. [Page 100]
Internet Draft RTSP June 06, 2002
[16] R. Elz, "A compact representation of IPv6 addresses," RFC 1924,
Internet Engineering Task Force, Apr. 1996.
[17] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
1994.
[18] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
2279, Internet Engineering Task Force, Jan. 1998.
[19] B. Braden, "T/TCP -- TCP extensions for transactions functional
specification," RFC 1644, Internet Engineering Task Force, July 1994.
[20] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
Reading, Massachusetts: Addison-Wesley, 1994.
[21] H. Schulzrinne, R. Lanphier, and A. Rao, "Real time streaming
protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
1998.
[22] T. Berners-Lee, R. Fielding, and L. Masinter, "Uniform resource
identifiers (URI): generic syntax," RFC 2396, Internet Engineering
Task Force, Aug. 1998.
[23] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
a transport protocol for real-time applications," RFC 1889, Internet
Engineering Task Force, Jan. 1996.
[24] M. Handley and V. Jacobson, "SDP: session description protocol,"
RFC 2327, Internet Engineering Task Force, Apr. 1998.
[25] R. Fielding, "Relative uniform resource locators," RFC 1808,
Internet Engineering Task Force, June 1995.
[26] R. Fielding, "Hypertext Transfer Protocol -- HTTP/1.1," RFC
2616, Internet Engineering Task Force, June 1999.
[27] T. Dierks, C. Allen, "The TLS Protocol, Version 1.0," RFC 2246,
Internet Engineering Task Force, Januari 1999.
[28] International Telecommunication Union, "Visual telephone systems
and equipment for local area networks which provide a non-guaranteed
quality of service," Recommendation H.323, Telecommunications Stan-
darization Sector of ITU, Geneva, Switzerland, May 1996.
[29] T. Narten, H. Alvestrand, "Guidelines for Writing an IANA Con-
siderations Section in RFCs," RFC2434, Internet Engineering Task
Force, October 1998.
H. Schulzrinne et. al. [Page 101]
Internet Draft RTSP June 06, 2002
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 docu-
ment 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 develop-
ing 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 MER-
CHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
H. Schulzrinne et. al. [Page 102]
Table of Contents
1 Introduction . . . . . . . . . . . . . . . . . . . . . . 3
1.1 Purpose . . . . . . . . . . . . . . . . . . . . . . . . . 3
1.2 Requirements . . . . . . . . . . . . . . . . . . . . . . 4
1.3 Terminology . . . . . . . . . . . . . . . . . . . . . . . 4
1.4 Protocol Properties . . . . . . . . . . . . . . . . . . . 6
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 . . . . . . . . . . . . . . . . . 11
3 Protocol Parameters . . . . . . . . . . . . . . . . . . . 12
3.1 RTSP Version . . . . . . . . . . . . . . . . . . . . . . 12
3.2 RTSP URL . . . . . . . . . . . . . . . . . . . . . . . . 12
3.3 Session Identifiers . . . . . . . . . . . . . . . . . . . 13
3.4 SMPTE Relative Timestamps . . . . . . . . . . . . . . . . 14
3.5 Normal Play Time . . . . . . . . . . . . . . . . . . . . 14
3.6 Absolute Time . . . . . . . . . . . . . . . . . . . . . . 15
3.7 Option Tags . . . . . . . . . . . . . . . . . . . . . . . 16
3.7.1 Registering New Option Tags with IANA . . . . . . . . . . 16
4 RTSP Message . . . . . . . . . . . . . . . . . . . . . . 17
4.1 Message Types . . . . . . . . . . . . . . . . . . . . . . 17
4.2 Message Headers . . . . . . . . . . . . . . . . . . . . . 17
4.3 Message Body . . . . . . . . . . . . . . . . . . . . . . 17
4.4 Message Length . . . . . . . . . . . . . . . . . . . . . 17
5 General Header Fields . . . . . . . . . . . . . . . . . . 18
6 Request . . . . . . . . . . . . . . . . . . . . . . . . . 18
6.1 Request Line . . . . . . . . . . . . . . . . . . . . . . 19
6.2 Request Header Fields . . . . . . . . . . . . . . . . . . 19
7 Response . . . . . . . . . . . . . . . . . . . . . . . . 20
7.1 Status-Line . . . . . . . . . . . . . . . . . . . . . . . 20
7.1.1 Status Code and Reason Phrase . . . . . . . . . . . . . . 21
7.1.2 Response Header Fields . . . . . . . . . . . . . . . . . 23
8 Entity . . . . . . . . . . . . . . . . . . . . . . . . . 25
8.1 Entity Header Fields . . . . . . . . . . . . . . . . . . 25
8.2 Entity Body . . . . . . . . . . . . . . . . . . . . . . . 25
9 Connections . . . . . . . . . . . . . . . . . . . . . . . 26
9.1 Pipelining . . . . . . . . . . . . . . . . . . . . . . . 26
9.2 Reliability and Acknowledgements . . . . . . . . . . . . 26
10 Method Definitions . . . . . . . . . . . . . . . . . . . 27
10.1 OPTIONS . . . . . . . . . . . . . . . . . . . . . . . . . 27
10.2 DESCRIBE . . . . . . . . . . . . . . . . . . . . . . . . 28
10.3 ANNOUNCE . . . . . . . . . . . . . . . . . . . . . . . . 30
10.4 SETUP . . . . . . . . . . . . . . . . . . . . . . . . . . 31
10.5 PLAY . . . . . . . . . . . . . . . . . . . . . . . . . . 32
10.6 PAUSE . . . . . . . . . . . . . . . . . . . . . . . . . . 34
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10.7 TEARDOWN . . . . . . . . . . . . . . . . . . . . . . . . 36
10.8 GET_PARAMETER . . . . . . . . . . . . . . . . . . . . . . 36
10.9 SET_PARAMETER . . . . . . . . . . . . . . . . . . . . . . 37
10.10 REDIRECT . . . . . . . . . . . . . . . . . . . . . . . . 38
10.11 RECORD . . . . . . . . . . . . . . . . . . . . . . . . . 39
10.12 PING . . . . . . . . . . . . . . . . . . . . . . . . . . 39
10.13 Embedded (Interleaved) Binary Data . . . . . . . . . . . 40
11 Status Code Definitions . . . . . . . . . . . . . . . . . 41
11.1 Success 2xx . . . . . . . . . . . . . . . . . . . . . . . 41
11.1.1 250 Low on Storage Space . . . . . . . . . . . . . . . . 41
11.2 Redirection 3xx . . . . . . . . . . . . . . . . . . . . . 41
11.3 Client Error 4xx . . . . . . . . . . . . . . . . . . . . 41
11.4 400 Bad Request . . . . . . . . . . . . . . . . . . . . . 42
11.4.1 405 Method Not Allowed . . . . . . . . . . . . . . . . . 42
11.4.2 451 Parameter Not Understood . . . . . . . . . . . . . . 42
11.4.3 452 reserved . . . . . . . . . . . . . . . . . . . . . . 42
11.4.4 453 Not Enough Bandwidth . . . . . . . . . . . . . . . . 42
11.4.5 454 Session Not Found . . . . . . . . . . . . . . . . . . 42
11.4.6 455 Method Not Valid in This State . . . . . . . . . . . 42
11.4.7 456 Header Field Not Valid for Resource . . . . . . . . . 42
11.4.8 457 Invalid Range . . . . . . . . . . . . . . . . . . . . 43
11.4.9 458 Parameter Is Read-Only . . . . . . . . . . . . . . . 43
11.4.10 459 Aggregate Operation Not Allowed . . . . . . . . . . . 43
11.4.11 460 Only Aggregate Operation Allowed . . . . . . . . . . 43
11.4.12 461 Unsupported Transport . . . . . . . . . . . . . . . . 43
11.4.13 462 Destination Unreachable . . . . . . . . . . . . . . . 43
11.5 Server Error 5xx . . . . . . . . . . . . . . . . . . . . 43
11.5.1 551 Option not supported . . . . . . . . . . . . . . . . 43
12 Header Field Definitions . . . . . . . . . . . . . . . . 43
12.1 Accept . . . . . . . . . . . . . . . . . . . . . . . . . 46
12.2 Accept-Encoding . . . . . . . . . . . . . . . . . . . . . 46
12.3 Accept-Language . . . . . . . . . . . . . . . . . . . . . 46
12.4 Accept-Ranges . . . . . . . . . . . . . . . . . . . . . . 46
12.5 Allow . . . . . . . . . . . . . . . . . . . . . . . . . . 48
12.6 Authorization . . . . . . . . . . . . . . . . . . . . . . 48
12.7 Bandwidth . . . . . . . . . . . . . . . . . . . . . . . . 49
12.8 Blocksize . . . . . . . . . . . . . . . . . . . . . . . . 49
12.9 Cache-Control . . . . . . . . . . . . . . . . . . . . . . 49
12.10 Connection . . . . . . . . . . . . . . . . . . . . . . . 51
12.11 Content-Base . . . . . . . . . . . . . . . . . . . . . . 51
12.12 Content-Encoding . . . . . . . . . . . . . . . . . . . . 52
12.13 Content-Language . . . . . . . . . . . . . . . . . . . . 52
12.14 Content-Length . . . . . . . . . . . . . . . . . . . . . 52
12.15 Content-Location . . . . . . . . . . . . . . . . . . . . 52
12.16 Content-Type . . . . . . . . . . . . . . . . . . . . . . 52
12.17 CSeq . . . . . . . . . . . . . . . . . . . . . . . . . . 52
12.18 Date . . . . . . . . . . . . . . . . . . . . . . . . . . 53
12.19 Expires . . . . . . . . . . . . . . . . . . . . . . . . . 53
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12.20 From . . . . . . . . . . . . . . . . . . . . . . . . . . 54
12.21 Host . . . . . . . . . . . . . . . . . . . . . . . . . . 54
12.22 If-Match . . . . . . . . . . . . . . . . . . . . . . . . 54
12.23 If-Modified-Since . . . . . . . . . . . . . . . . . . . . 54
12.24 Last-Modified . . . . . . . . . . . . . . . . . . . . . . 55
12.25 Location . . . . . . . . . . . . . . . . . . . . . . . . 55
12.26 Proxy-Authenticate . . . . . . . . . . . . . . . . . . . 55
12.27 Proxy-Require . . . . . . . . . . . . . . . . . . . . . . 55
12.28 Public . . . . . . . . . . . . . . . . . . . . . . . . . 55
12.29 Range . . . . . . . . . . . . . . . . . . . . . . . . . . 56
12.30 Referer . . . . . . . . . . . . . . . . . . . . . . . . . 57
12.31 Retry-After . . . . . . . . . . . . . . . . . . . . . . . 57
12.32 Require . . . . . . . . . . . . . . . . . . . . . . . . . 57
12.33 RTP-Info . . . . . . . . . . . . . . . . . . . . . . . . 58
12.34 Scale . . . . . . . . . . . . . . . . . . . . . . . . . . 59
12.35 Speed . . . . . . . . . . . . . . . . . . . . . . . . . . 60
12.36 Server . . . . . . . . . . . . . . . . . . . . . . . . . 61
12.37 Session . . . . . . . . . . . . . . . . . . . . . . . . . 61
12.38 Supported . . . . . . . . . . . . . . . . . . . . . . . . 61
12.39 Timestamp . . . . . . . . . . . . . . . . . . . . . . . . 62
12.40 Transport . . . . . . . . . . . . . . . . . . . . . . . . 62
12.41 Unsupported . . . . . . . . . . . . . . . . . . . . . . . 66
12.42 User-Agent . . . . . . . . . . . . . . . . . . . . . . . 66
12.43 Vary . . . . . . . . . . . . . . . . . . . . . . . . . . 66
12.44 Via . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
12.45 WWW-Authenticate . . . . . . . . . . . . . . . . . . . . 66
13 Caching . . . . . . . . . . . . . . . . . . . . . . . . . 66
14 Examples . . . . . . . . . . . . . . . . . . . . . . . . 67
14.1 Media on Demand (Unicast) . . . . . . . . . . . . . . . . 67
14.2 Streaming of a Container file . . . . . . . . . . . . . . 69
14.3 Single Stream Container Files . . . . . . . . . . . . . . 72
14.4 Live Media Presentation Using Multicast . . . . . . . . . 74
14.5 Recording . . . . . . . . . . . . . . . . . . . . . . . . 75
15 Syntax . . . . . . . . . . . . . . . . . . . . . . . . . 76
15.1 Base Syntax . . . . . . . . . . . . . . . . . . . . . . . 76
16 Security Considerations . . . . . . . . . . . . . . . . . 77
17 IANA Considerations . . . . . . . . . . . . . . . . . . . 79
17.1 Option-tags . . . . . . . . . . . . . . . . . . . . . . . 80
17.1.1 Description . . . . . . . . . . . . . . . . . . . . . . . 80
17.1.2 Registering New Option Tags with IANA . . . . . . . . . . 80
17.1.3 Registered entries . . . . . . . . . . . . . . . . . . . 81
17.2 RTSP Methods . . . . . . . . . . . . . . . . . . . . . . 81
17.2.1 Description . . . . . . . . . . . . . . . . . . . . . . . 81
17.2.2 Registering New Methods with IANA . . . . . . . . . . . . 81
17.2.3 Registered entries . . . . . . . . . . . . . . . . . . . 81
17.3 RTSP Headers . . . . . . . . . . . . . . . . . . . . . . 82
17.3.1 Description . . . . . . . . . . . . . . . . . . . . . . . 82
17.3.2 Registering New Headers with IANA . . . . . . . . . . . . 82
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17.3.3 Registered entries . . . . . . . . . . . . . . . . . . . 82
17.4 Parameters . . . . . . . . . . . . . . . . . . . . . . . 82
17.4.1 Description . . . . . . . . . . . . . . . . . . . . . . . 82
17.4.2 Registering New Parameters with IANA . . . . . . . . . . 83
17.4.3 Registered entries . . . . . . . . . . . . . . . . . . . 83
A RTSP Protocol State Machine . . . . . . . . . . . . . . . 83
A.1 States . . . . . . . . . . . . . . . . . . . . . . . . . 83
A.2 State variables . . . . . . . . . . . . . . . . . . . . . 84
A.3 Abbreviations . . . . . . . . . . . . . . . . . . . . . . 84
A.4 State Tables . . . . . . . . . . . . . . . . . . . . . . 84
B Interaction with RTP . . . . . . . . . . . . . . . . . . 89
C Use of SDP for RTSP Session Descriptions . . . . . . . . 90
C.1 Definitions . . . . . . . . . . . . . . . . . . . . . . . 90
C.1.1 Control URL . . . . . . . . . . . . . . . . . . . . . . . 90
C.1.2 Media Streams . . . . . . . . . . . . . . . . . . . . . . 91
C.1.3 Payload Type(s) . . . . . . . . . . . . . . . . . . . . . 91
C.1.4 Format-Specific Parameters . . . . . . . . . . . . . . . 91
C.1.5 Range of Presentation . . . . . . . . . . . . . . . . . . 91
C.1.6 Time of Availability . . . . . . . . . . . . . . . . . . 92
C.1.7 Connection Information . . . . . . . . . . . . . . . . . 92
C.1.8 Entity Tag . . . . . . . . . . . . . . . . . . . . . . . 92
C.2 Aggregate Control Not Available . . . . . . . . . . . . . 93
C.3 Aggregate Control Available . . . . . . . . . . . . . . . 93
D Minimal RTSP implementation . . . . . . . . . . . . . . . 94
D.1 Client . . . . . . . . . . . . . . . . . . . . . . . . . 94
D.1.1 Basic Playback . . . . . . . . . . . . . . . . . . . . . 95
D.1.2 Authentication-enabled . . . . . . . . . . . . . . . . . 95
D.2 Server . . . . . . . . . . . . . . . . . . . . . . . . . 96
D.2.1 Basic Playback . . . . . . . . . . . . . . . . . . . . . 96
D.2.2 Authentication-enabled . . . . . . . . . . . . . . . . . 97
E Changes . . . . . . . . . . . . . . . . . . . . . . . . . 97
F Author Addresses . . . . . . . . . . . . . . . . . . . . 98
G Acknowledgements . . . . . . . . . . . . . . . . . . . . 99
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