Internet Engineering Task Force MMUSIC WG
Internet Draft H. Schulzrinne
Columbia U.
A. Rao
Cisco
R. Lanphier
RealNetworks
M. Westerlund
Ericsson
A. Narasimhan
Sun
draft-ietf-mmusic-rfc2326bis-05.txt
October 27, 2003
Expires: April, 2004
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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Abstract
This memorandum is a revision of RFC 2326, which is currently a
Proposed Standard.
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery
sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and provide a means for choosing delivery
mechanisms based upon RTP (RFC 3550).
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1 Introduction
1.1 The Update of the RTSP Specification
This is the draft to an update of RTSP which is currently a proposed
standard defined in RFC 2326 [21]. Many flaws have been found in
RTSP since it was published. While this draft tries to address the
flaws, not all known issues have been resolved.
The goal of the current work on RTSP is to progress it to draft
standard status. Whether this is possible without first publishing
RTSP as a proposed standard depends on the changes necessary to make
the protocol work. The list of changes in chapter F indicates the
issues that have already been addressed. The currently open issues
are listed in chapter E.
There is also a list of reported bugs available at
"http://rtspspec.sourceforge.net". These bugs should be taken into
account when reading this specification. While a lot of these bugs
are addressed, not all are yet accounted for in this specification.
Input on the unresolved bugs and other issues can be sent via e-mail
to the MMUSIC WG's mailing list mmusic@ietf.org and the authors.
Take special notice of the following:
o The example section 15 has not yet been revised since the
changes to protocol have not been completed.
o The BNF chapter 16 has not been compiled completely.
o Not all of the contents of RFC 2326 are part of this draft.
In an attempt to prevent the draft from exploding in size, the
specification has been reduced and split. The content of this
draft is the core specification of the protocol. It contains
the general idea behind RTSP and the basic functionality
necessary to establish an on-demand play-back session. It also
contains the mechanisms for extending the protocol. Any other
functionality will be published as extension documents. Two
proposals exist at this time:
o NAT and FW traversal mechanisms for RTSP are described in a
document called "How to make Real-Time Streaming Protocol
(RTSP) traverse Network Address Translators (NAT) and interact
with Firewalls." [33].
o The MUTE extension [34] contains a proposal on adding
functionality to mute and unmute media streams in an
aggregated media session without affecting the time-line of
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the playback.
There have also been discussions about the following extensions to
RTSP:
o Transport security for RTSP messages (rtsps).
o Unreliable transport of RTSP messages (rtspu).
o The Record functionality.
o A text body type with suitable syntax for basic parameters to
be used in SET_PARAMETER, and GET_PARAMETER. Including IANA
registry within the defined name space.
o A RTSP MIB.
However, so far, they have not become concrete proposals.
1.2 Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls
single or several time-synchronized streams of continuous media such
as audio and video. Put simply, RTSP acts as a "network remote
control" for multimedia servers.
There is no notion of a RTSP connection in the protocol. Instead, a |
RTSP server maintains a session labelled by an identifier to |
associate groups of media streams and their states. A RTSP session is |
not tied to a transport-level connection such as a TCP connection. |
During a session, a client may open and close many reliable transport |
connections to the server to issue RTSP requests for that session.
This memorandum describes the use of RTSP over a reliable connection
based transport level protocol such as TCP. RTSP may be implemented
over an unreliable connectionless transport protocol such as UDP.
While nothing in RTSP precludes this, additional definition of this
problem area must be handled as an extension to the core
specification.
The mechanisms of RTSP's operation over UDP were left out
of this spec. because they were poorly defined in RFC 2336
[21] and the tradeoff in size and complexity of this spec.
for a small gain in a targeted problem space was not deemed
justifiable.
The set of streams to be controlled is defined by a presentation
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description. This memorandum does not define a format for the
presentation description. The streams controlled by RTSP may use RTP
[1] for their data transport, but the operation of RTSP does not
depend on the transport mechanism used to carry continuous media. The
protocol is intentionally similar in syntax and operation to HTTP/1.1
[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:
o RTSP introduces a number of new methods and has a different
protocol identifier.
o RTSP has the notion of a session built into the protocol.
o A RTSP server needs to maintain state by default in almost all
cases, as opposed to the stateless nature of HTTP.
o Both a RTSP server and client can issue requests.
o Data is usually carried out-of-band by a different protocol.
Session descriptions returned in a DESCRIBE response (see
Section 11.2) and interleaving of RTP with RTSP over TCP are
exceptions to this rule (see Section 11.11).
o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
8859-1, consistent with current HTML internationalization
efforts [3].
o The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1
[26] carries only the absolute path in the request and puts
the host name in a separate header field.
This makes "virtual hosting" easier, where a single
host with one IP address hosts several document trees.
The protocol supports the following operations:
Retrieval of media from media server: The client can request a
presentation description via HTTP or some other method. If
the presentation is being multicast, the presentation
description contains the multicast addresses and ports to
be used for the continuous media. If the presentation is
to be sent only to the client via unicast, the client
provides the destination for security reasons.
Invitation of a media server to a conference: A media server can
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be "invited" to join an existing conference to play back
media into the presentation. This mode is useful for
distributed teaching applications. Several parties in the
conference may take turns "pushing the remote control
buttons".
Addition of media to an existing presentation: Particularly for
live presentations, it is useful if the server can tell the
client about additional media becoming available.
RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1 [26].
1.3 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].
1.4 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 concept of controlling multiple streams
using a single timeline, generally maintained by the
server. A client, for example, uses aggregate control when
it issues a single play or pause message to simultaneously
control both the audio and video in a movie.
Aggregate control URI: The URI used in a RTSP request to refer
to and control an aggregated session. It normally, but not
always, corresponds to the presentation URI specified in
the session description. See Section 11.3 for more
information.
Conference: a multiparty, multimedia presentation, where "multi"
implies greater than or equal to one.
Client: The client requests media service 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
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the protocol.
Continuous media: Data where there is a timing relationship
between source and sink; that is, the sink must reproduce
the timing relationship that existed at the source. The
most common examples of continuous media are audio and
motion video. Continuous media can be real-time
(interactive), where there is a "tight" timing relationship
between source and sink, or streaming (playback), where the
relationship is less strict.
Entity: The information transferred as the payload of a request
or response. An entity consists of meta-information in the
form of entity-header fields and content in the form of an
entity-body, as described in Section 8.
Feature-tag: A tag representing a certain set of functionality,
i.e. a feature.
Media initialization: Datatype/codec specific initialization.
This includes such things as clockrates, color tables, etc.
Any transport-independent information which is required by
a client for playback of a media stream occurs in the media
initialization phase of stream setup.
Media parameter: Parameter specific to a media type that may be
changed before or during stream playback.
Media server: The server providing playback services for one or
more media streams. Different media streams within a
presentation may originate from different media servers. A
media server may reside on the same or a different host as
the web server the presentation is invoked from.
Media server indirection: Redirection of a media client to a
different media server.
(Media) stream: A single media instance, e.g., an audio stream
or a video stream as well as a single whiteboard or shared
application group. When using RTP, a stream consists of all
RTP and RTCP packets created by a source within an RTP
session. This is equivalent to the definition of a DSM-CC
stream([5]).
Message: The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined
in Section 16 and transmitted via a connection or a
connectionless protocol.
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Non-Aggregated Control: Control of a single media stream. Only
possible in RTSP sessions with a single media.
Participant: Member of a conference. A participant may be a
machine, e.g., a playback server.
Presentation: A set of one or more streams presented to the
client as a complete media feed, using a presentation
description as defined below. In most cases in the RTSP
context, this implies aggregate control of those streams,
but does not have to.
Presentation description: A presentation description contains
information about one or more media streams within a
presentation, such as the set of encodings, network
addresses and information about the content. Other IETF
protocols such as SDP (RFC 2327 [24]) use the term
"session" for a live presentation. The presentation
description may take several different formats, including
but not limited to the session description format SDP.
Response: A RTSP response. If an HTTP response is meant, that is
indicated explicitly.
Request: A RTSP request. If an HTTP request is meant, that is
indicated explicitly.
RTSP session: A stateful abstraction upon which the main control
methods of RTSP operate. A RTSP session is a server entity;
it is created, maintained and destroyed by the server. It
is established by a RTSP server upon the completion of a
successful SETUP request (when 200 OK response is sent) and
is labelled by a session identifier at that time. The
session exists until timed out by the server or explicitly
removed by a TEARDOWN request. A RTSP session is also a
stateful entity; a RTSP server maintains an explicit
session state machine (see Appendix A) where most state
transitions are triggered by client requests. The existence
of a session implies the existence of state about the
session's media streams and their respective transport
mechanisms. A given session can have zero or more media
streams associated with it. A RTSP server uses the session
to aggregate control over multiple media streams.
Transport initialization: The negotiation of transport
information (e.g., port numbers, transport protocols)
between the client and the server.
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1.5 Protocol Properties
RTSP has the following properties:
Extendable: New methods and parameters can be easily added to
RTSP.
Easy to parse: RTSP can be parsed by standard HTTP or MIME
parsers.
Secure: RTSP re-uses web security mechanisms, either at the
transport level (TLS, RFC 2246 [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 does not preclude the use of an
unreliable datagram protocol (UDP) (RFC 768 [7]), a
reliable datagram protocol (RDP, RFC 1151, not widely used
[8]) or a reliable stream protocol such as TCP (RFC 793
[9]) as it implements application-level reliability. The
use of a connectionless datagram protocol such as UDP or
RDP requires additional definition that may be provided as
extensions to the core RTSP specification.
Multi-server capable: Each media stream within a presentation
can reside on a different server. The client automatically
establishes several concurrent control sessions with the
different media servers. Media synchronization is
performed at the transport level.
Separation of stream control and conference initiation: Stream
control 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
presentation description must contain at least one RTSP
URI.
Proxy and firewall friendly: The protocol should be readily
handled by both application and transport-layer (SOCKS
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[11]) firewalls. A firewall may need to understand the
SETUP method to open a "hole" for the UDP media stream.
HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so
that the existing infrastructure can be reused. This
infrastructure includes PICS (Platform for Internet Content
Selection [12,13]) for associating labels with content.
However, RTSP does not just add methods to HTTP since the
controlling continuous media requires server state in most
cases.
Appropriate server control: If a client can start a stream, it
must be able to stop a stream. Servers should not start
streaming to clients in such a way that clients cannot stop
the stream.
Transport negotiation: The client can negotiate the transport
method prior to actually needing to process a continuous
media stream.
Capability negotiation: If basic features are disabled, there
must be some clean mechanism for the client to determine
which methods are not going to be implemented. This allows
clients to present the appropriate user interface. For
example, if seeking is not allowed, the user interface must
be able to disallow moving a sliding position indicator.
An earlier requirement in RTSP was multi-client capability.
However, it was determined that a better approach was to
make sure that the protocol is easily extensible to the
multi-client scenario. Stream identifiers can be used by
several control streams, so that "passing the remote" would
be possible. The protocol would not address how several
clients negotiate access; this is left to either a "social
protocol" or some other floor control mechanism.
1.6 Extending RTSP
Since not all media servers have the same functionality, media
servers by necessity will support different sets of requests. For
example:
o A server may not be capable of seeking (absolute positioning)
if it is to support live events only.
o Some servers may not support setting stream parameters and
thus not support GET_PARAMETER and SET_PARAMETER.
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A server SHOULD implement all header fields described in Section 13.
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.
RTSP can be extended in three ways, listed here in order of the
magnitude of changes supported:
o Existing methods can be extended with new parameters, as long
as these parameters can be safely ignored by the recipient.
(This is equivalent to adding new parameters to an HTML tag.)
If the client needs negative acknowledgement when a method
extension is not supported, a tag corresponding to the
extension may be added in the Require: field (see Section
13.32).
o New methods can be added. If the recipient of the message does
not understand the request, it responds with error code 501
(Not Implemented) and the sender should not attempt to use
this method again. A client may also use the OPTIONS method
to inquire about methods supported by the server. The server
SHOULD list the methods it supports using the Public response
header.
o A new version of the protocol can be defined, allowing almost
all aspects (except the position of the protocol version
number) to change.
The basic capability discovery mechanism can be used to both discover
support for a certain feature and to ensure that a feature is
available when performing a request. For detailed explanation of this
see chapter 10.
1.7 Overall Operation
Each presentation and media stream may be identified by a RTSP URL.
The overall presentation and the properties of the media the
presentation is made up of are defined by a presentation description
file, the format of which is outside the scope of this specification.
The presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored
on the media server.
For the purposes of this specification, a presentation description is
assumed to describe one or more presentations, each of which
maintains a common time axis. For simplicity of exposition and
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without loss of generality, it is assumed that the presentation
description contains exactly one such presentation. A presentation
may contain several media streams.
The presentation description file contains a description of the media
streams making up the presentation, including their encodings,
language, and other parameters that enable the client to choose the
most appropriate combination of media. In this presentation
description, each media stream that is individually controllable by
RTSP is identified by a RTSP URL, which points to the media server
handling that particular media stream and names the stream stored on
that server. Several media streams can be located on different
servers; for example, audio and video streams can be split across
servers for load sharing. The description also enumerates which
transport methods the server is capable of.
Besides the media parameters, the network destination address and
port need to be determined. Several modes of operation can be
distinguished:
Unicast: The media is transmitted to the source of the RTSP
request, with the port number chosen by the client.
Alternatively, the media is transmitted on the same
reliable stream as RTSP.
Multicast, server chooses address: The media server picks the
multicast address and port. This is the typical case for a
live or near-media-on-demand transmission.
Multicast, client chooses address: If the server is to
participate in an existing multicast conference, the
multicast address, port and encryption key are given by the
conference description, established by means outside the
scope of this specification.
1.8 RTSP States
RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel. For example, RTSP control may
occur on a TCP connection while the data flows via UDP. Thus, data
delivery continues even if no RTSP requests are received by the media
server. Also, during its lifetime, a single media stream may be
controlled by RTSP requests issued sequentially on different TCP
connections. Therefore, the server needs to maintain "session state"
to be able to correlate RTSP requests with a stream. The state
transitions are described in Appendix A.
Many methods in RTSP do not contribute to state. However, the
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following play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, PING
and TEARDOWN.
SETUP: Causes the server to allocate resources for a stream and
create a RTSP session.
PLAY: Starts data transmission on a stream allocated via SETUP.
PAUSE: Temporarily halts a stream without freeing server
resources.
REDIRECT: Indicates that the session should be moved to new
server / location
PING: Prevents the identified session from being timed out.
TEARDOWN: Frees resources associated with the stream. The RTSP
session ceases to exist on the server.
RTSP methods that contribute to state use the Session header field
(Section 13.37) to identify the RTSP session whose state is being
manipulated. The server generates session identifiers in response to
SETUP requests (Section 11.3).
1.9 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may
interact with HTTP in that the initial contact with streaming content
is often to be made through a web page. The current protocol
specification aims to allow different hand-off points between a web
server and the media server implementing RTSP. For example, the
presentation description can be retrieved using HTTP or RTSP, which
reduces roundtrips in web-browser-based scenarios, yet also allows
for standalone RTSP servers and clients which do not rely on HTTP at
all. However, RTSP differs fundamentally from HTTP in that most data
delivery takes place out-of-band in a different protocol. HTTP is an
asymmetric protocol where the client issues requests and the server
responds. In RTSP, both the media client and media server can issue
requests. RTSP requests are also stateful; they may set parameters
and continue to control a media stream long after the request has
been acknowledged.
Re-using HTTP functionality has advantages in at least two
areas, namely security and proxies. The requirements are
very similar, so having the ability to adopt HTTP work on
caches, proxies and authentication is valuable.
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RTSP assumes the existence of a presentation description format that
can express both static and temporal properties of a presentation
containing several media streams. Session Description Protocol (SDP)
[24] is generally the format of choice; however, RTSP is not bound to
it. For data delivery, most real-time media will use RTP as a
transport protocol. While RTSP works well with RTP, it is not tied to
RTP.
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
difference that this RTSP specification maintains the "#" notation
for comma-separated lists from [H2.1].
In this draft, we use indented and smaller-type paragraphs to provide
background and motivation. This is intended to give readers who were
not involved with the formulation of the specification an
understanding of why things are the way that they are in RTSP.
3 Protocol Parameters
3.1 RTSP Version
HTTP Specification Section [H3.1] applies, with HTTP replaced by
RTSP. This specification defines version 1.0 of RTSP.
3.2 RTSP URL
The "rtsp", "rtsps" and "rtspu" schemes are used to refer to network
resources via the RTSP protocol. This section defines the scheme-
specific syntax and semantics for RTSP URLs. The RTSP URL is case
sensitive.
rtsp_URL = ( "rtsp:" / "rtspu:" / "rtsps:" )
"//" host [ ":" port ] [ abs_path [ "?" query ]]
host = As defined by RFC 2732 [30]
abs_path = As defined by RFC 2396 [22]
port = *DIGIT
query = As defined by RFC 2396 [22]
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Note that fragment and query identifiers do not have a
well-defined meaning at this time, with the interpretation
left to the RTSP server.
The scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtspu
identifies an unreliable protocol (within the Internet, UDP). The
scheme rtsps identifies a reliable transport using secure transport,
perhaps TLS [27]. The rtspu and rtsps is not defined in this
specification, and are for future extensions of the protocol to
define.
If the port is empty or not given, port 554 SHALL be 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]). Note: Using qualified domain names in any URL is
one requirement for making it possible for RFC 2326 implementations
of RTSP to use IPv6. This specification is updated to allow for
literal IPv6 addresses in RTSP URLs using the host specification in
RFC 2732 [30].
A presentation or a stream is identified by a textual media
identifier, using the character set and escape conventions [H3.2] of
URLs (RFC 2396 [22]). URLs may refer to a stream or an aggregate of
streams, i.e., a presentation. Accordingly, requests described in
Section 11 can apply to either the whole presentation or an
individual stream within the presentation. Note that some request
methods can only be applied to streams, not presentations and vice
versa.
For example, the RTSP URL:
rtsp://media.example.com:554/twister/audiotrack
identifies the audio stream within the presentation "twister", which
can be controlled via RTSP requests issued over a TCP connection to
port 554 of host media.example.com
Also, the RTSP URL:
rtsp://media.example.com:554/twister
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identifies the presentation "twister", which may be composed of audio
and video streams.
This does not imply a standard way to reference streams in
URLs. The presentation description defines the hierarchical
relationships in the presentation and the URLs for the
individual streams. A presentation description may name a
stream "a.mov" and the whole presentation "b.mov".
The path components of the RTSP URL are opaque to the client and do
not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be
used with non-RTSP media control protocols simply by
replacing the scheme in the URL.
3.3 Session Identifiers
Session identifiers are strings of any arbitrary length. A session
identifier MUST be chosen randomly and MUST be at least eight
characters long to make guessing it more difficult. (See Section 17.)
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 )
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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
relative to the beginning of the presentation, not to be confused
with the Network Time Protocol (NTP). 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 only be used for live events,
and SHALL NOT be used for on-demand content.
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
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npt-ss = 1*2DIGIT ; 0-59
Examples:
npt=123.45-125
npt=12:05:35.3-
npt=now-
The syntax conforms to ISO 8601. The npt-sec notation is
optimized for automatic generation, the ntp-hhmmss notation
for consumption by human readers. The "now" constant allows
clients to request to receive the live feed rather than the
stored or time-delayed version. This is needed since
neither absolute time nor zero time are appropriate for
this case.
3.6 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 >
utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
fraction = 1*DIGIT
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC:
19961108T143720.25Z
3.7 Feature-tags
Feature-tags are unique identifiers used to designate features in
RTSP. These tags are used in Require (Section 13.32), Proxy-Require
(Section 13.27), Unsupported (Section 13.41), and Supported (Section
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13.38) header fields.
Syntax:
feature-tag = token
Feature tag needs to indicate if they apply to servers only, proxies |
only, or both server and proxies.
The creator of a new RTSP feature-tag should either prefix the
feature-tag with a reverse domain name (e.g., "com.foo.mynewfeature"
is an apt name for a feature whose inventor can be reached at
"foo.com"), or register the new feature-tag with the Internet
Assigned Numbers Authority (IANA), see IANA Section 18.
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
themselves as line terminators.
Text-based protocols make it easier to add optional
parameters in a self-describing manner. Since the number of
parameters and the frequency of commands is low, processing
efficiency is not a concern. Text-based protocols, if done
carefully, also allow easy implementation of research
prototypes in scripting languages such as Tcl, Visual Basic
and Perl.
The 10646 character set avoids tricky character set switching, but is
invisible to the application as long as US-ASCII is being used. This
is also the encoding used for RTCP. ISO 8859-1 translates directly
into Unicode with a high-order octet of zero. ISO 8859-1 characters
with the most-significant bit set are represented as 1100001x
10xxxxxx. (See RFC 2279 [18])
RTSP messages can be carried over any lower-layer transport protocol
that is 8-bit clean. RTSP messages are vulnerable to bit errors and
SHOULD NOT be subjected to them.
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.
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4.1 Message Types
See [H4.1].
4.2 Message Headers
See [H4.2].
4.3 Message Body
See [H4.3]
4.4 Message Length
When a message body is included with a message, the length of that
body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message body
(such as the 1xx, 204, and 304 responses) is always
terminated by the first empty line after the header fields,
regardless of the entity-header fields present in the
message. (Note: An empty line consists of only CRLF.)
2. If a Content-Length header field (section 13.14) 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
Content-Length header field.
Given the moderate length of presentation descriptions
returned, the server should always be able to determine its
length, even if it is generated dynamically, making the
chunked transfer encoding unnecessary.
5 General Header Fields
See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade,
and Warning headers are not defined. RTSP further defines the CSeq,
and Timestamp:
general-header = Cache-Control ; Section 13.9
/ Connection ; Section 13.10
/ CSeq ; Section 13.17
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/ Date ; Section 13.18
/ Timestamp ; Section 13.39
/ Via ; Section 13.44
6 Request
A request message from a client to a server or vice versa includes,
within the first line of that message, the method to be applied to
the resource, the identifier of the resource, and the protocol
version in use.
Request = Request-Line ; Section 6.1
*( general-header ; Section 5
/ request-header ; Section 6.2
/ entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
6.1 Request Line
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Method = "DESCRIBE" ; Section 11.2
/ "GET_PARAMETER" ; Section 11.7
/ "OPTIONS" ; Section 11.1
/ "PAUSE" ; Section 11.5
/ "PLAY" ; Section 11.4
/ "PING" ; Section 11.10
/ "REDIRECT" ; Section 11.9
/ "SETUP" ; Section 11.3
/ "SET_PARAMETER" ; Section 11.8
/ "TEARDOWN" ; Section 11.6
/ extension-method
extension-method = token
Request-URI = "*" / absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
6.2 Request Header Fields
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request-header = Accept ; Section 13.1
/ Accept-Encoding ; Section 13.2
/ Accept-Language ; Section 13.3
/ Authorization ; Section 13.6
/ Bandwidth ; Section 13.7
/ Blocksize ; Section 13.8
/ From ; Section 13.20
/ If-Modified-Since ; Section 13.23
/ Proxy-Require ; Section 13.27
/ Range ; Section 13.29
/ Referer ; Section 13.30
/ Require ; Section 13.32
/ Scale ; Section 13.34
/ Session ; Section 13.37
/ Speed ; Section 13.35
/ Supported ; Section 13.38
/ Transport ; Section 13.40
/ User-Agent ; Section 13.42
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 or proxy itself,
and is only allowed when the method used does not necessarily apply
to a resource.
One example would be as follows:
OPTIONS * RTSP/1.0
An OPTIONS in this form will determine the capabilities of the server
or the proxy that first receives the request. If one needs to address
the server explicitly, then one should use an absolute URL with the
server's address.
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OPTIONS rtsp://example.com RTSP/1.0
7 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version.
Also, RTSP defines additional status codes and does not define some
HTTP codes. The valid response codes and the methods they can be used
with are defined in Table 1.
After receiving and interpreting a request message, the recipient
responds with an RTSP response message.
Response = Status-Line ; Section 7.1
*( general-header ; Section 5
/ response-header ; Section 7.1.2
/ entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
7.1 Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version followed by a numeric status code, and the
textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
7.1.1 Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully
defined in Section 12. 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:
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o 1xx: Informational - Request received, continuing process
o 2xx: Success - The action was successfully received,
understood, and accepted
o 3rr: Redirection - Further action must be taken in order to
complete the request
o 4xx: Client Error - The request contains bad syntax or cannot
be fulfilled
o 5xx: Server Error - The server failed to fulfill an apparently
valid request
The individual values of the numeric status codes defined for
RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
presented below. The reason phrases listed here are only recommended
-- they may be replaced by local equivalents without affecting the
protocol. Note that RTSP adopts most HTTP/1.1 [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
/ "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
/ "350" ; Going Away
/ "351" ; Load Balancing
/ "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
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/ "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
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
RTSP status codes are extensible. RTSP applications are not required
to understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if an
unrecognized status code of 431 is received by the client, it can
safely assume that there was something wrong with its request and
treat the response as if it had received a 400 status code. In such
cases, user agents SHOULD present to the user the entity returned
with the response, since that entity is likely to include human-
readable information which will explain the unusual status.
7.1.2 Response Header Fields
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
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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 = Accept-Ranges ; Section
13.4
/ Location ; Section 13.25
/ Proxy-Authenticate ; Section 13.26
/ Public ; Section 13.28
/ Range ; Section 13.29
/ Retry-After ; Section 13.31
/ RTP-Info ; Section 13.33
/ Scale ; Section 13.34
/ Session ; Section 13.37
/ Server ; Section 13.36
/ Speed ; Section 13.35
/ Transport ; Section 13.40
/ Unsupported ; Section 13.41
/ Vary ; Section 13.43
/ WWW-Authenticate ; Section 13.45
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as
entity-header fields.
8 Entity
Request and Response messages MAY transfer an entity if not otherwise
restricted by the request method or response status code. An entity
consists of entity-header fields and an entity-body, although some
responses will only include the entity-headers.
In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity.
8.1 Entity Header Fields
Entity-header fields define optional meta-information about the
entity-body or, if no body is present, about the resource identified
by the request.
entity-header = Allow ; Section 13.5
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Code Reason Method
_______________________________________________________
100 Continue all
_______________________________________________________
200 OK all
201 Created RECORD
250 Low on Storage Space RECORD
_______________________________________________________
300 Multiple Choices all
301 Moved Permanently all
302 Found all
303 See Other all
305 Use Proxy all
350 Going Away all
351 Load Balancing 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 SET_PARAMETER
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, PAUSE
458 Parameter Is Read-Only SET_PARAMETER
459 Aggregate Operation Not Allowed all
460 Only Aggregate Operation Allowed all
461 Unsupported Transport all
462 Destination Unreachable all
_______________________________________________________
500 Internal Server Error all
501 Not Implemented all
502 Bad Gateway all
503 Service Unavailable all
504 Gateway Timeout all
505 RTSP Version Not Supported all
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Table 1: Status codes and their usage with RTSP methods
/ Content-Base ; Section 13.11
/ Content-Encoding ; Section 13.12
/ Content-Language ; Section 13.13
/ Content-Length ; Section 13.14
/ Content-Location ; Section 13.15
/ Content-Type ; Section 13.16
/ Expires ; Section 13.19
/ Last-Modified ; Section 13.24
/ 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] with the addition that a RTSP message with an entity body
MUST include a Content-Type header.
9 Connections
RTSP requests can be transmitted in several different ways:
o persistent transport connections used for several request-
response transactions;
o one connection per request/response transaction;
o connectionless mode.
The type of transport connection is defined by the RTSP URI (Section
3.2). For the scheme "rtsp", a connection is assumed, while the
scheme "rtspu" calls for RTSP requests to be sent without setting up
a connection.
Unlike HTTP, RTSP allows the media server to send requests to the
media client. However, this is only supported for persistent
connections, as the media server otherwise has no reliable way of
reaching the client. Also, this is the only way that requests from
media server to client are likely to traverse firewalls.
9.1 Pipelining
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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 |
The transmission of RTSP over UDP was optionally to implement and |
specified in RFC 2326. However that definition was not satisfactory |
for interoperable implementations. Due to lack of interest, this |
specification does not specify how RTSP over UDP shall be |
implemented. However to maintain backwards compatibility in the |
message format certain RTSP headers must be maintained. These |
mechanism are described below. The next section Unreliable Transport |
(section 9.3) provides documentation of certain features that are |
necessary for transport protocols like UDP. |
Any RTSP request according to this specification SHALL NOT be sent to |
a multicast address. Any RTSP request SHALL be acknowledged. If a |
reliable transport protocol is used to carry RTSP, requests MUST NOT |
be retransmitted; the RTSP application MUST instead rely on the |
underlying transport to provide reliability. |
If both the underlying reliable transport such as TCP and |
the RTSP application retransmit requests, it is possible |
that each packet loss results in two retransmissions. The |
receiver cannot typically take advantage of the |
application-layer retransmission since the transport stack |
will not deliver the application-layer retransmission |
before the first attempt has reached the receiver. If the |
packet loss is caused by congestion, multiple |
retransmissions at different layers will exacerbate the |
congestion. |
Each request carries a sequence number in the CSeq header (Section |
13.17), which MUST be incremented by one for each distinct request |
transmitted to the destination end-point. The initial sequence |
number MAY be chosen arbitrary, but is RECOMMENDED to begin with 0. |
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). |
9.3 Unreliable Transport |
This section provides some information to future specification of |
RTSP over unreliable transport. |
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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 RTSP is used over a small-RTT LAN, standard procedures for |
optimizing initial TCP round trip estimates, such as those used in |
T/TCP (RFC 1644) [19], can be beneficial. |
The Timestamp header (Section 13.39) is used to avoid the |
retransmission ambiguity problem [20] and obviates the need for |
Karn's algorithm. |
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). |
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. |
The default port for the RTSP server is 554 for UDP.
9.4 The usage of connections
Systems implementing RTSP MUST support carrying RTSP over TCP. The |
default port for the RTSP server is 554 for TCP. A number of RTSP |
packets destined for the same control end point may be 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 (entity). |
Otherwise, an RTSP packet is terminated with an empty line |
immediately following the last message header.
TCP can be used for both persistent connections and for one message
exchange per connection, as presented above. This section gives
further rules and recommendations on how to handle these connections
so maximum interoperability and flexibility can be achieved.
A server SHALL handle both persistent connections and one
request/response transaction per connection. A persistent connection
MAY be used for all transactions between the server and client,
including messages to multiple RTSP sessions. However the persistent
connection MAY also be closed after a few message exchanges, e.g. the
initial setup and play command in a session. Later when the client
wishes to send a new request, e.g. pause, to the session a new
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connection is opened. This connection may either be for a single
message exchange or can be kept open for several messages, i.e.
persistent.
A major motivation for allowing non-persistent connections are that
they ensure fault tolerance. A second one is to allow for application
layer mobility. A server and client supporting non-persistent
connection can survive a loss of a TCP connection, e.g. due to a NAT
timeout. When the client has discovered that the TCP connection has
been lost, it can set up a new one when there is need to communicate.
The client MAY close the connection at any time when no outstanding
request/response transactions exist. The server SHOULD NOT close the
connection unless at least one RTSP session timeout period has passed
without data traffic. A server MUST NOT initiate a close of a
connection directly after responding to a TEARDOWN request for the
whole session. A server MUST NOT close the connection as a result of
responding to a request with an error code. Doing this would prevent
or result in extra overhead for the client when testing advanced or
special types of requests.
The client SHOULD NOT have more than one connection to the server at
any given point. If a client or proxy handles multiple RTSP sessions
on the same server, it is RECOMMENDED to use only a single
connection.
Older services which was implemented according to RFC 2326 sometimes
requires the client to use persistent connection. The client closing
the connection may result in that the server removes the session. To
achieve interoperability with old servers any client is strongly
RECOMMENDED to use persistent connections.
A Client is also strongly RECOMMENDED to use persistent connections
as it allows the server to send request to the client. In cases
where no connection exist between the server and the client, this may
cause the server to be forced to drop the RTSP session without
notifying the client why,due to the lack of signalling channel. An
example of such a case is when the server desires to send a REDIRECT
request for a RTSP session to the client.
A server implemented according to this specification MUST respond
that it supports the "play.basic" feature-tag above. A client MAY
send a request including the Supported header in a request to
determine support of non-persistent connections. A server supporting
non-persistent connections will return the "play.basic" feature-tag
in its response. If the client receives the feature-tag in the
response, it can be certain that the server handles non-persistent
connections.
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9.5 Use of IPv6
This specification has been updated so that it supports IPv6.
However this support was not present in RFC 2326 therefore some
interoperability issues exist. A RFC 2326 implementation can support
IPv6 as long as no explicit IPv6 addresses are used within RTSP
messages. This require that any RTSP URL pointing at a IPv6 host must
use fully qualified domain name and not a IPv6 address. Further the
Transport header must not use the parameters source and destination.
Implementations according to this specification MUST understand IPv6
addresses in URLs, and headers. By this requirement the feature-tag
"play.basic" can be used to determine that a server or client is
capable of handling IPv6 within RTSP.
10 Capability Handling
This chapter describes the capability handling mechanism available in
RTSP which allows RTSP to be extended. Extensions to this version of
the protocol are basically done in two ways. First, new headers can
be added. Secondly, new methods can be added. The capability handling
mechanism is designed to handle these two cases.
When a method is added the involved parties can use the OPTIONS
method to discover if it is supported. This is done by issuing a
OPTIONS request to the other party. Depending on the URL it will
either apply in regards to a certain media resource, the whole server
in general, or simply the next hop. The OPTIONS response will contain
a Public header which declares all methods supported for the
indicated resource.
It is not necessary to use OPTIONS to discover support of a method, |
the client could simply try the method. If the receiver of the |
request does not support the method it will respond with an error |
code indicating the the method is either not implemented (501) or |
does not apply for the resource (405). The choice between the two |
discovery methods depends on the requirements of the service.
To handle functionality additions that are not new methods feature-
tags are defined. Each feature-tag represents a certain block of
functionality. The amount of functionality that a feature-tag
represents can vary significantly. A simple feature-tag can simple
represent the functionality a single header gives. Another feature-
tag is "play.basic" which represents the minimal playback
implementation according to the updated specification.
The feature-tags are then used to determine if the client, server or
proxy supports the functionality that is necessary to achieve the
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desired service. To determine support of a feature-tag several
different headers can be used, each explained below:
Supported: The supported header is used to determine the
complete set of functionality that both client and server
has. The intended usage is to determine before one needs to
use a functionality that it is supported. If can be used in
any method however OPTIONS is the most suitable as one at
the same time determines all methods that are implemented.
When sending a request the requestor declares all its
capabilities by including all supported feature-tags. The
results in that the receiver learns the requestors feature
support. The receiver then includes its set of features in
the response.
Require: The Require header can be included in any request where
the end point, i.e. the client or server, is required to
understand the feature to correctly perform the request.
This can for example be a SETUP request where the server
must understand a certain parameter to be able to set up
the media delivery correctly. Ignoring this parameter would
not have the desired effect and is not acceptable.
Therefore the end-point receiving a request containing a
Require must negatively acknowledge any feature that it
does not understand and not perform the request. The
response in cases where features are not understood are 551
(Option Not Supported). Also the features that are not
understood are given in the Unsupported header in the
response.
Proxy-Require: This method has the same purpose and workings as
Require except that it only applies to proxies and not the
end point. Features that needs to be supported by both
proxies and end-point needs to be included in both the
Require and Proxy-Require header.
Unsupported: This header is used in 551 error response to tell
which feature(s) that was not supported. Such a response is
only the result of the usage of the Require and/or Proxy-
Require header where one or more feature where not
supported. This information allows the requestor to make
the best of situations as it knows which features that was
not supported.
11 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
H. Schulzrinne et. al. [Page 32]
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defined in the future. Method names may not start with a $ character
(decimal 24) and must be a token as defined by the ABNF. Methods are
summarized in Table 2.
method direction object Server req. Client req.
___________________________________________________________________
DESCRIBE C -> S P,S recommended recommended
GET_PARAMETER C -> S, S -> C P,S optional optional
OPTIONS C -> S, S -> C P,S R=Req, Sd=Opt Sd=Req, R=Opt
PAUSE C -> S P,S recommended recommended
PING C -> S, S -> C P,S recommended optional
PLAY C -> S P,S required required
REDIRECT S -> C P,S optional optional
SETUP C -> S S required required
SET_PARAMETER C -> S, S -> C P,S optional optional
TEARDOWN C -> S P,S required required
Table 2: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on. Legend: R=Responde to,
Sd=Send, Opt: Optional, Req: Required, Rec: Recommended
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.
11.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An OPTIONS
request may be issued at any time, e.g., if the client is about to
try a nonstandard request. It does not influence the session state.
The Public header MUST be included in responses to indicate which
methods that are supported by the server. To specify which methods
that are possible to use for the specified resource, the Allow MAY be
used. By including in the OPTIONS request a Supported header, the
requester can determine which features the other part supports.
The request URI determines which scope the OPTIONS request has. By
giving the URI of a certain media the capabilities regarding this
media will be responded. By using the "*" URI the request regards the
next hop only, while having a URL with only the host address regards
the server without any media relevance.
The OPTIONS method can be used for RTSP session keep alive
signalling, however this method is not the most recommended one, see
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section 13.37 for a preference list. A keep alive OPTIONS request
SHOULD use the media or aggregated control URI.
Example:
C->S: OPTIONS * RTSP/1.0
CSeq: 1
User-Agent: PhonyClient/1.2
Require:
Proxy-Require: gzipped-messages
Supported: play-basic
S->C: RTSP/1.0 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Supported: play-basic, implicit-play, gzipped-messages
Server: PhonyServer/1.0
Note that some of the feature-tags in Require and Proxy-Require 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).
11.2 DESCRIBE
The DESCRIBE method retrieves the description of a presentation or
media object identified by the request URL from a server. It may use
the Accept header to specify the description formats that the client
understands. The server responds with a description of the requested
resource. The DESCRIBE reply-response pair constitutes the media
initialization phase of RTSP.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
User-Agent: PhonyClient 1.2
Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 OK
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
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Server: PhonyServer 1.0
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=application 32416 UDP WB
a=orient:portrait
The DESCRIBE response MUST contain all media initialization
information for the resource(s) that it describes. If a media client
obtains a presentation description from a source other than DESCRIBE
and that description contains a complete set of media initialization
parameters, the client SHOULD use those parameters and not then
request a description for the same media via RTSP.
Additionally, servers SHOULD NOT use the DESCRIBE response as a means
of media indirection.
By forcing a DESCRIBE response to contain all media
initialization for the set of streams that it describes,
and discouraging use of DESCRIBE for media indirection, we
avoid looping problems that might result from other
approaches.
Media initialization is a requirement for any RTSP-based system, but
the RTSP specification does not dictate that this must be done via
the DESCRIBE method. There are three ways that an RTSP client may
receive initialization information:
o via RTSP's DESCRIBE method;
o via some other protocol (HTTP, email attachment, etc.);
o via the command line or standard input (thus working as a
browser helper application launched with an SDP file or other
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media initialization format).
It is RECOMMENDED that minimal servers support the DESCRIBE method,
and highly recommended that minimal clients support the ability to
act as a "helper application" that accepts a media initialization
file from standard input, command line, and/or other means that are
appropriate to the operating environment of the client.
11.3 SETUP
The SETUP request for a URI specifies the transport mechanism to be |
used for the streamed media. The SETUP method may be used in two |
different cases; Create a RTSP session or add a media to a session, |
and change the transport parameters of already set up media stream. |
Using SETUP to create or add media to a session when in PLAY state |
are not allowed. Otherwise SETUP can be used in all three states; |
INIT, and READY, for both purposes and in PLAY to change the |
transport parameters. |
The Transport header, see section 13.40, specifies the transport |
parameters acceptable to the client for data transmission; the |
response will contain the transport parameters selected by the |
server. This allows the client to enumerate in priority order the |
transport mechanisms and parameters acceptable to it, while the |
server can select the most appropriate. All transport parameters |
SHOULD be included in the Transport header, the use of other headers |
for this purpose is discouraged due to middle boxes. |
For the benefit of any intervening firewalls, a client SHOULD |
indicate the transport parameters 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 initialization. |
In a SETUP response the server SHOULD include the Accept-Ranges |
header (see section 13.4 to indicate which time formats that are |
acceptable to use for this media resource. |
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 |
CSeq: 302 |
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Transport: RTP/AVP;unicast;client_port=4588-4589, |
RTP/AVP/TCP;unicast;interleave=0-1 |
S->C: RTSP/1.0 200 OK |
CSeq: 302 |
Date: 23 Jan 1997 15:35:06 GMT |
Server: PhonyServer 1.0 |
Session: 47112344 |
Transport: RTP/AVP;unicast;client_port=4588-4589; |
server_port=6256-6257;ssrc=2A3F93ED |
Accept-Ranges: NPT |
In the above example the client want to create a RTSP session |
containing the media resource "rtsp://example.com/foo/bar/baz.rm". |
The transport parameters acceptable to the client is either |
RTP/AVP/UDP (UDP per default) to be received on client port 4588 and |
4589 or RTP/AVP interleaved on the RTSP control channel. The server |
selects the RTP/AVP/UDP transport and adds the ports it will send and |
received RTP and RTCP from, and the RTP SSRC that will be used by the |
server. |
The server MUST generate a session identifier in response to a |
successful SETUP request, unless a SETUP request to a server includes |
a session identifier, in which case the server MUST bundle this setup |
request into the existing session (aggregated session) or return |
error 459 (Aggregate Operation Not Allowed) (see Section 12.4.11). |
An Aggregate control URI MUST be used to control an aggregated |
session. This URI MUST be different from the stream control URIs of |
the individual media streams included in the aggregate. The Aggregate |
control URI is to be specified by the session description if the |
server supports aggregated control and aggregated control is desired |
for the session. However even if aggregated control is offered the |
client MAY chose to not set up the session in aggregated control. |
If an Aggregate control URI is not specified in the session |
description, it is probably a indication that non-aggregated control |
should be used. However a client MAY try to SETUP the session in |
aggregated control. If the server refuse to aggregate the specified |
media, the server SHALL use the 459 error code. If the server allows |
the aggregation, then the client MUST create an URI for aggregate |
control of the session. This URI MUST contain the servers host |
address and MUST contain the port, if applicable (e.g. not default |
port). Once an URI is used to refer to an aggregation for a given |
session, that URI MUST be used to refer to that aggregation for the |
duration of the session. |
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While the session ID sometimes has enough information for |
aggregate control of a session, the Aggregate control URI |
is still important for some methods such as SET_PARAMETER |
where the control URI enables the resource in question to |
be easily identified. The Aggregate control URI is also |
useful for proxies, enabling them to route the request to |
the appropriate server, and for logging, where it is useful |
to note the actual resource that a request was operating |
on. Finally, presence of the Aggregate control URI allows |
for backwards compatibility with RFC 2326 [21].
A session will exist until it is either removed by a TEARDOWN request
or is timed-out by the server. The server MAY remove a session that
has not demonstrated liveness signs from the client within a certain
timeout period. The default timeout value is 60 seconds; the server
MAY set this to a different value and indicate so in the timeout
field of the Session header in the SETUP response. For further
discussion see chapter 13.37. Signs of liveness for a RTSP session
are:
o Any RTSP request from a client which includes a Session header
with that session's ID.
o If RTP is used as a transport for the underlying media
streams, an RTCP sender or receiver report from the client for
any of the media streams in that RTSP session.
If a SETUP request on a session fails for any reason, the session |
state, as well as transport and other parameters for associated |
streams SHALL remain unchanged from their values as if the SETUP |
request had never been received by the server. |
A client MAY issue a SETUP request for a stream that is already set |
up or playing in the session 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). Reasons to support |
changing transport parameters, is to allow for application layer |
mobility and flexibility to utilize the best available transport as |
it becomes available. |
In a SETUP response for a request to change the transport parameters |
while in Play state, the server SHOULD include the Range to indicate |
from what point the new transport parameters are used. Further if RTP |
is used for delivery the server SHOULD also include the RTP-Info |
header to indicate from what timestamp and RTP sequence number the |
change has taken place. If both RTP-Info and Range is included in the |
response the "rtp_time" parameter and range MUST be for the |
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corresponding time, i.e. be used in the same way as for PLAY to |
ensure the correct synchronization information is present. |
If the transport parameter change while in PLAY state results in a |
change of synchronization related information, for example changing |
RTP SSRC, the server MUST provide in the SETUP response the necessary |
synchronization information. However the server is RECOMMENDED to |
avoid changing the synchronization information if possible. |
11.4 PLAY
The PLAY method tells the server to start sending data via the |
mechanism specified in SETUP. A client MUST NOT issue a PLAY request |
until any outstanding SETUP requests have been acknowledged as |
successful. PLAY requests are valid when the session is in READY |
state; the use of PLAY requests when the session is in PLAY state is |
deprecated. A PLAY request MUST include a Session header to indicate |
which session the request applies to.
In an aggregated session the PLAY request MUST contain an aggregated
control URL. A server SHALL responde with error 460 (Only Aggregate
Operation Allowed) if the client PLAY request URI is for one of the
media. The media in an aggregate SHALL be played in sync. If a client
want individual control of the media it must use separate RTSP
sessions for each media.
The PLAY request SHALL position the normal play time to the beginning |
of the range specified by the Range header and delivers stream data |
until the end of the range if given, else to the end of the media is |
reached. To allow for precise composition multiple ranges MAY be |
specified in one PLAY Request. The range values are valid if all |
given ranges are part of any media within the aggregate. If a given |
range value points outside of the media, the response SHALL be the |
457 (Invalid Range) error code.
The below example 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, npt=20-25, npt=30-
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See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It SHALL start |
playing a stream from the beginning (npt=0-) unless the stream has |
been paused. If a stream has been paused via PAUSE, stream delivery |
resumes at the pause point. The stream SHALL play until the end of |
the media.
The Range header MUST NOT contain a time parameter. The usage of time
in PLAY method has been deprecated.
Server MUST include a "Range" header in any PLAY response. The |
response MUST use the same format as the request's range header |
contained. If no Range header was in the request, the NPT time format |
SHOULD be used unless the client showed support for an other format. |
Also for a session with live media streams the Range header MUST |
indicate a valid time. It is RECOMMENDED that normal play time is |
used, either the "now" indicator, for example "npt=now-", or the time |
since session start as an open interval, e.g. "npt=96.23-". An |
absolute time value (clock) for the corresponding time MAY be given, |
i.e. "clock=20030213T143205Z-". The UTC clock format SHOULD only be |
used if client has shown support for it. |
A media server only supporting playback MUST support the npt format |
and MAY support the clock and smpte formats. |
For a on-demand stream, the server MUST reply 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 start position SHALL still be returned in the reply. If |
the medias that are part of an aggregate has different lengths, the |
PLAY request SHALL be performed as long as the given range is valid |
for any media, for example the longest media. Media will be sent |
whenever it is available for the given play-out point. |
After playing the desired range, the presentation is NOT |
automatically paused, media delivery simply stops. A PAUSE request |
MUST be issued before another PLAY request can be issued. Note: This |
is one change resulting in a non-operability with RFC 2326 |
implementations. A client not issuing a PAUSE request before a new |
PLAY will be stuck in PLAY state. |
A client desiring to play the media from the beginning MUST send a |
PLAY request with a Range header pointing at the beginning, e.g. |
npt=0-. If a PLAY request is received without a Range header when |
media delivery has stopped at the end, the server SHOULD respond with |
a 457 "Invalid Range" error response. In that response the current |
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pause point in a Range header SHALL be included.
The following example plays the whole presentation starting at SMPTE
time code 0:10:20 until the end of the clip. Note: The RTP-Info
headers has been broken into several lines to fit the page.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
CSeq: 833
Session: 12345678
Range: smpte=0:10:20-
S->C: RTSP/1.0 200 OK
CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: smpte=0:10:22-
RTP-Info:url=rtsp://example.com/twister.en;
seq=14783;rtptime=2345962545
For playing back a recording of a live presentation, it may be
desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
CSeq: 835
Session: 12345678
Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT
Server:PhonyServer 1.0
Range: clock=19961108T142300Z-19961108T143520Z
RTP-Info:url=rtsp://example.com/meeting.en;
seq=53745;rtptime=484589019
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
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Range header. If the pause point is located later than the given end
value, a 457 (Invalid Range) response SHALL be given.
The queued play functionality described in RFC 2326 [21] is removed
and multiple ranges can be used to achieve a similar performance. If
a server receives a PLAY request while in the PLAY state, the server
SHALL responde using the error code 455 (Method Not Valid In This
State). This will signal the client that queued play are not
supported.
The use of PLAY for keep-alive signaling, i.e. PLAY request without a |
range header in PLAY state, has also been depreciated. Instead a |
client can use, PING, SET_PARAMETER or OPTIONS for keep alive. A |
server receiving a PLAY keep alive SHALL respond with the 455 error |
code.
11.5 PAUSE
The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. A PAUSE request MUST be done with the
aggregated control URI for aggregated sessions, resulting in all
media being halted, or the media URI for non-aggregated sessions.
Any attempt to do muting of a single media with an PAUSE request in
an aggregated session SHALL be responded with error 460 (Only
Aggregate Operation Allowed). After resuming playback,
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 |
Range: npt=45.76- |
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
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"pause point". The time parameter in the Range MUST NOT be used. The
Range 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 normal play time,
under the assumption that the PAUSE request reaches the server within
11 seconds of the PLAY request. Note that some lines has been broken
in an non-correct way to fit the page:
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
Server: PhonyServer 1.0
Range: npt=10-30
RTP-Info:url=rtsp://example.com/fizzle/audiotrack;
seq=5712;rtptime=934207921,
url=rtsp://example.com/fizzle/videotrack;
seq=57654;rtptime=2792482193
Session: 12345678
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
Server: PhonyServer 1.0
Range: npt=21-
Session: 12345678
The pause request becomes effective the first time the server is
encountering the time point specified in any of the multiple ranges.
If the Range header specifies a time outside any range from the PLAY
request, the error 457 (Invalid Range) SHALL be returned. If a media
unit (such as an audio or video frame) starts presentation at exactly
the pause point, it is not played. If the Range header is missing,
stream delivery is interrupted immediately on receipt of the message
and the pause point is set to the current normal play time. However,
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the pause point in the media stream MUST be maintained. A subsequent
PLAY request without Range header SHALL resume from the pause point
and play until media end.
If the server has already sent data beyond the time specified in the |
PAUSE request's Range header, a PLAY without range SHALL resume at |
the point in time specified by the PAUSE request's Range header, as |
it is assumed that the client has discarded data after that point. |
This ensures continuous pause/play cycling without gaps. |
The pause point after any PAUSE request SHALL be returned to the |
client by adding a Range header with what remains unplayed of the |
PLAY request's ranges, i.e. including all the remaining ranges part |
of multiple range specification. If one desires to resume playing a |
ranged request, one simple included the Range header from the PAUSE |
response. Note that this server behavior was not mandated previously |
and servers implementing according to RFC 2326 will probably not |
return the range header. |
For example, if the server have a play request 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 returns a 457 error message. To |
prevent that the second range is played and the server stops after |
completing the first range, a PAUSE request for 20 must be issued. |
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 range effectively being ignored, assuming the |
PAUSE request arrives before the server has started playing the |
second, overlapping range. Regardless of when the PAUSE request |
arrives, it sets the pause point to 14. The below example messages is |
for the above case when the PAUSE request arrives before the first |
occurrence of NPT=14. |
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0 |
CSeq: 834 |
Session: 12345678 |
Range: npt=10-15, npt=13-20 |
S->C: RTSP/1.0 200 OK |
CSeq: 834 |
Date: 23 Jan 1997 15:35:06 GMT |
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Server: PhonyServer 1.0 |
Range: npt=10-15, npt=13-20 |
RTP-Info:url=rtsp://example.com/fizzle/audiotrack; |
seq=5712;rtptime=934207921, |
url=rtsp://example.com/fizzle/videotrack; |
seq=57654;rtptime=2792482193 |
Session: 12345678 |
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 |
CSeq: 835 |
Session: 12345678 |
Range: npt=14- |
S->C: RTSP/1.0 200 OK |
CSeq: 835 |
Date: 23 Jan 1997 15:35:09 GMT |
Server: PhonyServer 1.0 |
Range: npt=14-15, npt=13-20 |
Session: 12345678 |
If a client issues a PAUSE request and the server acknowledges and |
enters the READY state, the proper server response, if the player |
issues another PAUSE, is still 200 OK. The 200 OK response MUST |
include the Range header with the current pause point, even if the |
PAUSE request is asking for some other pause point. See examples |
below:
Examples:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 834
Session: 12345678
Date: 23 Jan 1997 15:35:06 GMT
Range: npt=45.76-
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 835
Session: 12345678
Range: 86-
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S->C: RTSP/1.0 200 OK
CSeq: 835
Session: 12345678
Date: 23 Jan 1997 15:35:07 GMT
Range: npt=45.76-
11.6 TEARDOWN
The TEARDOWN client to server request stops the stream delivery for |
the given URI, freeing the resources associated with it. TEARDOWN |
MAY be done using either an aggregated or a media control URI. |
However some restrictions apply depending on the current state. The |
TEARDOWN request SHALL contain a Session header indicating what |
session the request applies to. |
A TEARDOWN using the aggregated control URI or the media URI in a |
session under non-aggregated control MAY be done in any state (Ready, |
and Play). A successful request SHALL result in that media delivery |
is immediately halted and the session state is destroyed. This SHALL |
be indicated through the lack of a Session header in the response. |
A TEARDOWN using a media URI in an aggregated session MAY only be |
done in Ready state. Such a request only removes the indicated media |
stream and associated resources from the session. This may result in |
that a session returns to non-aggregated control. In the response to |
TEARDOWN request resulting in that the session still exist SHALL |
contain a Session header to indicate this. |
Note, the indication with the session header if sessions state remain |
may not be done correctly by a RFC 2326 client, but will be for any |
server signalling the "play.basic" tag.
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
Server: PhonyServer 1.0
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11.7 GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a
presentation or stream specified in the URI. If the Session header is
present in a request, the value of a parameter MUST be retrieved in
the sessions context. The content of the reply and response is left
to the implementation. GET_PARAMETER with no entity body may be used
to test client or server liveness ("ping").
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 431
Content-Type: text/parameters
Session: 12345678
Content-Length: 15
packets_received
jitter
C->S: RTSP/1.0 200 OK
CSeq: 431
Content-Length: 46
Content-Type: text/parameters
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.
11.8 SET_PARAMETER
This method requests to set the value of a parameter for a
presentation or stream specified by the URI.
A request is RECOMMENDED to only contain a single parameter to allow
the client to determine why a particular request failed. If the
request contains several parameters, the server MUST only act on the
request if all of the parameters can be set successfully. A server
MUST allow a parameter to be set repeatedly to the same value, but it
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MAY disallow changing parameter values. If the receiver of the
request does not understand or can locate a parameter error 451
(Parameter Not Understood) SHALL be used. In the case a parameter is
not allowed to change the error code 458 (Parameter Is Read-Only).
The response body SHOULD contain only the parameters that has errors.
Otherwise no body SHALL be returned.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
Restricting setting transport parameters to SETUP is for
the benefit of firewalls.
The parameters are split in a fine-grained fashion so that
there can be more meaningful error indications. However, it
may make sense to allow the setting of several parameters
if an atomic setting is desirable. Imagine device control
where the client does not want the camera to pan unless it
can also tilt to the right angle at the same time.
Example:
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 421
Content-length: 20
Content-type: text/parameters
barparam: barstuff
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.
11.9 REDIRECT
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A redirect request informs the client that it MUST connect to another |
server location. The REDIRECT request MAY contain the header |
Location, which indicates that the client should issue requests for |
that URL. The lack of a Location header in the response SHALL be |
interpreted as that the server can't any longer fulfill the current |
request, but has no alternative at the present where the client |
continue. |
If a REDIRECT request contains a Session header, it is end-to-end and |
applies only to the given session. If there are proxies in the |
request chain, they SHOULD NOT disconnect the control channel unless |
there are no remaining sessions. If the Location header is included |
it SHALL contain a full absolute URI pointing out the resource to |
reconnect too, i.e. the Location SHALL NOT contain only host and |
port. |
If a REDIRECT request does not contain a Session header, it is next- |
hop and applies also to the control connection. If the Location |
header is included it SHOULD only contain an absolute URI containing |
the host address and OPTIONAL the port number. If there are proxies |
in the request chain, they SHOULD do all of the following: (1) |
respond to the REDIRECT request, (2) disconnect the control channel |
from the requestor, (3) reconnect to the given host address, and (4) |
pass the request to each applicable client (typically those clients |
with an active session or unanswered request from the requestor). |
Note that the proxy is responsible for accepting the REDIRECT |
response from its clients and these responses MUST NOT be passed on |
to either the requesting or the destination server.
The redirect request MAY contain the header Range, which indicates
when the redirection takes effect. If the Range contains a "time="
value that is the wall clock time that the redirection MUST at the
latest take place. When the "time=" parameter is present the range
value MUST be ignored. However the range entered MUST be syntactical
correct and SHALL point at the beginning of any on-demand content. If
no time parameter is part of the Range header then redirection SHALL
take place when the media playout from the server reaches the given
time. The range value MUST be a single value in the open ended form,
e.g. npt=59-.
A server upon receiving a successful (2xx) response for a REDIRECT |
request without any Range header SHALL consider the session as |
removed and can free any session state. For this type of requests the |
rest of this paragraph applies. The server MAY close the signalling |
connection upon receiving the response for REDIRECT requests without |
a Session header. The client SHOULD close the signaling connection |
after having given the 2xx response to a REDIRECT response, unless it |
has several sessions on the server. If the client has multiple |
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session on the server it SHOULD close the connection when it has |
received and responded to REDIRECT requests for all sessions. |
A client receiving a REDIRECT request with a Range header SHALL issue |
a TEARDOWN request when the in indicated redirect point is reached. |
The client SHOULD for REDIRECT requests with Range header close the |
signalling connection after a 2xx response on its TEARDOWN request. |
The normal connection considerations apply for the server. This |
differentiation from REDIRECT requests without range headers is to |
allow clear an explicit state handling. As the state in the server |
needs to be kept until the point of redirection, the handling becomes |
more clear if the client is required to tear down the session at that |
point. |
If the client wants to continue to send or receive media for this |
resource, the client will have to establish a new session with the |
designated host. A client SHOULD issue a new DESCRIBE request with |
the URL given in the Location header, unless the URL only contains a |
host address. In the cases the Location only contains a host address |
the client MAY assume that the media on the server it is redirected |
to is identical. Identical media means that all media configuration |
information from the old session still is valid except for the host |
address. In the case of absolute URLs in the location header the |
media redirected to can be either identical, slightly different or |
totally different. This is the reason why a new DESCRIBE request |
SHOULD be issued.
This example request redirects traffic for this session to the new
server at the given absolute time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 732
Location: rtsp://bigserver.com:8001
Range: npt=0- ;time=19960213T143205Z
Session: uZ3ci0K+Ld-M
11.10 PING
This method is a bi-directional mechanism for server or client
liveness checking. It has no side effects. The issuer of the request
MUST include a session header with the session ID of the session that
is being checked for liveness.
Prior to using this method, an OPTIONS method is RECOMMENDED to be
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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. Note: That an 501 (Not Implemented) response means that the
keep-alive timer has not been updated.
When a proxy is in use, PING with a * indicates a single-hop liveness
check, whereas PING with a URL including an host address indicates an
end-to-end liveness check.
Example:
C->S: PING * RTSP/1.0
CSeq: 123
Session:12345678
S->C: RTSP/1.0 200 OK
CSeq: 123
Session:12345678
11.11 Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server
to interleave RTSP messages and media stream data. This interleaving
should generally be avoided unless necessary since it complicates
client and server operation and imposes additional overhead. Also
head of line blocking may cause problems. Interleaved binary data
SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte channel identifier,
followed by the length of the encapsulated binary data as a binary,
two-byte integer in network byte order. The stream data follows
immediately afterwards, without a CRLF, but including the upper-layer
protocol headers. Each $ block contains exactly one upper-layer
protocol data unit, e.g., one RTP packet.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| "$" = 24 | Channel ID | Length in bytes |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Length number of bytes of binary data :
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The channel identifier is defined in the Transport header with the
interleaved parameter(Section 13.40).
When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. The usage of RTCP messages is
indicated by including a range containing a second channel in the
interleaved parameter of the Transport header, see section 13.40. If
RTCP is used, packets SHALL be sent on the first available channel
higher than the RTP channel. The channels are bi-directional and
therefore RTCP traffic are sent on the second channel in both
directions.
RTCP is needed for synchronization when two or more streams
are interleaved in such a fashion. Also, this provides a
convenient way to tunnel RTP/RTCP packets through the TCP
control connection when required by the network
configuration and transfer them onto UDP when possible.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
CSeq: 2
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
S->C: RTSP/1.0 200 OK
CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;unicast;interleaved=5-6
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
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S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
S->C: $006{2 byte length}{"length" bytes RTCP packet}
12 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. All
error messages, 4xx and 5xx MAY return a body containing further
information about the error.
12.1 Success 1xx
12.1.1 100 Continue
See, [H10.1.1].
12.2 Success 2xx
12.2.1 250 Low on Storage Space
The server returns this warning after receiving a RECORD request that
it may not be able to fulfill completely due to insufficient storage
space. If possible, the server should use the Range header to
indicate what time period it may still be able to record. Since other
processes on the server may be consuming storage space
simultaneously, a client should take this only as an estimate.
12.3 Redirection 3xx
The notation "3rr" indicates response codes from 300 to 399 inclusive
which are meant for redirection. The response code 304 is excluded
from this set, as it is not used for redirection.
See [H10.3] for definition of status code 300 to 305. However
comments are given for some to how they apply to RTSP.
Within RTSP, redirection may be used for load balancing or
redirecting stream requests to a server topologically closer to the
client. Mechanisms to determine topological proximity are beyond the
scope of this specification.
If the the Location header is used in a response it SHALL contain an |
absolute URI pointing out the media resource the client is redirected |
to, the URI SHALL NOT only contain the host name.
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12.3.1 300 Multiple Choices
12.3.2 301 Moved Permanently
The request resource are moved permanently and resides now at the URI
given by the location header. The user client SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message-body.
12.3.3 302 Found
The requested resource reside temporarily at the URI given by the
Location header. The Location header MUST be included in the
response. Is intended to be used for many types of temporary
redirects, e.g. load balancing. It is RECOMMENDED that one set the
reason phrase to something more meaningful than "Found" in these
cases. The user client SHOULD redirect automatically to the given
URI. This response MUST NOT contain a message-body.
12.3.4 303 See Other
This status code SHALL NOT be used in RTSP. However as it was allowed
to use in RFC 2326 it is possible that such response may be received.
12.3.5 304 Not Modified
If the client has performed a conditional DESCRIBE or SETUP (see
12.23) and the requested resource has not been modified, the server
SHOULD send a 304 response. This response MUST NOT contain a
message-body.
The response MUST include the following header fields:
o Date
o ETag and/or Content-Location, if the header would have been
sent in a 200 response to the same request.
o Expires, Cache-Control, and/or Vary, if the field-value might
differ from that sent in any previous response for the same
variant.
This response is independent for the DESCRIBE and SETUP requests.
That is, a 304 response to DESCRIBE does NOT imply that the resource
content is unchanged and a 304 response to SETUP does NOT imply that
the resource description is unchanged. The ETag and If-Match headers
may be used to link the DESCRIBE and SETUP in this manner.
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12.3.6 305 Use Proxy
See [H10.3.6].
12.4 Client Error 4xx
12.4.1 400 Bad Request
The request could not be understood by the server due to malformed
syntax. The client SHOULD NOT repeat the request without
modifications [H10.4.1]. If the request does not have a CSeq header,
the server MUST NOT include a CSeq in the response.
12.4.2 405 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.
12.4.3 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
contained in the request.When returning this error message the sender
SHOULD return a entity body containing the offending parameter(s).
12.4.4 452 reserved
This error code was removed from RFC 2326 [21] and is obsolete.
12.4.5 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.
12.4.6 454 Session Not Found
The RTSP session identifier in the Session header is missing,
invalid, or has timed out.
12.4.7 455 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
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recovery easier.
12.4.8 456 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. This error message may also be used for specifying when the
time format in Range is impossible for the resource. In that case the
Accept-Ranges header SHOULD be returned to inform the client of which
format(s) that are allowed.
12.4.9 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
12.4.10 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not
modified. When returning this error message the sender SHOULD return
a entity body containing the offending parameter(s).
12.4.11 459 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
media URL.
12.4.12 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URL in question since
it is not an aggregate control (presentation) URL. The method may be
applied on the aggregate control URL.
12.4.13 461 Unsupported Transport
The Transport field did not contain a supported transport
specification.
12.4.14 462 Destination Unreachable
The data transmission channel could not be established because the
client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid Destination
parameter in the Transport field.
12.5 Server Error 5xx
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12.5.1 551 Option not supported
An feature-tag given in the Require or the Proxy-Require fields was
not supported. The Unsupported header SHOULD be returned stating the
feature for which there is no support.
13 Header Field Definitions
method direction object acronym Body
_________________________________________________
DESCRIBE C -> S P,S DES r
GET_PARAMETER C -> S, S -> C P,S GPR R,r
OPTIONS C -> S P,S OPT
S -> C
PAUSE C -> S P,S PSE
PING C -> S, S -> C P,S PNG
PLAY C -> S P,S PLY
REDIRECT S -> C P,S RDR
SETUP C -> S S STP
SET_PARAMETER C -> S, S -> C P,S SPR R,r
TEARDOWN C -> S P,S 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: It is allowed for all error messages 4xx and 5xx to
have a body
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
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
processing is summarized in Table 4 and Table 5.
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 when abbreviated, are according to table 3:
c: Conditional; requirements on the header field depend on the
context 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.
*: The header field is required if the message body is not
empty. See sections 13.14, 13.16 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 13.32). 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
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
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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
double 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.
13.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
The "level" parameter for presentation descriptions is
properly defined as part of the MIME type registration, not
here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl q=1.0, application/sdp;level=2
13.2 Accept-Encoding
See [H14.3]
13.3 Accept-Language
See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media
content.
13.4 Accept-Ranges
The Accept-Ranges response-header field allows the server to indicate
its acceptance of range requests and possible formats for a resource: |
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Accept-Ranges = "Accept-Ranges" ":" acceptable-ranges |
acceptable-ranges = 1#range-unit / "none" |
range-unit = NPT / SMPTE / UTC / extension-format |
extension-format = token |
This header has the same syntax as [H14.5]. However new range-units |
are defined. Inclusion of any of the time formats indicates |
acceptance by the server for PLAY and PAUSE requests with this |
format. The headers value is valid for the resource specified by the |
URI in the request, this response corresponds to. A server is SHOULD |
to use this header in SETUP responses to indicate to the client which |
range time formats the media supports. The header SHOULD also be |
included in "456" responses which is a result of use of unsupported |
range formats. |
13.5 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. See [H14.7] for syntax definition.
Example of use:
Allow: SETUP, PLAY, SET_PARAMETER
13.6 Authorization
See [H14.8]
13.7 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:
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Header Where Proxy DES OPT SETUP PLAY PAUSE TRD
_____________________________________________________________
Accept R o - - - - -
Accept-Encoding R r o - - - - -
Accept-Language R r o - - - - -
Accept-Ranges r r - - o - - -
Accept-Ranges 456 r - - - o o -
Allow r - o - - - -
Allow 405 - - - m m -
Authorization R o o o o o o
Bandwidth R o o o o - -
Blocksize R o - o o - -
Cache-Control r - - o - - -
Connection o o o o o o
Content-Base r o - - - - -
Content-Base 4xx o o o o o o
Content-Encoding R r - - - - - -
Content-Encoding r r o - - - - -
Content-Encoding 4xx r o o o o o o
Content-Language R r - - - - - -
Content-Language r r o - - - - -
Content-Language 4xx r o o o o o o
Content-Length r r * - - - - -
Content-Length 4xx r * * * * * *
Content-Location r o - - - - -
Content-Location 4xx o o o o o o
Content-Type r * - - - - -
Content-Type 4xx * * * * * *
CSeq Rc m m m m m m
Date am o o o o o o
Expires r r o - - - - -
From R r o o o o o o
Host - - - - - -
If-Match R r - - o - - -
If-Modified-Since R r o - o - - -
Last-Modified r r o - - - - -
Location 3rr o o o o o o
Proxy-Authenticate 407 amr m m m m m m
Proxy-Require R ar o o o o o o
Public r admr - m* - - - -
Public 501 admr m* m* m* m* m* m*
Range R - - - o o -
Range r - - c m* m* -
Referer R o o o o o o
Require R o o o o o o
Retry-After 3rr,503 o o o - - -
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Header Where Proxy DES OPT SETUP PLAY PAUSE TRD
_________________________________________________________
Scale - - - o - -
Session R - o o m m m
Session r - c m m m o
Server R - o - - - -
Server r o o o o o o
Speed - - - o - -
Supported R o o o o o o
Supported r c c c c c c
Timestamp R o o o o o o
Timestamp c m m m m m m
Transport - - m - - -
Unsupported r c c c c c c
User-Agent R m* m* m* m* m* m*
Vary r c c c c c c
Via R amr o o o o o o
Via c dr m m m m m m
WWW-Authenticate 401 m m m m m m
_________________________________________________________
Header Where Proxy DES OPT SETUP PLAY PAUSE TRD
Table 4: Overview of RTSP header fields related to methods DESCRIBE,
OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.
Bandwidth: 4000
13.8 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
(400) if the value is syntactically invalid.
Blocksize = "Blocksize" ":" 1*DIGIT
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Header Where Proxy GPR SPR RDR PNG
__________________________________________________
Allow 405 - - - -
Authorization R o o o o
Bandwidth R - o - -
Blocksize R - o - -
Connection o o o - Content-
Base R o o - - Content-
Base r o o - - Content-
Base 4xx o o o - Content-
Encoding R r o o - - Content-
Encoding r r o o - - Content-
Encoding 4xx r o o o - Content-
Language R r o o - - Content-
Language r r o o - - Content-
Language 4xx r o o o - Content-
Length R r * * - - Content-
Length r r * * - - Content-
Length 4xx r * * * - Content-
Location R o o - - Content-
Location r o o - - Content-
Location 4xx o o o - Content-
Type R * * - - Content-
Type r * * - - Content-
Type 4xx * * * -
CSeq Rc m m m m
Date am o o o o
From R r o o o o
Host - - - - Last-
Modified R r - - - - Last-
Modified r r o - - -
Location 3rr o o o o
Location R - - m - Proxy-
Authenticate 407 amr m m m m Proxy-
Require R ar o o o o
Public 501 admr m* m* m* m*
Range R - - o -
Referer R o o o -
Require R o o o o Retry-
After 3rr,503 o o - -
Scale - - - -
Session R o o o m
Session r c c o m
Server R o o o o
Server r o o - o
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Timestamp R o o o o
Timestamp c m m m m
Unsupported r c c c c User-
Agent R m* m* - m* User-
Agent r - - m* -
Vary r c c - -
Via R amr o o o o
Via c dr m m m m WWW-
Authenticate 401 m m m m
__________________________________________________
Header Where Proxy GPR SPR RDR PNG
Table 5: Overview of RTSP header fields related to methods
GET_PARAMETER, SET_PARAMETER,REDIRECT, and PING.
13.9 Cache-Control
The Cache-Control general-header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the
request/response chain.
Cache directives must be passed through by a proxy or gateway
application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a cache-
directive for a specific cache.
Cache-Control should only be specified in a SETUP request and its
response. Note: Cache-Control does not govern the caching of
responses as for HTTP, but rather of the stream identified by the
SETUP request. Responses to RTSP requests are not cacheable, except
for responses to DESCRIBE.
Cache-Control = "Cache-Control" ":" 1#cache-directive
cache-directive = cache-request-directive
/ cache-response-directive
cache-request-directive = "no-cache"
/ "max-stale" ["=" delta-seconds]
/ "min-fresh" "=" delta-seconds
/ "only-if-cached"
/ cache-extension
cache-response-directive = "public"
/ "private"
/ "no-cache"
/ "no-transform"
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/ "must-revalidate"
/ "proxy-revalidate"
/ "max-age" "=" delta-seconds
/ cache-extension
cache-extension = token [ "=" ( token / quoted-string ) ]
delta-seconds = 1*DIGIT
no-cache: Indicates that the media stream MUST NOT be cached
anywhere. This allows an origin server to prevent caching
even by caches that have been configured to return stale
responses to client requests.
public: Indicates that the media stream is cacheable by any
cache.
private: Indicates that the media stream is intended for a
single user and MUST NOT be cached by a shared cache. A
private (non-shared) cache may cache the media stream.
no-transform: An intermediate cache (proxy) may find it useful
to convert the media type of a certain stream. A proxy
might, for example, convert between video formats to save
cache space or to reduce the amount of traffic on a slow
link. Serious operational problems may occur, however, when
these transformations have been applied to streams intended
for certain kinds of applications. For example,
applications for medical imaging, scientific data analysis
and those using end-to-end authentication all depend on
receiving a stream that is bit-for-bit identical to the
original entity-body. Therefore, if a response includes the
no-transform directive, an intermediate cache or proxy MUST
NOT change the encoding of the stream. Unlike HTTP, RTSP
does not provide for partial transformation at this point,
e.g., allowing translation into a different language.
only-if-cached: In some cases, such as times of extremely poor
network connectivity, a client may want a cache to return
only those media streams that it currently has stored, and
not to receive these from the origin server. To do this,
the client may include the only-if-cached directive in a
request. If it receives this directive, a cache SHOULD
either respond using a cached media stream that is
consistent with the other constraints of the request, or
respond with a 504 (Gateway Timeout) status. However, if a
group of caches is being operated as a unified system with
good internal connectivity, such a request MAY be forwarded
within that group of caches.
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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.)
proxy-revalidate: The proxy-revalidate directive has the same
meaning as the must-revalidate directive, except that it
does not apply to non-shared user agent caches. It can be
used on a response to an authenticated request to permit
the user's cache to store and later return the response
without needing to revalidate it (since it has already been
authenticated once by that user), while still requiring
proxies that service many users to revalidate each time (in
order to make sure that each user has been authenticated).
Note that such authenticated responses also need the public
cache control directive in order to allow them to be cached
at all.
max-age: When an intermediate cache is forced, by means of a
max-age=0 directive, to revalidate its own cache entry, and
the client has supplied its own validator in the request,
the supplied validator might differ from the validator
currently stored with the cache entry. In this case, the
cache MAY use either validator in making its own request
without affecting semantic transparency.
However, the choice of validator might affect performance.
The best approach is for the intermediate cache to use its
own validator when making its request. If the server
replies with 304 (Not Modified), then the cache can return
its now validated copy to the client with a 200 (OK)
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response. If the server replies with a new entity and cache
validator, however, the intermediate cache can compare the
returned validator with the one provided in the client's
request, using the strong comparison function. If the
client's validator is equal to the origin server's, then
the intermediate cache simply returns 304 (Not Modified).
Otherwise, it returns the new entity with a 200 (OK)
response.
13.10 Connection
See [H14.10]. The use of the connection option "close" in RTSP
messages SHOULD be limited to error messages when the server is
unable to recover and therefore see it necessary to close the
connection. The reason is that the client shall have the choice of
continue using a connection indefinitely as long as it sends valid
messages.
13.11 Content-Base
The Content-Base entity-header field may be used to specify the base
URI for resolving relative URLs within the entity.
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.
13.12 Content-Encoding
See [H14.11]
13.13 Content-Language
See [H14.12]
13.14 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
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[H14.13].
13.15 Content-Location
See [H14.14]
13.16 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.
13.17 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
number. Any retransmitted request must contain the same sequence
number as the original (i.e. the sequence number is not incremented
for retransmissions of the same request). For each new RTSP request
the CSeq value SHALL be incremented by one. The initial sequence
number MAY be any number. Each sequence number series is unique
between each requester and responder, i.e. the client has one series
for its request to a server and the server has another when sending
request to the client. Each requester and responder is identified
with its network address.
CSeq = "Cseq" ":" 1*DIGIT
13.18 Date
See [H14.18]. An RTSP message containing a body MUST include a Date
header if the sending host has a clock. Servers SHOULD include a Date
header in all other RTSP messages.
13.19 Expires
The Expires entity-header field gives a date and time after which the
description or media-stream should be considered stale. The
interpretation depends on the method:
DESCRIBE response: The Expires header indicates a date and time
after which the description should be considered stale.
A stale cache entry may not normally be returned by a cache (either a
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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 14 for further discussion of the
expiration model.
The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.
The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format:
Expires = "Expires" ":" HTTP-date
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occurred in the past
(i.e., already expired).
To mark a response as "already expired," an origin server should use
an Expires date that is equal to the Date header value. To mark a
response as "never expires," an origin server SHOULD use an Expires
date approximately one year from the time the response is sent.
RTSP/1.0 servers SHOULD NOT send Expires dates more than one year in
the future.
The presence of an Expires header field with a date value of some
time in the future on a media stream that otherwise would by default
be non-cacheable indicates that the media stream is cacheable, unless
indicated otherwise by a Cache-Control header field (Section 13.9).
13.20 From
See [H14.22].
13.21 Host
The Host HTTP request header field [H14.23] is not needed for RTSP, |
and SHALL NOT be sent. It SHALL be silently ignored if received.
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13.22 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.
13.23 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
13.24 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, the header field indicates the last modification date and
time of the description, for SETUP that of the media stream.
13.25 Location
See [H14.30].
13.26 Proxy-Authenticate
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See [H14.33].
13.27 Proxy-Require
The Proxy-Require request-header field is used to indicate proxy- |
sensitive features that MUST be supported by the proxy. Any Proxy- |
Require header features that are not supported by the proxy MUST be |
negatively acknowledged by the proxy to the client using the |
Unsupported header. Any feature tag included in the Proxy-Require |
does not apply to the server. To ensure that a feature is supported |
by both proxies and servers the tag must be included in also a |
Require header.
See Section 13.32 for more details on the mechanics of this message
and a usage example.
Proxy-Require = "Proxy-Require" ":" 1#feature-tag |
Example of use: |
Proxy-Require: play.basic |
13.28 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
methods. 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, SETUP, PLAY, PAUSE, TEARDOWN
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This header field applies only to the server directly connected to the
client (i.e., the nearest neighbor in a chain of connections). 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.
13.29 Range
The Range request and response header field specifies a range of
time. The range can be specified in a number of units. This
specification 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 normally not meaningful. The header MAY contain a time parameter
in UTC, specifying the time at which the operation is to be made
effective. This functionality SHALL only be used with the REDIRECT
method. Servers supporting the Range header MUST understand the NPT
range format and SHOULD understand the SMPTE range format. The Range
response header indicates what range of time is actually being
played. 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 first point, but
excluding the second 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
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in the future, although a server may refuse to keep server
resources for extended idle periods.
By default, range intervals increase, where the second point is
larger than the first point.
Example:
Range: npt=10-15
However, range intervals can also decrease if the Scale header (see
section 13.34) indicates a negative scale value. For example, this
would be the case when a playback in reverse is desired.
Example:
Scale: -1
Range: npt=15-10
Decreasing ranges are still half open intervals as described above.
Thus, For range A-B, A is closed and B is open. In the above example,
15 is closed and 10 is open. An exception to this rule is the case
when B=0 in a decreasing range. In this case, the range is closed on
both ends, as otherwise there would be no way to reach 0 on a reverse
playback.
Example:
Scale: -1
Range: npt=15-0
In this range both 15 and 0 are closed.
A decreasing range interval without a corresponding negative Scale
header is not valid.
13.30 Referer
See [H14.36]. The URL refers to that of the presentation description,
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typically retrieved via HTTP.
13.31 Retry-After
See [H14.37].
13.32 Require
The Require request-header field is used by clients or servers to
ensure that the other end-point supports features that are required
in respect to this request. It can also be used to query if the
other end-point supports certain features, however the use of the
Supported (Section 13.38) is much more effective in this purpose.
The server MUST respond to this header by using the Unsupported
header to negatively acknowledge those feature-tags which are NOT
supported. The response SHALL use the error code 551 (Option Not
Supported). This header does not apply to proxies, for the same
functionality in respect to proxies see, header Proxy-Require
(Section 13.27).
This is to make sure that the client-server interaction
will proceed without delay when all features are understood
by both sides, and only slow down if features are not
understood (as in the example below). 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" ":" feature-tag *("," feature-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
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CSeq: 303
S->C: RTSP/1.0 200 OK
CSeq: 303
In this example, "funky-feature" is the feature-tag which indicates
to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.
Proxies and other intermediary devices SHOULD ignore features that
are not understood in this field. If a particular extension requires
that intermediate devices support it, the extension should be tagged
in the Proxy-Require field instead (see Section 13.27).
13.33 RTP-Info
The RTP-Info response-header field is used to set RTP-specific
parameters in the PLAY response. For streams using RTP as transport
protocol the RTP-Info header SHALL be part of a 200 response to PLAY.
The RTP-Info response-header field is used to set RTP-specific
parameters in the PLAY response. These parameters correspond to the
synchronization source identified by the ssrc parameter of the
Transport response header in the SETUP reponse. For streams using RTP
as transport protocol the RTP-Info header SHALL be part of a 200
response to PLAY.
url: Indicates the stream URL which for which the following RTP
parameters correspond, this URL MUST be the same used in
the SETUP request for this media stream. Any relative URL
SHALL use the request URL as base URL.
seq: Indicates the sequence number of the first packet of the
stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that
originated after the seek.
rtptime: Indicates the RTP timestamp corresponding to the time
value in the Range response header. (Note: For aggregate
control, a particular stream may not actually generate a
packet for the Range time value returned or implied. Thus,
there is no guarantee that the packet with the sequence
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number indicated by seq actually has the timestamp
indicated by rtptime.) The client uses this value to
calculate the mapping of RTP time to NPT.
A mapping from RTP timestamps to NTP timestamps (wall
clock) is available via RTCP. However, this
information is not sufficient to generate a mapping
from RTP timestamps to NPT. Furthermore, in order to
ensure that this information is available at the
necessary time (immediately at startup or after a
seek), and that it is delivered reliably, this mapping
is placed in the RTSP control channel.
In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to
NTP, using initial RTCP sender reports to do the mapping,
and later reports to check drift against the mapping.
Additionally, the RTP-Info header parameter fields only apply to a |
single SSRC within a stream (the SSRC reported in the transport |
response header; see section 13.40). If there are multiple |
synchronization sources (SSRCs) present within a RTP session, RTCP |
must be used to map RTP and NTP timestamps for those sources, for |
both synchronization and drift-checking.
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
quoted-url = "url" "=" <"> needquote-url <">
safe-url = url
needquote-url = url //That contains ; or ,
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] with RFC
2732 [30] applied.
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Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
13.34 Scale
A scale value of 1 indicates normal play at the normal forward
viewing rate. If not 1, the value corresponds to the rate with
respect to normal viewing rate. For example, a ratio of 2 indicates
twice the normal viewing rate ("fast forward") and a ratio of 0.5
indicates half the normal viewing rate. In other words, a ratio of 2
has normal play time increase at twice the wallclock rate. For every
second of elapsed (wallclock) time, 2 seconds of content will be
delivered. A negative value indicates reverse direction.
Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected key frames. For audio, it may time-scale
the audio while preserving pitch or, less desirably, deliver
fragments of audio.
The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response
MUST contain the actual scale value chosen by the server.
If the server does not implement the possibility to scale, it will
not return a Scale header. A server supporting Scale operations for
PLAY SHALL indicate this with the use of the "play.scale" feature-
tags.
Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
When indicating a negative scale for a reverse playback, the Range
header must indicate a decreasing range as described in section
13.29.
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
Range: npt=15-10
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13.35 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.
Implementation by the server is OPTIONAL. The default is the bit rate
of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of
2.0 indicates that data is to be delivered twice as fast as normal. A
speed of zero is invalid. All speeds may not be possible to support.
Therefore the actual used speed MUST be included in the response.
The lack of a response header is indication of lack of support from
the server of this functionality. Support of the speed functionality
are indicated by the "play.speed" feature-tag.
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 |
presentation at a higher or lower rate is necessary. Implementors |
should keep in mind that bandwidth for the session may be negotiated |
beforehand (by means other than RTSP), and therefore re-negotiation |
may be necessary. When data is delivered over UDP, it is highly |
recommended that means such as RTCP be used to track packet loss |
rates. If the data transport is performed over public best-effort |
networks the sender SHOULD perform congestion control of the |
stream(s). This can result in that the communicated speed is |
impossible to maintain.
13.36 Server
See [H14.38], however the header syntax is here corrected.
Server = "Server" ":" ( product / comment ) *(SP (product / comment))
13.37 Session
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The Session request-header and response-header field identifies an |
RTSP session. An RTSP session is created by the server as a result of |
a successful SETUP request and in the response the session identifier |
is given to the client. The RTSP session exist until destroyed by a |
TEARDOWN or timed out by the server. |
The session identifier is chosen by the server (see Section 3.3) and |
MUST be returned in the SETUP response. Once a client receives a |
session identifier, it SHALL be included in any request related to |
that session. This means that the Session header MUST be included in |
a request using the following methods: PLAY, PAUSE, PING, and |
TEARDOWN, and MAY be included in SETUP, OPTIONS, SET_PARAMETER, |
GET_PARAMETER, and REDIRECT, and SHALL NOT be included in DESCRIBE. |
In a RTSP response the session header SHALL be included in methods, |
SETUP, PING, PLAY, and PAUSE, and MAY be included in methods, |
TEARDOWN, and REDIRECT, and if included in the request of the |
following methods it SHALL also be included in the response, OPTIONS, |
GET_PARAMETER, and SET_PARAMETER, and SHALL NOT be included in |
DESCRIBE. |
Note that RFC 2326 servers and client may in some cases not include |
or return a Session header when expected according to the above text. |
Any client or server is RECOMMENDED to be forgiving of this error if |
possible (which it is in many cases).
Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
The timeout parameter MAY be included in a response, and SHALL NOT be |
included in requests. The server uses it to indicate to the client |
how long the server is prepared to wait between RTSP commands or |
other signs of life before closing the session due to lack of |
activity (see below and Section A). The timeout is measured in |
seconds, with a default of 60 seconds (1 minute). |
The mechanisms for showing liveness of the client is, any RTSP |
request with a Session header, if RTP & RTCP is used an RTCP message, |
or through any other used media protocol capable of indicating |
liveness of the RTSP client. It is RECOMMENDED that a client does not |
wait to the last second of the timeout before trying to send a |
liveness message. The RTSP message may be lost or when using reliable |
protocols, such as TCP, the message may take some time to arrive |
safely at the receiver. To show liveness between RTSP request issued |
to accomplish other things, the following mechanisms can be used, in |
descending order of preference: |
RTCP: If RTP is used for media transport RTCP SHOULD be used. If |
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RTCP is used to report transport statistics, it SHALL also |
work as keep alive. The server can determine the client by |
used network address and port together with the fact that |
the client is reporting on the servers SSRC(s). A downside |
of using RTCP is that it only gives statistical guarantees |
to reach the server. However that probability is so low |
that it can be ignored in most cases. For example, a |
session with 60 seconds timeout and enough bitrate assigned |
to RTCP messages to send a message from client to server on |
average every 5 seconds. That client have for a network |
with 5 % packet loss, the probability to fail showing |
liveness sign in that session within the timeout interval |
of 2.4*E-16. In sessions with shorter timeout times, or |
much higher packet loss, or small RTCP bandwidths SHOULD |
also use any of the mechanisms below. |
PING: The use of the PING method is the best of the RTSP based |
methods. It has no other effects than updating the timeout |
timer. In that way it will be a minimal message, that also |
does not cause any extra processing for the server. The |
downside is that it may not be implemented. A client SHOULD |
use a OPTIONS request to verify support of the PING at the |
server. It is also possible to detect support by sending a |
PING to the server. If a 200 (OK) message is received the |
server supports it. In case a 501 (Not Implemented) is |
received it does not support PING and there is no meaning |
in continue trying. Also the reception of a error message |
will also mean that the liveness timer has not been |
updated. |
SET_PARAMETER: When using SET_PARAMETER for keep alive, no body |
SHOULD be included. This method is basically as good as |
PING, however the implementation support of the method is |
today limited. The same considerations as for PING apply |
regarding checking of support in server and proxies. |
OPTIONS: This method does also work. However it causes the |
server to perform unnecessary processing and result in |
bigger responses than necessary for the task. The reason |
for this is that the Public is always included creating |
overhead. |
Note that a session identifier identifies an RTSP session across |
transport sessions or connections. RTSP requests for a given session |
can use different URIs (Presentation and media URIs). Note, that |
there are restrictions depending on the session which URIs that are |
acceptable for a given method. However, multiple "user" sessions for |
the same URI from the same client will require use of different |
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session identifiers. |
The session identifier is needed to distinguish several |
delivery requests for the same URL coming from the same |
client. |
The response 454 (Session Not Found) SHALL be returned if the session |
identifier is invalid.
13.38 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 corresponding Supported header.
The Supported header field contains a list of feature-tags, described
in Section 3.7, that are understood by the client or server.
Supported = "Supported" ":" [feature-tag *("," feature-tag)]
Example:
C->S: OPTIONS rtsp://example.com/ RTSP/1.0
Supported: foo, bar, blech
S->C: RTSP/1.0 200 OK
Supported: bar, blech, baz
13.39 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. It also
resolves retransmission ambiguities for unreliable transport of RTSP.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
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delay = *(DIGIT) [ "." *(DIGIT) ]
13.40 Transport
The Transport request- and response- header field indicates which
transport protocol is to be used and configures its parameters such
as destination address, compression, multicast time-to-live and
destination port for a single stream. It sets those values not
already determined by a presentation description.
Transports are comma separated, listed in order of preference.
Parameters may be added to each transport, separated by a semicolon.
The Transport header field MAY also be used to change certain
transport parameters. A server MAY refuse to change parameters of an
existing stream.
The server MAY return a Transport response-header field in the
response to indicate the values actually chosen.
A Transport request header field MAY contain a list of transport
options acceptable to the client, in the form of multiple
transportspec entries. In that case, the server MUST return the
single option (transport-spec) which was actually chosen.
A transport-spec transport option may only contain one of any given
parameter within it. Parameters may be given in any order.
Additionally, it may only contain the unicast or multicast transport
parameter.
The Transport header field is restricted to describing a
single media 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:
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General parameters:
unicast / multicast: This parameter is a mutually exclusive
indication of whether unicast or multicast delivery will be
attempted. One of the two values MUST be specified. Clients
that are capable of handling both unicast and multicast
transmission MUST indicate such capability by including two
full transport-specs with separate parameters for each.
destination: The address of the stream recipient to which a
stream will be sent. The client originating the RTSP
request may specify the destination address of the stream
recipient with the destination parameter. When the
destination field is specified, the recipient may be a
different party than the originator of the request. To
avoid becoming the unwitting perpetrator of a remote-
controlled denial-of-service attack, a server SHOULD
authenticate the client originating the request and SHOULD
log such attempts before allowing the client to direct a
media stream to a recipient address not chosen by the
server. While, this is particularly important if RTSP
commands are issued via UDP, implementations cannot rely on
TCP as reliable means of client identification by itself
either.
The server SHOULD NOT allow the destination field to be set
unless a mechanism exists in the system to authorize the
request originator to direct streams to the recipient. It
is preferred that this authorization be performed by the
recipient itself and the credentials passed along to the
server. However, in certain cases, such as when recipient
address is a multicast group, or when the recipient is
unable to communicate with the server in an out-of-band
manner, this may not be possible. In these cases server may
chose another method such as a server-resident
authorization list to ensure that the request originator
has the proper credentials to request stream delivery to
the recipient.
This parameter SHALL NOT be used when src_addr and dst_addr |
is used in a transport declaration. For IPv6 addresses it |
is RECOMMENDED that they be given as fully qualified domain |
to make it backwards compatible with RFC 2326 |
implementations.
source: If the source address for the stream is different than
can be derived from the RTSP endpoint address (the server
in playback), the source address SHOULD be specified. To
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maintain backwards compatibility with RFC 2326, any IPv6
host's address must be given as a fully qualified domain
name. This parameter SHALL NOT be used when src_addr and
dst_addr is used in a transport declaration.
This information may also be available through SDP.
However, since this is more a feature of transport
than media initialization, the authoritative source
for this information should be in the SETUP response.
layers: The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses
starting at the destination address.
dest_addr: A general destination address parameter that can
contain one or more address and port pair. For each
combination of Protocol/Profile/Lower Transport the
interpretation of the address or addresses needs to be
defined. The client or server SHALL NOT use this parameter
unless both client and server has shown support. This
parameter MUST be supported by client and servers that
implements this specification. Support is indicated by the
use of the feature-tag "play.basic". This parameter SHALL
NOT be used in the same transport specification as any of
the parameters "destination", "source", "port",
"client_port", and "server_port".
The same security consideration that are given for the
"Destination" parameter does also applies to this
parameter. This parameter can be used for redirecting
traffic to recipient not desiring the media traffic.
src_addr: A General source address parameter that can contain
one or more address and port pair. For each combination of
Protocol/Profile/Lower Transport the interpretation of the
address or addresses needs to be defined. The client or
server SHALL NOT use this parameter unless both client and
server has shown support. This parameter MUST be supported
by client and servers that implements this specification.
Support is indicated by the use the feature-tag
"play.basic". This parameter SHALL NOT be used in the same
transport specification as any of the parameters
"destination", "source", "port", "client_port", and
"server_port".
The address or addresses indicated in the src_addr
parameter SHOULD be used both for sending and receiving of
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the media streams data packet. The main reasons are two:
First by sending from the indicated ports the source
address will be known by the receiver of the packet.
Secondly, in the presence of NATs some traversal mechanism
requires either knowledge from which address and port a
packet flow is coming, or having the possibility to send
data to the sender port.
mode: The mode parameter indicates the methods to be supported
for this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY. The RECORD value was
defined in RFC 2326 and is deprecated in this
specification.
append: The append parameter was used together with RECORD and
is now deprecated.
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 11.11. The argument provides the channel
number to be used in the $ statement and MUST be present.
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, e.g. for RTP with RTCP. The
channel number given in the request are only a guidance
from the client to the server on what channel number(s) to
use. The server MAY set any valid channel number in the
response. The declared channel(s) are bi-directional, so
both end-parties MAY send data on the given channel. One
example of such usage is the second channel used for RTCP,
where both server and client sends RTCP packets on the same
channel.
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:
These parameters are MAY only be used if the media transport protocol
is RTP.
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port: This parameter provides the RTP/RTCP port pair for a
multicast session. It is should be 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 is used in a transport declaration.
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 is used in a transport declaration.
ssrc: The ssrc parameter, if included in a SETUP response,
indicates the RTP SSRC [23] value that will be used by the
media server for RTP packets within the stream. It is
expressed as an eight digit hexadecimal value. If the
server does not act as a synchronization source for stream
data (for instance, server is a translator, reflector,
etc.) the value will be the "packet sender's SSRC" that
would have been used in the RTCP Receiver Reports generated
by the server, regardless of whether the server actually
generates RTCP RRs. If there are multiple sources within
the stream, the ssrc parameter only indicates the value for
a single synchronization source. Other sources must be
deduced from the actual RTP/RTCP stream.
The functionality of specifying the ssrc parameter in a
SETUP request is deprecated as it is incompatible with the
specification of RTP in RFC 3550. If the parameter is
included in the transport header of a SETUP request, the
server MAY ignore it, and choose an appropriate SSRC for
the stream. The server MAY set the ssrc parameter in the
transport header of the response.
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" "=" host
/ ";" "destination" [ "=" host ]
/ ";" "interleaved" "=" channel [ "-" channel ]
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/ ";" "append"
/ ";" "ttl" "=" ttl
/ ";" "layers" "=" 1*DIGIT
/ ";" "port" "=" port-spec
/ ";" "client_port" "=" port-spec
/ ";" "server_port" "=" port-spec
/ ";" "ssrc" "=" ssrc
/ ";" "mode" "=" mode-spec
/ ";" "dest_addr" "=" addr-list
/ ";" "src_addr" "=" addr-list
/ ";" trn-parameter-extension
port-spec = port [ "-" port ]
trn-parameter-extension = par-name "=" trn-par-value
par-name = token
trn-par-value = *unreserved
ttl = 1*3(DIGIT)
ssrc = 8*8(HEX)
channel = 1*3(DIGIT)
mode-spec = <"> 1#mode <"> / mode
mode = "PLAY" / "RECORD" / token
addr-list = quoted-host-port *("/" quoted-host-port)
quoted-host-port = <"> host [":" port]<">
host = see chapter 16
port = see chapter 16
The combination of transport protocol, profile and lower transport
needs to be defined. A number of combinations are defined in the
appendix B.
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
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"
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13.41 Unsupported
The Unsupported response-header field lists the features not
supported by the server. In the case where the feature was specified
via the Proxy-Require field (Section 13.27), if there is a proxy on
the path between the client and the server, the proxy MUST send a
response message with a status code of 551 (Option Not Supported).
The request SHALL NOT be forwarded.
See Section 13.32 for a usage example.
Unsupported = "Unsupported" ":" feature-tag *("," feature-tag)
13.42 User-Agent
See [H14.43] for explanation, however the syntax is clarified due to
an error in RFC 2616. A Client SHOULD include this header in all RTSP
messages it sends.
User-Agent = "User-Agent" ":" ( product / comment ) 0*(SP
(product / comment)
13.43 Vary
See [H14.44]
13.44 Via
See [H14.45].
13.45 WWW-Authenticate
See [H14.47].
14 Caching
In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE.
(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.
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On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up-to-date by
issuing a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is not
up-to-date, it modifies the SETUP transport parameters as appropriate
and forwards the request to the origin server. Subsequent control
commands such as PLAY or PAUSE then pass the proxy unmodified. The
proxy delivers the continuous media data to the client, while
possibly making a local copy for later reuse. The exact behavior
allowed to the cache is given by the cache-response directives
described in Section 13.9. A cache MUST answer any DESCRIBE requests
if it is currently serving the stream to the requestor, as it is
possible that low-level details of the stream description may have
changed on the origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
through" variety. Rather than retrieving the whole resource from the
origin server, the cache simply copies the streaming data as it
passes by on its way to the client. Thus, it does not introduce
additional latency.
To the client, an RTSP proxy cache appears like a regular media
server, to the media origin server like a client. Just as an HTTP
cache has to store the content type, content language, and so on for
the objects it caches, a media cache has to store the presentation
description. Typically, a cache eliminates all transport-references
(that is, multicast information) from the presentation description,
since these are independent of the data delivery from the cache to
the client. Information on the encodings remains the same. If the
cache is able to translate the cached media data, it would create a
new presentation description with all the encoding possibilities it
can offer.
15 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.
15.1 Media on Demand (Unicast)
Client C requests a movie from media servers A ( audio.example.com )
and V (video.example.com ). The media description is stored on a web
server W. The media description contains descriptions of the
presentation and all its streams, including the codecs that are
available, dynamic RTP payload types, the protocol stack, and content
information such as language or copyright restrictions. It may also
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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
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
e=adm@example.com
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
User-Agent: PhonyClient/1.2
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
User-Agent: PhonyClient/1.2
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
CSeq: 2
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User-Agent: PhonyClient/1.2
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
User-Agent: PhonyClient/1.2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/1.0 200 OK
CSeq: 2
User-Agent: PhonyClient/1.2
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
User-Agent: PhonyClient/1.2
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
User-Agent: PhonyClient/1.2
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.
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15.2 Streaming of a Container file
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
control all the streams via a single control message using an
aggregate URL.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URL to
the container file.
C->M: DESCRIBE rtsp://example.com/twister RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 OK
CSeq: 1
Date: 23 Jan 1997 15:35:06 GMT
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
e=adm@example.com
a=control:rtsp://example.com/twister
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t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://example.com/twister/audio
m=video 0 RTP/AVP 26
a=control:rtsp://example.com/twister/video
C->M: SETUP rtsp://example.com/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://example.com/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://example.com/twister RTSP/1.0
CSeq: 4
Range: npt=0-
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 4
Session: 12345678
Range: npt=0-
RTP-Info: url=rtsp://example.com/twister/video;
seq=12345;rtptime=3450012,
url=rtsp://example.com/twister/audio;
seq=54321;rtptime=2876889
C->M: PAUSE rtsp://example.com/twister/video RTSP/1.0
CSeq: 5
Session: 12345678
M->C: RTSP/1.0 460 Only aggregate operation allowed
CSeq: 5
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C->M: PAUSE rtsp://example.com/twister RTSP/1.0
CSeq: 6
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 6
Session: 12345678
C->M: SETUP rtsp://example.com/twister RTSP/1.0
CSeq: 7
Transport: RTP/AVP;unicast;client_port=10000
Session: 12345678
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 not allowed
as this session is set up for aggregated control. 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.
15.3 Single Stream Container Files
Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients SHOULD use the rules set forth in the session
description for request URLs, rather than assuming that a consistent
URL may always be used throughout. Here's an example of how a multi-
stream server might expect a single-stream file to be served:
C->S DESCRIBE rtsp://foo.com/test.wav RTSP/1.0
Accept: application/x-rtsp-mh, application/sdp
CSeq: 1
S->C RTSP/1.0 200 OK
CSeq: 1
Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp
Content-length: 48
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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;
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
Range: npt=0-600
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
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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:
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.
15.4 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
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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;
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
Range:npt=now-
16 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as defined in RFC 2234 [14]. Also the "#" rule from RFC 2616 [26] is
also defined and used in this syntax description.
16.1 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
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(octets 0 - 31) and DEL (127)>
CR = <US-ASCII CR, carriage return (13)>
LF = <US-ASCII LF, linefeed (10)>
SP = <US-ASCII SP, space (32)>
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
16.2 RTSP Protocol Definition
16.2.1 Message Syntax
RTSP-message = Request / Response ; RTSP/1.0 messages
generic-message = start-line
*(message-header CRLF)
CRLF
[ message-body ]
start-line = Request-Line / Status-Line
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Request = Request-Line ; Section 6.1 *(
general-header ; Section 5 / request-header ;
Section 6.2 / entity-header ) ; Section 8.1
CRLF [
message-body ] ; Section 4.3 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
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
Method = "DESCRIBE" ; Section 11.2
/ "GET_PARAMETER" ; Section 11.7
/ "OPTIONS" ; Section 11.1
/ "PAUSE" ; Section 11.5
/ "PLAY" ; Section 11.4
/ "PING" ; Section 11.10
/ "REDIRECT" ; Section 11.9
/ "SETUP" ; Section 11.3
/ "SET_PARAMETER" ; Section 11.8
/ "TEARDOWN" ; Section 11.6
/ extension-method
extension-method = token
Request-URI = "*" / absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
Status-Code = "100" ; Continue
/ "200" ; OK
/ "201" ; Created
/ "250" ; Low on Storage Space
/ "300" ; Multiple Choices
/ "301" ; Moved Permanently
/ "302" ; Moved Temporarily
/ "303" ; See Other
/ "304" ; Not Modified
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/ "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
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
general-header = Cache-Control ; Section 13.9
/ Connection ; Section 13.10
/ CSeq ; Section 13.17
/ Date ; Section 13.18
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/ Timestamp ; Section 13.39
/ Via ; Section 13.44
request-header = Accept ; Section 13.1
/ Accept-Encoding ; Section 13.2
/ Accept-Language ; Section 13.3
/ Authorization ; Section 13.6
/ Bandwidth ; Section 13.7
/ Blocksize ; Section 13.8
/ From ; Section 13.20
/ If-Modified-Since ; Section 13.23
/ Proxy-Require ; Section 13.27
/ Range ; Section 13.29
/ Referer ; Section 13.30
/ Require ; Section 13.32
/ Scale ; Section 13.34
/ Session ; Section 13.37
/ Speed ; Section 13.35
/ Supported ; Section 13.38
/ Transport ; Section 13.40
/ User-Agent ; Section 13.42
response-header = Accept-Ranges ; Section 13.4
/ Location ; Section 13.25
/ Proxy-Authenticate ; Section 13.26
/ Public ; Section 13.28
/ Range ; Section 13.29
/ Retry-After ; Section 13.31
/ RTP-Info ; Section 13.33
/ Scale ; Section 13.34
/ Session ; Section 13.37
/ Server ; Section 13.36
/ Speed ; Section 13.35
/ Transport ; Section 13.40
/ Unsupported ; Section 13.41
/ Vary ; Section 13.43
/ WWW-Authenticate ; Section 13.45
rtsp_URL = ( "rtsp:" / "rtspu:" / "rtsps" )
"//" host [ ":" port ] [ abs_path ] [ "#" fragment ]
host = As defined by RFC 2732 [30]
abs_path = As defined by RFC 2396 [22]
port = *DIGIT
smpte-range = smpte-type "=" smpte-range-spec
smpte-range-spec = ( smpte-time "-" [ smpte-time ] ) / ( "-" smpte-time )
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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 ] ]
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
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 >
utc-time = 6DIGIT [ "." fraction ]; < HHMMSS.fraction >
fraction = 1*DIGIT
feature-tag = token
16.2.2 Header Syntax
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" "=" host
/ ";" "destination" [ "=" host ]
/ ";" "interleaved" "=" channel [ "-" channel ]
/ ";" "append"
/ ";" "ttl" "=" ttl
/ ";" "layers" "=" 1*DIGIT
/ ";" "port" "=" port-spec
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/ ";" "client_port" "=" port-spec
/ ";" "server_port" "=" port-spec
/ ";" "ssrc" "=" ssrc
/ ";" "client_ssrc" "=" ssrc
/ ";" "mode" "=" mode-spec
/ ";" "dest_addresses" "=" addr-list
/ ";" "src_addresses" "=" addr-list
/ ";" trn-parameter-extension
port-spec = port [ "-" port ]
trn-parameter-extension = par-name "=" trn-par-value
par-name = token
trn-par-value = *unreserved
ttl = 1*3(DIGIT)
ssrc = 8*8(HEX)
channel = 1*3(DIGIT)
mode-spec = <"> 1#mode <"> / mode
mode = "PLAY" / "RECORD" / token
addr-list = quoted-host-port *("/" quoted-host-port)
quoted-host-port = <"> host [":" port]<">
17 Security Considerations
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [H15]
apply. Specifically, please note the following:
Authentication Mechanisms: RTSP and HTTP share common
authentication schemes, and thus should follow the same
prescriptions with regards to authentication . See chapter
15.1 of [2] for client authentication issues, and chapter
15.2 of [2] for issues regarding support for multiple
authentication mechanisms. Also see [H15.6].
Abuse of Server Log Information: RTSP and HTTP servers will
presumably 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
sensitive than that normally transmitted via HTTP.
Therefore, all of the precautions regarding the protection
of data privacy and user privacy apply to implementors of
RTSP clients, servers, and proxies. See [H15.1.2] for
further details.
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Attacks Based On File and Path Names: Though RTSP URLs are
opaque handles that do not necessarily have file system
semantics, it is anticipated that many implementations will
translate portions of the request URLs directly to file
system calls. In such cases, file systems SHOULD follow the
precautions outlined in [H15.5], such as checking for ".."
in path components.
Personal Information: RTSP clients are often privy to the same
information that HTTP clients are (user name, location,
etc.) and thus should be equally. See [H15.1] for further
recommendations.
Privacy Issues Connected to Accept Headers: Since may of the
same "Accept" headers exist in RTSP as in HTTP, the same
caveats outlined in [H15.1.4] with regards to their use
should be followed.
DNS Spoofing: Presumably, given the longer connection times
typically associated to RTSP sessions relative to HTTP
sessions, RTSP client DNS optimizations should be less
prevalent. Nonetheless, the recommendations provided in
[H15.3] 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 invalidate 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
[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
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this case, this is not always useful in prevention of more
attacks or ascertaining the attackers identity. Thus, an
RTSP server SHOULD only allow client-specified destinations
for RTSP-initiated traffic flows if the server has verified
the client's identity, either against a database of known
users using RTSP authentication mechanisms (preferably
digest authentication or stronger), or other secure means.
Session hijacking: Since there is no or little relation between
a transport layer connection and an RTSP session, it is
possible for a malicious client to issue requests with
random session identifiers which would affect unsuspecting
clients. The server SHOULD use a large, random and non-
sequential session identifier to minimize the possibility
of this kind of attack.
Authentication: Servers SHOULD implement both basic and digest
[6] authentication. In environments requiring tighter
security for the control messages, transport layer
mechanisms such as TLS (RFC 2246 [27]) SHOULD be used.
Stream issues: RTSP only provides for stream control. Stream
delivery issues are not covered in this section, nor in the
rest of this draft. RTSP implementations will most likely
rely on other protocols such as RTP, IP multicast, RSVP and
IGMP, and should address security considerations brought up
in those and other applicable specifications.
Persistently suspicious behavior: RTSP servers SHOULD return
error code 403 (Forbidden) upon receiving a single instance
of behavior which is deemed a security risk. RTSP servers
SHOULD also be aware of attempts to probe the server for
weaknesses and entry points and MAY arbitrarily disconnect
and ignore further requests clients which are deemed to be
in violation of local security policy.
18 IANA Considerations
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
register. See also the section 1.6 "Extending RTSP". There is also a
IANA registration of two SDP attributes.
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".
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A registration request to IANA MUST contain the following
information:
o A name of the item to register according to the rules
specified by the intended registry.
o Indication of who has change control over the feature (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium or a particular company or group of
companies);
o A reference to a further description, if available, for
example (in order of preference) an RFC, a published standard,
a published paper, a patent filing, a technical report,
documented source code or a computer manual;
o For proprietary features, contact information (postal and
email address);
18.1 Feature-tags
18.1.1 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
contains named entries representing certain functionality.
The usage of feature-tags is explained in section 10 and 11.1.
18.1.2 Registering New Feature-tags with IANA
The registering of feature-tags is done on a first come, first served
basis.
The name of the feature 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, or control characters. The |
registration SHALL indicate if the feature tag applies to servers |
only, proxies only or both server and proxies. Any proprietary |
feature SHALL have as the first part of the name a vendor tag, which |
identifies the organization.
18.1.3 Registered entries
The following feature-tags are in this specification defined and
hereby registered. The change control belongs to the Authors and the
IETF MMUSIC WG.
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play.basic: The minimal implementation for playback operations |
according to section D. Applies for both servers and |
proxies. |
play.scale: Support of scale operations for media playback. |
Applies only for servers. |
play.speed: Support of the speed functionality for playback. |
Applies only for servers |
18.2 RTSP Methods
18.2.1 Description
What a method is, is described in section 11. Extending the protocol
with new methods allow for totally new functionality.
18.2.2 Registering New Methods with IANA
A new method MUST be registered through an IETF standard track
document. 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:
o A method name which follows the BNF rules for methods.
o 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.
o A list or table specifying which of the registered headers
that are allowed to use with the method in request or/and
response.
o Describe how the method relates to network proxies.
18.2.3 Registered Entries
This specification, RFCXXXX, registers 10 methods: DESCRIBE,
GET_PARAMETER, OPTIONS, PAUSE, PING, PLAY, REDIRECT, SETUP,
SET_PARAMETER, and TEARDOWN.
18.3 RTSP Status Codes
18.3.1 Description
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A status code is the three digit numbers used to convey information
in RTSP response messages, see 7. The number space is limited and
care should be taken not to fill the space.
18.3.2 Registering New Status Codes with IANA
A new status code can only be registered by an IETF standards track
document. A specification for a new status code MUST specify the
following:
o The requested number.
o A description what the status code means and the expected
behavior of the sender and receiver of the code.
18.3.3 Registered Entries
RFCXXX, registers the numbered status code defined in the BNF entry
"Status-Code" except "extension-code" in section 7.1.1.
18.4 RTSP Headers
18.4.1 Description
By specifying new headers a method(s) can be enhanced in many
different ways. An unknown header will be ignored by the receiving
entity. If the new header is vital for a certain functionality, a
feature-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 feature-tag.
18.4.2 Registering New Headers with IANA
A public available specification is required to register a header.
The specification SHOULD be a standards document, preferable an IETF
RFC.
The specification MUST contain the following information:
o The name of the header.
o A BNF specification of the header syntax.
o A list or table specifying when the header may be used,
encompassing all methods, their request or response, the
direction (C -> S or S -> C).
o How the header shall be handled by proxies.
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o A description of the purpose of the header.
18.4.3 Registered entries
All headers specified in section 13 in RFCXXXX are to be registered.
Furthermore the following RTSP headers defined in other
specifications are registered:
o x-wap-profile defined in [35].
o x-wap-profile-diff defined in [35].
o x-wap-profile-warning defined in [35].
o x-predecbufsize defined in [35].
o x-initpredecbufperiod defined in [35].
o x-initpostdecbufperiod defined in [35].
Note: The use of "X-" is NOT RECOMMENDED but the above headers
in the register list was defined prior to the clarification.
18.5 Transport Header registries
The transport header contains a number of parameters which have
possibilities for future extensions. Therefore registries for these
must be defined.
18.5.1 Transport Protocols
A registry for the parameter transport-protocol shall be defined with
the following rules:
o Registering requires public available standards specification.
o A contact person or organization with address and email.
o A value definition that are following the BNF token
definition.
o A describing text that explains how the registered value are
used in RTSP.
This specification register 1 value:
o Use of the RTP [23] protocol for media transport. The usage
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is explained in RFC XXXX, appendix B.1.
18.5.2 Profile
A registry for the parameter profile shall be defined with the
following rules:
o Registering requires public available standards specification.
o A contact person or organization with address and email.
o A value definition that are following the BNF token
definition.
o A definition of which Transport protocol(s) that this profile
is valid for.
o A describing text that explains how the registered value are
used in RTSP.
o The "RTP profile for audio and video conferences with minimal
control" [1] MUST only be used when the transport headers
transport-protocol is "RTP".
18.5.3 Lower Transport
A registry for the parameter lower-transport shall be defined with
the following rules:
o Registering requires public available standards specification.
o A contact person or organization with address and email.
o A value definition that are following the BNF token
definition.
o A describing text that explains how the registered value are
used in RTSP. This includes
o Indicates the use of the "User datagram protocol" [7] for
media transport.
o Indicates the use Transmission control protocol [9] for media
transport.
18.5.4 Transport modes
A registry for the transport parameter mode shall be defined with the
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following rules:
o Registering requires a IETF standard tracks document.
o A contact person or organization with address and email.
o A value definition that are following the BNF token
definition.
o A describing text that explains how the registered value are
used in RTSP.
o See RFC XXXX.
o See RFC XXXX.
18.6 Cache Directive Extensions
There exist a number of cache directives which can be sent in the
Cache-Control header. A registry for this cache directives shall be
defined with the following rules:
o Registering requires a IETF standard tracks document.
o A registration shall name a contact person.
o Name of the directive and a definition of the value, if any.
o A describing text that explains how the cache directive is
used for RTSP controlled media streams.
18.7 SDP attributes
This specification defines two SDP [24] attributes that it is requested
that IANA register.
SDP Attribute ("att-field"):
Attribute name: range
Long form: Media Range Attribute
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
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Values: See ABNF definition.
Attribute name: control
Long form: RTSP control URL
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: Absolute or Relative URLs.
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 |
one 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 three 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: Session is ready to start playing. |
Play: Session is playing, i.e. sending media stream data in the |
direction S -> C. |
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. |
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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 the variable is not mandated. |
A.3 Abbreviations |
To make the state tables more compact a number of abbreviations are |
used, which are explained below. |
IFI: IF Implemented. |
md: Media |
PP: Pause Point, the point in the presentation time line at |
which the presentation was paused. |
Prs: Presentation, the complete multimedia presentation. |
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 |
exceptions shall be given a response according to the response |
column. If 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 3rr will result in that the session is ended and its |
state is changed to Init. A response code of 304 results in no state |
change. However there exist restrictions to when a 3xx response may |
be used. A 5xx response SHALL not result in any change of the session |
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state, except if the error is not possible to recover from. A |
unrecoverable error SHALL result the ending of the session. As it in |
the general case can't be determined if it was a unrecoverable error |
or not the client will be required to test. In the case that the next |
request after a 5xx is responded with 454 (Session Not Found) the |
client SHALL assume that the session has been ended. |
The server will timeout the session after the period of time |
specified 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 3rr 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
Table 6: None state-machine changing events
The methods in Table 6 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. |
Action Requisite New State Response
________________________________________________
SETUP Ready NRM=1, RP=0.0
SETUP Needs Redirect Init 3rr Redirect
Table 7: State: Init
The initial state of the state machine, see Table 7 can only be left |
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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 3rr response. |
Action Requisite New State Response
______________________________________________________________________
SETUP New URL Ready
NRM+=1 SETUP Setten up URL
Ready Change transport param. TEARDOWN Prs
URL,NRM>1 Init No session hdr TEARDOWN md
URL,NRM=1 Init No 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 PAUSE Prs URL
Ready Return PP S -> C:REDIRECT Range
hdr Ready Set RedP S -> C:REDIRECT no
range hdr Init Session is removed
Timeout
Init RedP
reached Ready TEARDOWN of session
Table 8: State: Ready
In the Ready state, see Table 8, some of the actions are depending on |
the number of media streams (NRM) 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 the presentations |
URI the whole session is torn down. If a media URL is used in the |
TEARDOWN request and more than one media exist in the session, the |
session will remain and a session header MUST be returned in the |
response. If only a single media stream remains in the session when |
performing a TEARDOWN with a media URL the session is removed. The |
number of media streams remaining 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
________________________________________________________________________
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
Range Hdr. 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 Play
455 SETUP Setuped URL
Play 455 SETUP Setuped URL,
IFI Play Change transport param. TEARDOWN Prs
URL,NRM>1 Init No session hdr TEARDOWN
md URL,NRM=1 Init No Session hdr, NRM=0
TEARDOWN md URL Play
455 S -> C:REDIRECT Range hdr
Play Set RedP S -> C:REDIRECT no range
hdr Init Session is removed RedP
reached Play TEARDOWN of session
Timeout Init Stop Media
playout
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 |
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 |
automatic 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 |
transitions it must be prepared for getting error message as the |
state may or may not have changed.
B Media Transport Alternatives
This chapter defines how certain combinations of protocols, profiles
and lower transports are used. This includes the usage of the
Transport header's general source and destination parameters
"src_addresses" and "dst_addresses".
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B.1 RTP
This section defines the interaction and needed media transport
signalling in regards to the RTP protocol [23].
RTSP allows media clients to control selected, non-contiguous
sections 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 13.34), 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 13.35) 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 13.33) header
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provides the first sequence number of the next segment.
Note that more than one SSRC MAY be sent in the media stream. |
However without further extensions RTSP can't synchronize more than |
the single one indicated in the Transport header. In these cases RTCP |
needs to be used for synchronization. |
The transmission of RTCP SHOULD be done the whole time from RTP |
session creation by the SETUP request and continue until the session |
is removed by the TEARDOWN request, that is including during the |
period in Ready state. This ensures that neither end part times out |
the other. Thus ensuring liveness information to both end-points, |
allow for packet-loss detection at the end of playout period, ensure |
media synchronization in cases multiple SSRCs are used, and to keep |
synchronization information updated allowing for correct synch also |
at the beginning of a stream before any PLAY response has arrived. |
The sending of the RTCP BYE message is connected to the existence of |
the RTCP session and the SSRC. Therefore a client or server SHALL not |
send an RTCP BYE message until it has finished using an SSRC. A |
server SHOULD not stop using an SSRC until the RTP session is |
terminated. This is due to that a SSRC that has been used has an |
established synchronization context that ensures synchronization also |
if the PLAY response is late, for an subsequent PLAY request after |
the first one. Changing the server side SSRC will also prevent the |
server from synchronizing that new SSRC within RTSP as it is |
connected to the one declared in the SSRC parameter in the Transport |
header.
Below the available RTP profiles and lower layer transports are given
together with the necessary rules on how to signal that combination.
B.1.1 AVP
The usage of the "RTP Profile for Audio and Video Conferences with
Minimal Control" [1] when using RTP for media transport over
different lower layer transport protocols are defined below in
regards to RTSP.
On such case is defined within this document, the use of embedded
(interleaved) binary data as defined in section 11.11. The usage of
this method is indicated by include the "interleaved" parameter.
When using embedded binary data the "src_addresses" and
"dst_addresses" SHALL NOT be used. This addressing and multiplexing
is used as defined with use of channel numbers and the interleaved
parameter.
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B.1.2 AVP/UDP
This part descibes sending of RTP [23] over lower transport layer UDP
[7] according to the profile "RTP Profile for Audio and Video
Conferences with Minimal Control" defined in RFC 3551 [1].
This profiles requires that one or two uni- or bi-directional UDP
flows per media stream. The first UDP flow is for RTP and the second
is for RTCP. Embedded (interleaved) data when RTSP messages is
transported over UDP SHOULD NOT be performed.
The RTP/UDP and RTCP/UDP flows can be established in two ways using
the Transport header's parameters. The way provided in RFC 2326 was
to use the necessary parameters from the set of "source",
"destination", "client_port", and "server_port". This has the
advantage of being compatible with all RTP capable RTSP servers and
clients. However this method does not provide a possibility to
specify non-continues port ranges for RTP and RTCP. The other way is
to use the parameters "src_addresses", and "dst_addresses". This
method provides total flexibility in specifying address and port
number for each transport flow. However the disadvantage is that it
is not supported by non-updated clients, i.e. clients not supporting
the "play.basic" feature-tag.
When using the "source", "destination", "client_port", and
"server_port" the packets are be addressed in the following way for
media playback:
o RTP/UDP packet from the server to the client SHALL be sent to
the address specified in the "destination" parameter and first
even port number given in client_port range. If there is only
a single port number given that MUST be given.
o The server SHOULD send its RTP/UDP packets from the address
specified in "source" parameter and from the first even port
number specified in "server_port" parameter.
o If there is specified a range in "client_port" parameter that
contains at least two port numbers, the RTCP/UDP packets from
server to client SHALL be sent to address specified in the
"destination" parameter and first odd port number part of the
range specified in the client_port parameter.
o The Server SHOULD send its RTCP/UDP packets from the address
specified in "source" parameter and from the first odd port
number specified in "server_port" parameter.
o RTCP/UDP packets from the client to the server SHALL be sent
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to the address specified in the "source" parameter and first
odd port number given in client_port range.
o The client SHOULD send its RTCP/UDP packets from the address
specified in "destination" parameter and from the first odd
port number specified in "server_port" parameter.
The usage of "src_addresses" and "dst_addresses" parameters to
specify the address and port numbers are done in the following way
for media playback, i.e. Mode=PLAY:
o The "src_addresses" and "dst_addresses" parameters MUST
contain either 1 or 2 address and port pairs.
o Each address and port pair MUST contain both and address and a
port number.
o The first address and port pair given in either of the
parameters applies to the RTP stream. The second address and
port pair if present applies to the RTCP stream.
o The RTP/UDP packets from the server to the client SHALL be
sent to the address and port given by first address and port
pair of the "dst_addresses" parameter.
o The RTCP/UDP packets from the server to the client SHALL be
sent to the address and port given by the second address and
port pair of the "dst_addresses" parameter. If no second pair
is given RTCP SHALL NOT be sent.
o The RTCP/UDP packets from the client to the server SHALL be
sent to the address and port given by the second address and
port pair of the "dst_addresses" parameter. If no second pair
is given RTCP SHALL NOT be sent.
o RTP and RTCP Packets SHOULD be sent from the corresponding
receiver port, i.e. RTCP packets from server should be sent
from the "src_addresses" parameters second address port pair.
B.1.3 AVP/TCP
Note that this combination is not yet defined using sperate TCP
connections. However the use of embedded (interleaved) binary data
transported on the RTSP connection is possible as specified in
section 11.11. When using this declared combination of interleaved
binary data the RTSP messages MUST be transported over TCP.
A possible future for this profile would be to define the use of a
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combination of the two drafts "Connection-Oriented Media Transport in
SDP" [36] and "Framing RTP and RTCP Packets over Connection-Oriented
Transport" [37].
B.2 Future Additions
It is the intention that any future protocol or profile regarding
both for media delivery and lower transport should be easy to add to
RTSP. This chapter provides the necessary steps that needs to be
meet.
The following things needs to be considered when adding a new
protocol of profile for use with RTSP:
o The protocol or profile needs to define a name tag
representing it. This tag is required to be a ABNF "token" to
be possible to use in the Transport header specification.
o The useful combinations of protocol/profile/lower-layer needs
to be defined and for each combination declare the necessary
parameters to use in the Transport header.
o For new media protocols the interaction with RTSP needs to be
addressed. One important factor will be the media
synchronization.
See the IANA section ( 18) on how to register the necessary
attributes.
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 |
typically returned in reply to a DESCRIBE request on a URL from a |
server to a client, or received via HTTP from a server to a client. |
This appendix describes how an SDP file determines the operation of |
an RTSP session. SDP as is 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). However the |
SDP extension "Grouping of Media Lines in the Session Description |
Protocol (SDP)" [40] may provide such functionality depending on |
need. Also future grouping semantics may in the future be developed.
C.1 Definitions
The terms "session-level", "media-level" and other key/attribute
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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 |
controlling that particular media stream. If found at the session |
level, the attribute indicates the URL for aggregate control |
(presentation URL). The session level URL SHALL be different from any |
media level URI. The presence of a session level control attribute |
SHALL be interpreted as support for aggregated control. The control |
attribute SHALL be present on media level unless the presentation |
only contains a single media stream, in which case the attribute MAY |
only be present on the session level.
control-attribute = "a=" "control" ":" url
Example:
a=control:rtsp://example.com/foo
This attribute MAY contain either relative and absolute URLs,
following the rules and conventions set out in RFC 2396 [22].
Implementations SHALL 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 SHALL |
be treated as if it were an empty embedded URL, and thus inherit the |
entire base URL. |
For SDP retrieved from a container file, there are certain things to |
consider. Lets say that the container file has the following URL: |
"rtsp://example.com/container.mp4". A media level relative URL needs |
to contain the file name container.mp4 in the beginning to be |
resolved correctly relative to the before given URL. An alternative |
if one does not desire to enter the container files name is to ensure |
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that the base URL for the SDP document becomes: |
"rtsp://example.com/container.mp4/", i.e. an extra trailing slash. |
When using the URL resolution rules in RFC 2396 that will resolve |
correctly. However, please note that if the session level control URL |
is a *, that control URL will be equal to |
"rtsp://example.com/container.mp4/" and include the slash. |
C.1.2 Media Streams |
The "m=" field is used to enumerate the streams. It is expected that |
all the specified streams will be rendered with appropriate |
synchronization. If the session is a multicast, the port number |
indicated SHOULD be used for reception. The client MAY try to |
override the destination port, through the Transport header. The |
servers MAY allow this, the response will indicate if allowed or not. |
If the session is unicast, the port number is the ones RECOMMENDED by |
the server to the client, about which receiver ports to use; the |
client MUST still include its receiver ports in its SETUP request. |
The client MAY ignore this recommendation. If the server has no |
preference, it SHOULD set the port number value to zero. |
The "m=" lines contain information about what transport protocol, |
profile, and possibly lower-layer shall be used for the media stream. |
The combination of transport, profile and lower layer, like |
RTP/AVP/UDP needs to be defined for how to be used with RTSP. The |
currently defined combinations are defined in section B, further |
combinations MAY be specified. |
TODO: Write something about the usage of Grouping of media line, RFC |
3388 [40]. |
Example:
m=audio 0 RTP/AVP 31
C.1.3 Payload Type(s)
The payload type(s) are specified in the "m=" field. In case the |
payload type is a static payload type from RFC 3551 [1], no other |
information is required. In case it is a dynamic payload type, the |
media attribute "rtpmap" is used to specify what the media is. The |
"encoding name" within the "rtpmap" attribute may be one of those |
specified in RFC 3551 (Sections 5 and 6), or an MIME type registered |
with IANA, or an experimental encoding with a "X-" prefix as |
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specified in SDP (RFC 2327 [24]). Codec-specific parameters are not |
specified in this field, but rather in the "fmtp" attribute described |
below. |
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 some of the |
format specific parameters may be specified outside of the fmtp |
parameters, like for example the "ptime" attribute for most audio |
encodings. |
C.1.5 Range of Presentation |
The "a=range" attribute defines the total time range of the stored |
session or an individual media. Non-seekable Live sessions can be |
indicated, while the length of live sessions can be deduced from the |
"t" and "r" SDP parameters. |
The attribute is both a session and a media level attribute. For |
presentations that contains media streams of the same durations, the |
range attribute SHOULD only be used at session-level. In case of |
different length the range attribute MUST be given at media level for |
all media, and SHOULD NOT be given at session level. If the attribute |
is present at both media level and session level the media level |
values SHALL be used.
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. Any open
ended range (start-), i.e. without stop range, is of unspecified
duration and SHALL be considered as non-seekable content unless this
property is overridden.
This attribute is defined in ABNF [14] as:
a-range-def = "a" "=" "range" ":" ranges-specifier CRLF
Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203
Non seekable stream of unknown duration:
a=range:npt=0-
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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. 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.
C.1.7 Connection Information
In SDP, the "c=" field contains the destination address for the media |
stream. For a media destination address that is a IPv6 one, the SDP |
extension defined in [38] needs to be used. For on-demand unicast |
streams and some multicast streams, the destination address MAY be |
specified by the client via the SETUP request, thus overriding any |
specified address. To identify streams without a fixed destination |
address, where the client must specify a destination address, the |
"c=" field SHOULD be set to a null value. For addresses of type |
"IP4", this value SHALL be "0.0.0.0", and for type "IP6", this value |
SHALL be "0:0:0:0:0:0:0:0", i.e. the unspecified address according to |
RFC 3513 [39].
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 13.22) 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
functionality. However, it does so in a manner that would
put constraints on servers that need to support multiple
session description types other than SDP for the same piece
of media content.
C.2 Aggregate Control Not Available
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If a presentation does not support aggregate control no session level |
"a=control:" attribute is specified. For a SDP with multiple media |
sections specified, each section will have its own 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
e=adm@example.com
c=IN IP4 0.0.0.0
t=0 0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid
Note that the position of the control URL in the description implies
that the client establishes separate RTSP control sessions to the
servers audio.com and video.com
It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media
client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.
C.3 Aggregate Control Available
In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control:" attributes, which are used to specify the stream URLs,
and a session-level "a=control:" attribute which is used as the
request URL for aggregate control. If the media-level URL is
relative, it is resolved to absolute URLs according to Section C.1.1
above.
Example: |
C->M: DESCRIBE rtsp://example.com/movie RTSP/1.0 |
CSeq: 1 |
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M->C: RTSP/1.0 200 OK |
CSeq: 1 |
Date: 23 Jan 1997 15:35:06 GMT |
Content-Type: application/sdp |
Content-Base: rtsp://example.com/movie/ |
Content-Length: 164 |
v=0 |
o=- 2890844256 2890842807 IN IP4 204.34.34.32 |
s=I contain |
i=<more info> |
e=adm@example.com |
c=IN IP4 0.0.0.0 |
t=0 0 |
a=control:* |
m=video 8002 RTP/AVP 31 |
a=control:trackID=1 |
m=audio 8004 RTP/AVP 3 |
a=control:trackID=2 |
In this example, the client is required to establish a single RTSP |
session to the server, and uses the URLs |
rtsp://example.com/movie/trackID=1 and |
rtsp://example.com/movie/trackID=2 to set up the video and audio |
streams, respectively. The URL rtsp://example.com/movie/ , which is |
resolved from the "*", controls the whole presentation (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. |
C.4 RTSP external SDP delivery |
There are some considerations that needs to be made when the session |
description is delivered to client outside of RTSP, for example in |
HTTP or email. |
First of all the SDP needs to contain absolute URIs, relative will in |
most cases not work as the delivery will not correctly forward the |
base URI. And as SDP might be temporarily stored on file system |
before being loaded into a RTSP capable client, thus if possible to |
transport the base URI it still would need to be merged into the |
file. |
The writing of the SDP session availability information, i.e. "t=" |
and "r=", needs to be carefully considered. When the SDP is fetched |
H. Schulzrinne et. al. [Page 127]
Internet Draft RTSP October 27, 2003
by the DESCRIBE method it is with very high probability that the it |
is valid. However the same are much less certain for SDPs distributed |
using other methods. Therefore the publisher of the SDP should take |
care to follow the recommendations about availability in the SDP |
specification [24].
D Minimal RTSP implementation
D.1 Client
A client implementation MUST be able to do the following :
o Generate the following requests: SETUP, TEARDOWN, PLAY.
o Include the following headers in requests: CSeq, Connection,
Session, Transport.
o Parse and understand the following headers in responses:
CSeq, Connection, Session, Transport, Content-Language,
Content-Encoding, Content-Length, Content-Type.
o Understand the class of each error code received and notify
the end-user, if one is present, of error codes in classes 4xx
and 5xx. The notification requirement may be relaxed if the
end-user explicitly does not want it for one or all status
codes.
o Expect and respond to asynchronous requests from the server,
such as REDIRECT. This does not necessarily mean that it
should implement the REDIRECT method, merely that it MUST
respond positively or negatively to any request received from
the server.
Though not required, the following are RECOMMENDED.
o Implement RTP/AVP/UDP as a valid transport.
o Inclusion of the User-Agent header.
o Understand SDP session descriptions as defined in Appendix C
o Accept media initialization formats (such as SDP) from
standard input, command line, or other means appropriate to
the operating environment to act as a "helper application" for
other applications (such as web browsers).
There may be RTSP applications different from those
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initially envisioned by the contributors to the RTSP
specification for which the requirements above do not make
sense. Therefore, the recommendations above serve only as
guidelines instead of strict requirements.
D.1.1 Basic Playback
To support on-demand playback of media streams, the client MUST
additionally be able to do the following:
o generate the PAUSE request;
o implement the REDIRECT method, and the Location header.
D.1.2 Authentication-enabled
In order to access media presentations from RTSP servers that require
authentication, the client MUST additionally be able to do the
following:
o recognize the 401 (Unauthorized) status code;
o parse and include the WWW-Authenticate header;
o implement Basic Authentication and Digest Authentication.
D.2 Server
A minimal server implementation MUST be able to do the following:
o Implement the following methods: SETUP, TEARDOWN, OPTIONS and
PLAY.
o Include the following headers in responses: Connection,
Content-Length, Content-Type, Content-Language, Content-
Encoding, Timestamp, Transport, Public, and Via, and
Unsupported. RTP-compliant implementations MUST also
implement the RTP-Info field.
o Parse and respond appropriately to the following headers in
requests: Connection, Proxy-Require, Session, Transport, and
Require.
Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial
implementations and/or to be a "good citizen".
o Implement RTP/AVP/UDP as a valid transport.
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o Inclusion of the Server header.
o Implement the DESCRIBE method.
o Generate SDP session descriptions as defined in Appendix C
There may be RTSP applications different from those
initially envisioned by the contributors to the RTSP
specification for which the requirements above do not make
sense. Therefore, the recommendations above serve only as
guidelines instead of strict requirements.
D.2.1 Basic Playback
To support on-demand playback of media streams, the server MUST
additionally be able to do the following:
o Recognize the Range header, and return an error if seeking is
not supported.
o Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface
features, the following are highly recommended for on-demand media
servers:
o Include and parse the Range header, with NPT units.
Implementation of SMPTE units is recommended.
o Include the length of the media presentation in the media
initialization information.
o Include mappings from data-specific timestamps to NPT. When
RTP is used, the rtptime portion of the RTP-Info field may be
used to map RTP timestamps to NPT.
Client implementations may use the presence of length
information to determine if the clip is seekable, and
visably disable seeking features for clips for which the
length information is unavailable. A common use of the
presentation length is to implement a "slider bar" which
serves as both a progress indicator and a timeline
positioning tool.
Mappings from RTP timestamps to NPT are necessary to ensure correct
positioning of the slider bar.
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D.2.2 Authentication-enabled
In order to correctly handle client authentication, the server MUST
additionally be able to do the following:
o Generate the 401 (Unauthorized) status code when
authentication is required for the resource.
o Parse and include the WWW-Authenticate header
o Implement Basic Authentication and Digest Authentication
E Open Issues
1. Should we add the header Accept-Ranges as proposed in this
specification?
2. Upon receiving a response on a REDIRECT request can the
server close the session or should it wait for a TEARDOWN
request from the client?
3. The proxy indications in the two header tables in chapter
13 needs review.
4. Should the Allow header be possible to use optional in
request or responses besides the now specified 405 error
code?
5. What text should be written on use of authorization in this
spec?
6. How does entity tags relate to the If-Match header? The
usage in SDP must also be clarified related to syntax, etc.
7. Should the Last-Modified header be required on other level
than optional?
8. How to handle range headers for negative scale playback.
9. The minimal implementation must be looked over to see if it
complies with the specification. All must and should shall
be included in the minimal. Feature-tags for these needs to
be defined. Further feature-tags needs to be discussed.
10. The list specifying which status codes are allowed on which
request methods seem to be in error and need review.
F Changes
H. Schulzrinne et. al. [Page 131]
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Compared to RFC 2326, the following issues are addressed:
o http://rtsp.org/bug448521 - "URLs in Rtp-Info need to be
quoted". URLs in RTP-info header now MAY be quoted if needed.
o http://rtsp.org/bug448525 - Syntax for SSRC should be
clarified. Require 8*8 HEX and corresponding text added.
o http://rtsp.org/bug461083 - "Body w/o Content-Length
clarification". This is clarified and any message with a
message body is required to have a Content-Length header.
o http://rtsp.org/bug477407 - Transport BNF doesn't properly
deal with semicolon and comma
o http://rtsp.org/bug477413 - Transport BNF: mode parameter
issues
o http://rtsp.org/bug477416 - "BNF error section 3.6 NPT", Added
an optional [NPT] definition. Fixed so that the same
possibilities exist for all time formats.
o http://rtsp.org/bug477421 - "When to send response". A
clarifying note in the status code chapter that when sending
400 responses, the server MUST NOT add cseq if missing.
o http://rtsp.org/bug507347 - Removal of destination redirection
in the transport header.
o http://rtsp.org/bug477404 - "Errors in table in chapter 12".
The table has been updated using the SIP structure. However
the table become to big to fit in a single page and has been
split.
o http://rtsp.org/bug477419 - Updating HTTP references to
rfc2616 by adding public, and content-base header. Section
references in header chapter updated. Known effects on RTSP
due to HTTP clarifications:
- Content-Encoding header can include encoding of type
"identity".
o http://rtsp.org/bug500803 - Rewritten the complete chapter on
the state machine.
o http://rtsp.org/bug513753 - Created a IANA section defining
several registries.
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o http://rtsp.org/bug477427 - A new subsection in the
connections chapter clarifying how the server and client may
handle transport connections.
o - Accept-Ranges response header is added. This header
clarifies which range formats that can be used for a resource.
o - Added Headers Timestamp, Via, Unsupported as required for a
minimal server implementation.
o http://rtsp.org/bug477425 - "Inconsistency between
timeformats". Fixed so that all formats has the same
capabilities as NPT.
o http://rtsp.org/bug499573 - "Incorrect grammar on Server
header". Added corrected BNF for User-Agent and Server header
as a complement to the reference.
o The definition in the introduction of the RTSP session has
been changed.
o Updated RTSP URL's and source and destination parameters in
the transport header to handle IPv6 addresses.
o All BNF definitions are updated according to the rules defined
in RFC 2234 [14].
o The use of status code 303 "See Other" has been decapitated as
it does not make sense to use in RTSP.
o Added status code 350, 351 and updated usage of the other
redirect status codes, see chapter 12.3.
o Removed Queued play (http://rtsp.org/bug508211) and
decapitated use of PLAY for keep-alive while in playing state.
o Explicitly wrote out the possibilities to use multiple ranges
to allow for editing.
o Text specifying the special behavior of PLAY for live content.
o When sending response 451 and 458 the response body should
contain the offending parameters.
o Fixed the missing definitions for the Cache-Control header.
Also added to the syntax definition the missing delta-seconds
for max-stale and min-fresh parameters.
H. Schulzrinne et. al. [Page 133]
Internet Draft RTSP October 27, 2003
o Added wording on the usage of Connection:Close for RTSP.
o Put requirement on CSeq header that the value is increased by
one for each new RTSP request.
o Added requirement that the Date header must be used for all
messages with entity. Also the Server should always include
it.
o Removed possibility to use Range header combined with Scale
header to indicate when it shall be activated, due to that it
can't work as defined. Also added rule that lack of scale
header in response indicate lack of support. Feature-tags for
scaled playback defined.
o The Speed header must now be responded to indicate support and
the actual speed going to be used. A feature-tag is defined.
Notes on congestion control was also added.
o The Supported header was borrowed from SIP to help with the
feature negotiation in RTSP.
o Clarified that the timestamp header can be used to resolve
retransmission ambiguities.
o Added two transport header parameters to be used to signal
RTCP port for server and client when not assigned in pairs.
Shall be used for NAT traversal with mechanisms like STUN. The
interoperability issue is solved by requiring a client to know
that a server supports this specification.
o Defined a IANA registries for the transport headers
parameters, transport-protocol, profile, lower-transport, and
mode.
o The OPTIONS method has been clarified on how to use the Public
and Allow headers.
o The Session header text has been expanded with a explanation
on keep alive and which methods to use.
o http://rtsp.org/bug503949 - Range header format for PAUSE is
unclear. This has been resolved by requiring a ranged pause to
only contain a single value as a beginning of an open range.
o The transport headers interleave parameter's text was made
more strict and use formal requirements levels. However no
change on how it is used was made.
H. Schulzrinne et. al. [Page 134]
Internet Draft RTSP October 27, 2003
o Added a fragment part to the RTSP URL. This seem to be
indicated by the note below the definition however it was not
part of the BNF.
o The RECORD and ANNOUNCE methods are removed as they are
lacking implementation and not considered necessary in the
core specification. Any work on these methods should be done
as a extension document to RTSP.
o The description on how rtspu and rtsps is not part of the core
specification and will require external description.
o The Transport headers RTP port parameters has been updated to
support non-continuous port numbers. Also a possibility for
the client to specify SSRC has been added.
o Clarified that RTP-Info URLs that are relative uses the
request URL as base URL. Also clarified that the URL that must
be used is the SETUP.
o Included two new general address parameters "src_addresses"
and "dst_addresses" to be used to give address source and
destination of media traffic.
o Updated the text on the transport headers "destination"
parameter regarding what security precautions the server shall
perform.
o Wrote a new chapter about how to setup different media
transport alternatives and their profiles, and lower layer
protocols. This resulted that the appendix on RTP interaction
was moved there instead in the part describing RTP. The
chapter also includes guidelines what to think of when writing
usage guidelines for new protocols and profiles.
o The embedded (interleaved) binary data and its transport
parameter was clarified to being symmetric and that it is the
server that sets the channel numbers.
o Added a new chapter describing the available mechanisms to
determine if functionality is supported, called "Capability
Handling". Renamed option-tags to feature-tags.
o Added a contributors chapter with people who has contribute
actual text to the specification.
o Added text that requires the Range to always be present in
PLAY responses. Clarified what should be sent in case of live
H. Schulzrinne et. al. [Page 135]
Internet Draft RTSP October 27, 2003
streams.
o Clarified the usage of "a=range" and how to indicate live
content that are not seekable with this header.
o Depreciated the use of the Range header "time=" parameter due
to synchronization problems in PLAY and PAUSE methods.
Note that this list does not reflect minor changes in wording or
correction of typographical errors.
A word-by-word diff from RFC 2326 can be found at
http://rtsp.org/2002/drafts
G Author Addresses
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Anup Rao
Cisco
USA
electronic mail: anrao@cisco.com
Robert Lanphier
RealNetworks
P.O. Box 91123
Seattle, WA 98111-9223
USA
electronic mail: robla@real.com
Magnus Westerlund
Ericsson AB, ERA/TVA/A
Torshamsgatan 23
SE-164 80 STOCKHOLM
SWEDEN
electronic mail: magnus.westerlund@ericsson.com
Aravind Narasimhan
Sun Microsystems, Inc.
101 Park Avenue, 3rd & 4th Floor
New York, NY
USA
H. Schulzrinne et. al. [Page 136]
Internet Draft RTSP October 27, 2003
electronic mail: aravind.narasimhan@sun.com
H Contributors
The following people has made written contribution included in the
specification:
o Tom Marshall has contributed with text about the usage of 3rr
status codes.
o Thomas Zheng has contributed with text regarding the usage of
the Range in PLAY responses.
I 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
mentioned, the following individuals have contributed to this
specification:
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, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,
Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,
Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
Chesire, David Walker, and Geetha Srikantan.
[1] H. Schulzrinne, "RTP profile for audio and video conferences with
minimal control," RFC 3351, Internet Engineering Task Force, July
2003.
[2] R. Fielding, J. Gettys, J. Mogul, H. Nielsen, and T. Berners-Lee,
"Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
Engineering Task Force, Jan. 1997.
[3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst,
"Internationalization of the hypertext markup language," RFC 2070,
Internet Engineering Task Force, Jan. 1997.
H. Schulzrinne et. al. [Page 137]
Internet Draft RTSP October 27, 2003
[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
pictures and associated audio informaiton -- part 6: extension for
digital storage media and control," Draft International Standard ISO
13818-6, International Organization for Standardization ISO/IEC
JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.
[6] 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
protocol (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
rating systems (and their machine readable descriptions),"
Recommendation REC-PICS-services-961031, W3C (World Wide Web
Consortium), Boston, Massachusetts, Oct. 1996.
[13] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
distribution label syntax and communication protocols,"
Recommendation REC-PICS-labels-961031, W3C (World Wide Web
Consortium), Boston, Massachusetts, Oct. 1996.
[14] D. Crocker and P. Overell, "Augmented BNF for syntax
specifications: ABNF," RFC 2234, Internet Engineering Task Force,
Nov. 1997.
[15] B. Braden, "Requirements for internet hosts - application and
support," RFC STD 3, 1123, Internet Engineering Task Force, Oct.
1989.
H. Schulzrinne et. al. [Page 138]
Internet Draft RTSP October 27, 2003
[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 3550, Internet
Engineering Task Force, July 2003.
[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
Standarization Sector of ITU, Geneva, Switzerland, May 1996.
[29] T. Narten, H. Alvestrand, "Guidelines for Writing an IANA
Considerations Section in RFCs," RFC2434, Internet Engineering Task
Force, October 1998.
H. Schulzrinne et. al. [Page 139]
Internet Draft RTSP October 27, 2003
[30] R. Hinden, B. Carpenter, L. Masinter, "Format for Literal IPv6
Addresses in URL's," RFC 2732, Internet Engineering Task Force,
December 1999.
[31] J. Rosenberg, J. Weinberger, C. Huitema, R. Mahy, "STUN - Simple
Traversal of UDP Through Network Address Translators," Internet
Engineering Task Force, Work in Progress, October 2002.
[32] P. Srisuresh, K. Egevang, "Traditional IP Network Address
Translator (Traditional NAT)," RFC 3022, Internet Engineering Task
Force, January 2001.
[33] M. Westerlund, "How to make Real-Time Streaming Protocol (RTSP)
traverse Network Address Translators (NAT) and interact with
Firewalls.", Internet Engineering Task Force Draft, draft-ietf-
mmusic-rtsp-nat-00.txt, Work in Progress, Feb 2003.
[34] A. Narasimhan, A. Narasimhan, "MUTE and UNMUTE extension to
RTSP", Internet Engineering Task Force Draft, draft-sergent-rtsp-
mute-00.txt, Work in Progress, Feb 2002.
[35] Third Generation Partnership Project (3GPP), "Transparent end-
to-end Packet-switched Streaming Service (PSS); Protocols and codecs"
3GPP Technical Specification 26.234, Release 5.
[36] D. Yon, "Connection-Oriented Media Transport in SDP", Internet
Engineering Task Force Draft, draft-ietf-mmusic-sdp-comedia-04.txt,
July 2002.
[37] John Lazzaro, "Framing RTP and RTCP Packets over Connection-
Oriented Transport", Internet Engineering Task Force Draft , draft-
lazzaro-avt-rtp-framing-contrans-00.txt, January 2003.
[38] S. Olson, G. Camarill, A. B. Roach, "Support for IPv6 in Session
Description Protocol (SDP)," RFC 3266, Internet Engineering Task
Force, June 2002.
[39] R. Hinden, S. Deering, "Internet Protocol Version 6 (IPv6)
Addressing Architecture," RFC 3513, Internet Engineering Task Force,
April 2003.
[40] G. Camarillo, G. Eriksson, J. Holler, H. Schulzrinne, "Grouping
of Media Lines in the Session Description Protocol (SDP)," RFC 3388,
Internet Engineering Task Force, December 2002.
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IPR Notice
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proprietary rights by implementors or users of this specification can
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The IETF invites any interested party to bring to its attention any
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this standard. Please address the information to the IETF Executive
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Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implmentation may be prepared, copied, published and
distributed, in whole or in part, without restriction of any kind,
provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
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copyrights defined in the Internet Standards process must be
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The limited permissions granted above are perpetual and will not be
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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
H. Schulzrinne et. al. [Page 141]
Internet Draft RTSP October 27, 2003
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
H. Schulzrinne et. al. [Page 142]
Table of Contents
1 Introduction ........................................ 2
1.1 The Update of the RTSP Specification ................ 2
1.2 Purpose ............................................. 3
1.3 Requirements ........................................ 5
1.4 Terminology ......................................... 5
1.5 Protocol Properties ................................. 8
1.6 Extending RTSP ...................................... 9
1.7 Overall Operation ................................... 10
1.8 RTSP States ......................................... 11
1.9 Relationship with Other Protocols ................... 12
2 Notational Conventions .............................. 13
3 Protocol Parameters ................................. 13
3.1 RTSP Version ........................................ 13
3.2 RTSP URL ............................................ 13
3.3 Session Identifiers ................................. 15
3.4 SMPTE Relative Timestamps ........................... 15
3.5 Normal Play Time .................................... 16
3.6 Absolute Time ....................................... 17
3.7 Feature-tags ........................................ 17
4 RTSP Message ........................................ 18
4.1 Message Types ....................................... 19
4.2 Message Headers ..................................... 19
4.3 Message Body ........................................ 19
4.4 Message Length ...................................... 19
5 General Header Fields ............................... 19
6 Request ............................................. 20
6.1 Request Line ........................................ 20
6.2 Request Header Fields ............................... 20
7 Response ............................................ 22
7.1 Status-Line ......................................... 22
7.1.1 Status Code and Reason Phrase ....................... 22
7.1.2 Response Header Fields .............................. 24
8 Entity .............................................. 25
8.1 Entity Header Fields ................................ 25
8.2 Entity Body ......................................... 27
9 Connections ......................................... 27
9.1 Pipelining .......................................... 27
9.2 Reliability and Acknowledgements .................... 28 |
9.3 Unreliable Transport ................................ 28 |
9.4 The usage of connections ............................ 29
9.5 Use of IPv6 ......................................... 31
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10 Capability Handling ................................. 31
11 Method Definitions .................................. 32
11.1 OPTIONS ............................................. 33
11.2 DESCRIBE ............................................ 34
11.3 SETUP ............................................... 36
11.4 PLAY ................................................ 39
11.5 PAUSE ............................................... 42
11.6 TEARDOWN ............................................ 46
11.7 GET_PARAMETER ....................................... 47
11.8 SET_PARAMETER ....................................... 47
11.9 REDIRECT ............................................ 48
11.10 PING ................................................ 50
11.11 Embedded (Interleaved) Binary Data .................. 51
12 Status Code Definitions ............................. 53
12.1 Success 1xx ......................................... 53
12.1.1 100 Continue ........................................ 53
12.2 Success 2xx ......................................... 53
12.2.1 250 Low on Storage Space ............................ 53
12.3 Redirection 3xx ..................................... 53
12.3.1 TBW ................................................. 54
12.3.2 301 Moved Permanently ............................... 54
12.3.3 302 Found ........................................... 54
12.3.4 303 See Other ....................................... 54
12.3.5 304 Not Modified .................................... 54
12.3.6 305 Use Proxy ....................................... 55
12.4 Client Error 4xx .................................... 55
12.4.1 400 Bad Request ..................................... 55
12.4.2 405 Method Not Allowed .............................. 55
12.4.3 451 Parameter Not Understood ........................ 55
12.4.4 452 reserved ........................................ 55
12.4.5 453 Not Enough Bandwidth ............................ 55
12.4.6 454 Session Not Found ............................... 55
12.4.7 455 Method Not Valid in This State .................. 55
12.4.8 456 Header Field Not Valid for Resource ............. 56
12.4.9 457 Invalid Range ................................... 56
12.4.10 458 Parameter Is Read-Only .......................... 56
12.4.11 459 Aggregate Operation Not Allowed ................. 56
12.4.12 460 Only Aggregate Operation Allowed ................ 56
12.4.13 461 Unsupported Transport ........................... 56
12.4.14 462 Destination Unreachable ......................... 56
12.5 Server Error 5xx .................................... 56
12.5.1 551 Option not supported ............................ 57
13 Header Field Definitions ............................ 57
13.1 Accept .............................................. 59
13.2 Accept-Encoding ..................................... 59
13.3 Accept-Language ..................................... 59
13.4 Accept-Ranges ....................................... 59
13.5 Allow ............................................... 60
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13.6 Authorization ....................................... 60
13.7 Bandwidth ........................................... 60
13.8 Blocksize ........................................... 62
13.9 Cache-Control ....................................... 64
13.10 Connection .......................................... 67
13.11 Content-Base ........................................ 67
13.12 Content-Encoding .................................... 67
13.13 Content-Language .................................... 67
13.14 Content-Length ...................................... 67
13.15 Content-Location .................................... 68
13.16 Content-Type ........................................ 68
13.17 CSeq ................................................ 68
13.18 Date ................................................ 68
13.19 Expires ............................................. 68
13.20 From ................................................ 69
13.21 Host ................................................ 69
13.22 If-Match ............................................ 70
13.23 If-Modified-Since ................................... 70
13.24 Last-Modified ....................................... 70
13.25 Location ............................................ 70
13.26 Proxy-Authenticate .................................. 70
13.27 Proxy-Require ....................................... 71
13.28 Public .............................................. 71
13.29 Range ............................................... 72
13.30 Referer ............................................. 73
13.31 Retry-After ......................................... 74
13.32 Require ............................................. 74
13.33 RTP-Info ............................................ 75
13.34 Scale ............................................... 77
13.35 Speed ............................................... 78
13.36 Server .............................................. 78
13.37 Session ............................................. 78
13.38 Supported ........................................... 81
13.39 Timestamp ........................................... 81
13.40 Transport ........................................... 82
13.41 Unsupported ......................................... 88
13.42 User-Agent .......................................... 88
13.43 Vary ................................................ 88
13.44 Via ................................................. 88
13.45 WWW-Authenticate .................................... 88
14 Caching ............................................. 88
15 Examples ............................................ 89
15.1 Media on Demand (Unicast) ........................... 89
15.2 Streaming of a Container file ....................... 92
15.3 Single Stream Container Files ....................... 94
15.4 Live Media Presentation Using Multicast ............. 96
16 Syntax .............................................. 97
16.1 Base Syntax ......................................... 97
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16.2 RTSP Protocol Definition ............................ 98
16.2.1 Message Syntax ...................................... 98
16.2.2 Header Syntax ....................................... 102
17 Security Considerations ............................. 103
18 IANA Considerations ................................. 105
18.1 Feature-tags ........................................ 106
18.1.1 Description ......................................... 106
18.1.2 Registering New Feature-tags with IANA .............. 106
18.1.3 Registered entries .................................. 106
18.2 RTSP Methods ........................................ 107
18.2.1 Description ......................................... 107
18.2.2 Registering New Methods with IANA ................... 107
18.2.3 Registered Entries .................................. 107
18.3 RTSP Status Codes ................................... 107
18.3.1 Description ......................................... 107
18.3.2 Registering New Status Codes with IANA .............. 108
18.3.3 Registered Entries .................................. 108
18.4 RTSP Headers ........................................ 108
18.4.1 Description ......................................... 108
18.4.2 Registering New Headers with IANA ................... 108
18.4.3 Registered entries .................................. 109
18.5 Transport Header registries ......................... 109
18.5.1 Transport Protocols ................................. 109
18.5.2 Profile ............................................. 110
18.5.3 Lower Transport ..................................... 110
18.5.4 Transport modes ..................................... 110
18.6 Cache Directive Extensions .......................... 111
18.7 SDP attributes ...................................... 111
A RTSP Protocol State Machine ......................... 112
A.1 States .............................................. 112 |
A.2 State variables ..................................... 112 |
A.3 Abbreviations ....................................... 113 |
A.4 State Tables ........................................ 113 |
B Media Transport Alternatives ........................ 116
B.1 RTP ................................................. 117
B.1.1 AVP ................................................. 118
B.1.2 AVP/UDP ............................................. 119
B.1.3 AVP/TCP ............................................. 120
B.2 Future Additions .................................... 121
C Use of SDP for RTSP Session Descriptions ............ 121
C.1 Definitions ......................................... 121
C.1.1 Control URL ......................................... 122
C.1.2 Media Streams ....................................... 123 |
C.1.3 Payload Type(s) ..................................... 123
C.1.4 Format-Specific Parameters .......................... 124 |
C.1.5 Range of Presentation ............................... 124 |
C.1.6 Time of Availability ................................ 125
C.1.7 Connection Information .............................. 125
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C.1.8 Entity Tag .......................................... 125
C.2 Aggregate Control Not Available ..................... 125
C.3 Aggregate Control Available ......................... 126
C.4 RTSP external SDP delivery .......................... 127 |
D Minimal RTSP implementation ......................... 128
D.1 Client .............................................. 128
D.1.1 Basic Playback ...................................... 129
D.1.2 Authentication-enabled .............................. 129
D.2 Server .............................................. 129
D.2.1 Basic Playback ...................................... 130
D.2.2 Authentication-enabled .............................. 131
E Open Issues ......................................... 131
F Changes ............................................. 131
G Author Addresses .................................... 136
H Contributors ........................................ 137
I Acknowledgements .................................... 137
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