Internet Engineering Task Force MMUSIC WG
Internet Draft H. Schulzrinne, A. Rao, R. Lanphier
ietf-mmusic-rtsp-02.txt Columbia U./Netscape/Progressive Networks
March 27, 1997
Expires: September 26, 1997
Real Time Streaming Protocol (RTSP)
STATUS OF THIS MEMO
This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as ``work in progress''.
To learn the current status of any Internet-Draft, please check the
``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow
Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe),
munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or
ftp.isi.edu (US West Coast).
Distribution of this document is unlimited.
ABSTRACT
The Real Time Streaming Protocol, or RTSP, is an
application-level protocol for control over the delivery
of data with real-time properties. RTSP provides an
extensible framework to enable controlled, on-demand
delivery of real-time data, such as audio and video.
Sources of data can include both live data feeds and
stored clips. This protocol is intended to control
multiple data delivery sessions, provide a means for
choosing delivery channels such as UDP, multicast UDP and
TCP, and delivery mechanisms based upon RTP (RFC 1889).
1 Introduction
1.1 Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls
H. Schulzrinne, A. Rao, R. Lanphier [Page 1]
Internet Draft RTSP March 27, 1997
either a single or several time-synchronized streams of continuous
media such as audio and video. It does not typically deliver the
continuous streams itself, although interleaving of the continuous
media stream with the control stream is possible (see Section 9.11).
In other words, RTSP acts as a "network remote control" for
multimedia servers.
The set of streams to be controlled is defined by a presentation
description. This memorandum does not define a format for a
presentation description.
There is no notion of an RTSP connection; instead, a server maintains
a session labeled by an identifier. An RTSP session is in no way tied
to a transport-level connection such as a TCP connection. During an
RTSP session, an RTSP client may open and close many reliable
transport connections to the server to issue RTSP requests.
Alternatively, it may use a connectionless transport protocol such as
UDP.
The streams controlled by RTSP may use RTP [1], but the operation of
RTSP does not depend on the transport mechanism used to carry
continuous media.
The protocol is intentionally similar in syntax and operation to
HTTP/1.1, so that extension mechanisms to HTTP can in most cases also
be added to RTSP. However, RTSP differs in a number of important
aspects from HTTP:
o RTSP introduces a number of new methods and has a different
protocol identifier.
o An RTSP server needs to maintain state by default in almost
all cases, as opposed to the stateless nature of HTTP. (RTSP
servers and clients MAY use the HTTP state maintenance
mechanism [2].)
o Both an RTSP server and client can issue requests.
o Data is carried out-of-band, by a different protocol. (There
is an exception to this.)
o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
8859-1, consistent with current HTML internationalization
efforts [3].
o The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1
carries only the absolute path in the request
H. Schulzrinne, A. Rao, R. Lanphier [Page 2]
Internet Draft RTSP March 27, 1997
This makes virtual hosting easier. However, this is
incompatible with HTTP/1.1, which may be a bad idea.
The protocol supports the following operations:
Retrieval of media from media server: The client can request a
presentation description via HTTP or some other method. If the
presentation is being multicast, the presentation description
contains the multicast addresses and ports to be used for the
continuous media. If the presentation is to be sent only to the
client via unicast, the client provides the destination for
security reasons.
Invitation of a media server to a conference: A media server can be
"invited" to join an existing conference, either to play back
media into the presentation or to record all or a subset of the
media in a presentation. This mode is useful for distributed
teaching applications. Several parties in the conference may
take turns "pushing the remote control buttons".
Addition of media to an existing presentation: Particularly for live
presentations, it is useful if the server can tell the client
about additional media becoming available.
RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1.
1.2 Requirements
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC xxxx [4].
1.3 Terminology
Some of the terminology has been adopted from HTTP/1.1 [5]. Terms
not listed here are defined as in HTTP/1.1.
Conference: a multiparty, multimedia presentation, where "multi"
implies greater than or equal to one.
Client: The client requests continuous media data from the media
server.
Connection: A transport layer virtual circuit established between two
programs for the purpose of communication.
Continuous media: Data where there is a timing relationship between
H. Schulzrinne, A. Rao, R. Lanphier [Page 3]
Internet Draft RTSP March 27, 1997
source and sink, that is, the sink must reproduce the timing
relationshop that existed at the source. The most common
examples of continuous media are audio and motion video.
Continuous media can be realtime (interactive) , where there is
a "tight" timing relationship between source and sink, or
streaming (playback) , where the relationship is less strict.
Participant: Participants are members of conferences. A participant
may be a machine, e.g., a media record or playback server.
Media server: The network entity providing playback or recording
services for one or more media streams. Different media streams
within a presentation may originate from different media
servers. A media server may reside on the same or a different
host as the web server the presentation is invoked from.
Media parameter: Parameter specific to a media type that may be
changed while the stream is being played or prior to it.
(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.
Message: The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in
Section 14 and transmitted via a connection or a connectionless
protocol.
Presentation: A set of one or more streams which the server allows
the client to manipulate together. A presentation has a single
time axis for all streams belonging to it. Presentations are
defined by presentation descriptions (see below). A presentation
description contains RTSP URIs that define which streams can be
controlled individually and an RTSP URI to control the whole
presentation. A movie or live concert consisting of one or more
audio and video streams is be an example of a presentation.
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 [6] use the term "session" for a live presentation. The
presentation description may take several different formats,
including but not limited to the session description format SDP.
Response: An RTSP response. If an HTTP response is meant, that is
H. Schulzrinne, A. Rao, R. Lanphier [Page 4]
Internet Draft RTSP March 27, 1997
indicated explicitly.
Request: An RTSP request. If an HTTP request is meant, that is
indicated explicitly.
RTSP session: A complete RTSP "transaction", e.g., the viewing of a
movie. A session typically consist of a client setting up a
transport mechanism for the continuous media stream ( SETUP),
starting the stream with PLAY or RECORD and closing the stream
with TEARDOWN.
1.4 Protocol Properties
RTSP has the following properties:
Extendable: New methods and parameters can be easily added to RTSP.
Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers.
Secure: RTSP re-uses web security mechanisms, either at the transport
level (TLS [7]) or within the protocol itself. All HTTP
authentication mechanisms such as basic [5] and digest
authentication [8] are directly applicable.
Transport-independent: RTSP may use either an unreliable datagram
protocol (UDP) [9], a reliable datagram protocol (RDP, not
widely used [10]) or a reliable stream protocol such as TCP [11]
as it implements application-level reliability.
Multi-server capable: Each media stream within a presentation can
reside on a different server. The client automatically
establishes several concurrent control sessions with the
different media servers. Media synchronization is performed at
the transport level.
Control of recording devices: The protocol can control both recording
and playback devices, as well as devices that can alternate
between the two modes ("VCR").
Separation of stream control and conference initiation: Stream
control is divorced from inviting a media server to a
conference. The only requirement is that the conference
initiation protocol either provides or can be used to create a
unique conference identifier. In particular, SIP [12] or H.323
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
H. Schulzrinne, A. Rao, R. Lanphier [Page 5]
Internet Draft RTSP March 27, 1997
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 [13]) firewalls.
A firewall may need to understand the SETUP method to open a
"hole" for the UDP media stream.
HTTP-friendly: Where sensible, RTSP re-uses HTTP concepts, so that
the existing infrastructure can be re-used. This infrastructure
includes JEPI (the Joint Electronic Payment Initiative) for
electronic payments and PICS (Platform for Internet Content
Selection) for associating labels with content. However, RTSP
does not just add methods to HTTP, since the controlling
continuous media requires server state in most cases.
Appropriate server control: If a client can start a stream, it must
be able to stop a stream. Servers should not start streaming to
clients in such a way that clients cannot stop the stream.
Transport negotiation: The client can negotiate the transport method
prior to actually needing to process a continuous media stream.
Capability negotiation: If basic features are disabled, there must be
some clean mechanism for the client to determine which methods
are not going to be implemented. This allows clients to present
the appropriate user interface. For example, if seeking is not
allowed, the user interface must be able to disallow moving a
sliding position indicator.
An earlier requirement in RTSP' was multi-client
capability. However, it was determined that a better
approach was to make sure that the protocol is easily
extensible to the multi-client scenario. Stream identifiers
can be used by several control streams, so that "passing
the remote" would be possible. The protocol would not
address how several clients negotiate access; this is left
to either a "social protocol" or some other floor control
mechanism.
1.5 Extending RTSP
Since not all media servers have the same functionality, media
H. Schulzrinne, A. Rao, R. Lanphier [Page 6]
Internet Draft RTSP March 27, 1997
servers by necessity will support different sets of requests. For
example:
o A server may only be capable of playback, not recording and
thus has no need to support the RECORD request.
o A server may not be capable of seeking (absolute positioning),
say, if it is to support live events only.
o Some servers may not support setting stream parameters and
thus not support GET_PARAMETER and SET_PARAMETER.
A server SHOULD implement all header fields described in Section 11.
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, where
the methods described in [H19.6] are not likely to be supported
across all servers.
RTSP can be extended in three ways, listed 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.)
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 can then attempt an earlier,
less functional version.
o A new version of the protocol can be defined, allowing almost
all aspects (except the position of the protocol version
number) to change.
1.6 Overall Operation
Each presentation and media stream may be identified by an RTSP URL.
The overall presentation and the properties of the media the
presentation is made up of are defined by a presentation description
file, the format of which is outside the scope of this specification.
The presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored
on the media server.
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
H. Schulzrinne, A. Rao, R. Lanphier [Page 7]
Internet Draft RTSP March 27, 1997
without loss of generality, it is assumed that the presentation
description contains exactly one such presentation. A presentation
may contain several media streams.
The presentation description file contains a description of the media
streams making up the presentation, including their encodings,
language, and other parameters that enable the client to choose the
most appropriate combination of media. In this presentation
description, each media stream that is individually controllable by
RTSP is identified by an RTSP URL, which points to the media server
handling that particular media stream and names the stream stored on
that server. Several media streams can be located on different
servers; for example, audio and video streams can be split across
servers for load sharing. The description also enumerates which
transport methods the server is capable of.
Besides the media parameters, the network destination address and
port need to be determined. Several modes of operation can be
distinguished:
Unicast: The media is transmitted to the source of the RTSP request,
with the port number chosen by the client. Alternatively, the
media is transmitted on the same reliable stream as RTSP.
Multicast, server chooses address: The media server picks the
multicast address and port. This is the typical case for a live
or near-media-on-demand transmission.
Multicast, client chooses address: If the server is to participate in
an existing multicast conference, the multicast address, port
and encryption key are given by the conference description,
established by means outside the scope of this specification.
1.7 RTSP States
RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel. For example, RTSP control may
occur on a TCP connection while the data flows via UDP. Thus, data
delivery continues even if no RTSP requests are received by the media
server. Also, during its lifetime, a single media stream may be
controlled by RTSP requests issued sequentially on different TCP
connections. Therefore, the server needs to maintain "session state"
to be able to correlate RTSP requests with a stream. The state
transitions are described in Section A.
Many methods in RTSP do not contribute to state. However, the
following play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and
H. Schulzrinne, A. Rao, R. Lanphier [Page 8]
Internet Draft RTSP March 27, 1997
TEARDOWN.
SETUP: Causes the server to allocate resources for a stream and start
an RTSP session.
PLAY and RECORD: Starts data transmission on a stream allocated via
SETUP.
PAUSE: Temporarily halts a stream, without freeing server resources.
TEARDOWN: Frees resources associated with the stream. The RTSP
session ceases to exist on the server.
1.8 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may
interact with HTTP in that the initial contact with streaming content
is often to be made through a web page. The current protocol
specification aims to allow different hand-off points between a web
server and the media server implementing RTSP. For example, the
presentation description can be retrieved using HTTP or RTSP. Having
the presentation description be returned by the web server makes it
possible to have the web server take care of authentication and
billing, by handing out a presentation description whose media
identifier includes an encrypted version of the requestor's IP
address and a timestamp, with a shared secret between web and media
server.
However, RTSP differs fundamentally from HTTP in that data delivery
takes place out-of-band, in a different protocol. HTTP is an
asymmetric protocol, where the client issues requests and the server
responds. In RTSP, both the media client and media server can issue
requests. RTSP requests are also not stateless, in that they may set
parameters and continue to control a media stream long after the
request has been acknowledged.
Re-using HTTP functionality has advantages in at least two
areas, namely security and proxies. The requirements are
very similar, so having the ability to adopt HTTP work on
caches, proxies and authentication is valuable.
While most real-time media will use RTP as a transport protocol, RTSP
is not tied to RTP.
RTSP assumes the existence of a presentation description format that
can express both static and temporal properties of a presentation
containing several media streams.
H. Schulzrinne, A. Rao, R. Lanphier [Page 9]
Internet Draft RTSP March 27, 1997
2 Notational Conventions
Since many of the definitions and syntax are identical to HTTP/1.1,
this specification only points to the section where they are defined
rather than copying it. For brevity, [HX.Y] is to be taken to refer
to Section X.Y of the current HTTP/1.1 specification (RFC 2068).
All the mechanisms specified in this document are described in both
prose and an augmented Backus-Naur form (BNF) similar to that used in
RFC 2068 [H2.1]. It is described in detail in [14].
In this draft, we use indented and smaller-type paragraphs to provide
background and motivation. Some of these paragraphs are marked with
HS, AR and RL, designating opinions and comments by the individual
authors which may not be shared by the co-authors and require
resolution.
3 Protocol Parameters
3.1 RTSP Version
applies, with HTTP replaced by RTSP.
3.2 RTSP URL
The "rtsp" and "rtspu" schemes are used to refer to network resources
via the RTSP protocol. This section defines the scheme-specific
syntax and semantics for RTSP URLs.
rtsp_URL = ( "rtsp:" | "rtspu:" ) "//" host [ ":" port ] [abs_path]
host = <A legal Internet host domain name of IP address
(in dotted decimal form), as defined by Section 2.1
of RFC 1123>
port = *DIGIT
abs_path is defined in [H3.2.1].
Note that fragment and query identifiers do not have a
well-defined meaning at this time, with the interpretation
left to the RTSP server.
The scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtspu
identifies an unreliable protocol (within the Internet, UDP).
H. Schulzrinne, A. Rao, R. Lanphier [Page 10]
Internet Draft RTSP March 27, 1997
If the port is empty or not given, port 554 is assumed. The semantics
are that the identified resource can be controlled be 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 [15]).
A presentation or a stream is identified by an textual media
identifier, using the character set and escape conventions [H3.2] of
URLs [16]. Requests described in Section 9 can refer to either the
whole presentation or an individual stream within the presentation.
Note that some methods can only be applied to streams, not
presentations and vice versa. A specific instance of a presentation
or stream, e.g., one of several concurrent transmissions of the same
content, an RTSP session , is indicated by the Session header field
(Section 11.26) where needed.
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
This does not imply a standard way to reference streams in
URLs. The presentation description defines the hierarchical
relationships in the presentation and the URLs for the
individual streams. A presentation description may name a
stream 'a.mov' and the whole presentation 'b.mov'.
The path components of the RTSP URL are opaque to the client and do
not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be
used with non-RTSP media control protocols, simply by
replacing the scheme in the URL.
3.3 Conference Identifiers
Conference identifiers are opaque to RTSP and are encoded using
standard URI encoding methods (i.e., LWS is escaped with %). They can
contain any octet value. The conference identifier MUST be globally
H. Schulzrinne, A. Rao, R. Lanphier [Page 11]
Internet Draft RTSP March 27, 1997
unique. For H.323, the conferenceID value is to be used.
conference-id = 1*OCTET ; LWS must be URL-escaped
Conference identifiers are used to allow to allow RTSP
sessions to obtain parameters from multimedia conferences
the media server is participating in. These conferences are
created by protocols outside the scope of this
specification, e.g., H.323 [17] or SIP [12]. Instead of the
RTSP client explicitly providing transport information, for
example, it asks the media server to use the values in the
conference description instead. If the conference
participant inviting the media server would only supply a
conference identifier which is unique for that inviting
party, the media server could add an internal identifier
for that party, e.g., its Internet address. However, this
would prevent that the conference participant and the
initiator of the RTSP commands are two different entities.
3.4 SMPTE Relative Timestamps
A SMPTE relative time-stamp 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
,
with the origin at the start of the clip. For NTSC, the frame rate is
29.97 frames per second. This is handled by dropping the first frame
index of every minute, except every tenth minute. If the frame value
is zero, it may be omitted.
smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ]
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ "." 1*2DIGIT ]
Examples:
smpte=10:12:33.40-
smpte=10:7:33-
smpte=10:7:0-10:7:33
H. Schulzrinne, A. Rao, R. Lanphier [Page 12]
Internet Draft RTSP March 27, 1997
3.5 Normal Play Time
Normal play time (NPT) indicates the stream absolute position
relative to the beginning of the presentation, measured in seconds
and microseconds. The beginning of a presentation corresponds to 0
seconds and 0 microseconds. Negative values are not defined. The
microsecond field is always less than 1,000,000. 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." [18]
npt-range = "npt" "=" npt-time "-" [ npt-time ]
npt-time = 1*DIGIT [ ":" *DIGIT ]
Examples:
npt=123:45-125
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-time "-" [ utc-time ]
utc-time = utc-date "T" utc-time "Z"
utc-date = 8DIGIT ; < YYYYMMDD >
utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC:
19961108T143720.25Z
Example
4 RTSP Message
H. Schulzrinne, A. Rao, R. Lanphier [Page 13]
Internet Draft RTSP March 27, 1997
RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2044). 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.
RTSP messages can be carried over any lower-layer transport protocol
that is 8-bit clean.
Requests contain methods, the object the method is operating upon and
parameters to further describe the method. Methods are idempotent,
unless otherwise noted. Methods are also designed to require little
or no state maintenance at the media server.
4.1 Message Types
See [H4.1]
4.2 Message Headers
See [H4.2]
4.3 Message Body
See [H4.3]
4.4 Message Length
When a message-body is included with a message, the length of that
body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message-body
(such as the 1xx, 204, and 304 responses) is always
H. Schulzrinne, A. Rao, R. Lanphier [Page 14]
Internet Draft RTSP March 27, 1997
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 11.12) is
present, its value in bytes represents the length of the
message-body. If this header field is not present, a value
of zero is assumed.
3. By the server closing the connection. (Closing the
connection cannot be used to indicate the end of a request
body, since that would leave no possibility for the server
to send back a response.)
Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
transfer coding and requires the presence of the Content-Length
header field.
Given the moderate length of presentation descriptions
returned, the server should always be able to determine its
length, even if it is generated dynamically, making the
chunked transfer encoding unnecessary. Even though
Content-Length must be present if there is any entity body,
the rules ensure reasonable behavior even if the length is
not given explicitly.
5 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 CRLF
*request-header
CRLF
[ message-body ]
Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no CRLF
Method = "DESCRIBE" ; Section
| "GET_PARAMETER" ; Section
| "OPTIONS" ; Section
| "PAUSE" ; Section
| "PLAY" ; Section
| "RECORD" ; Section
H. Schulzrinne, A. Rao, R. Lanphier [Page 15]
Internet Draft RTSP March 27, 1997
| "REDIRECT" ; Section
| "SETUP" ; Section
| "SET_PARAMETER" ; Section
| "TEARDOWN" ; Section
| extension-method
extension-method = token
Request-URI = "*" | absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
seq-no = 1*DIGIT
Note that in contrast to HTTP/1.1, RTSP requests always contain the
absolute URL (that is, including the scheme, host and port) rather
than just the absolute path.
The asterisk "*" in the Request-URI means that the request does not
apply to a particular resource, but to the server itself, and is only
allowed when the method used does not necessarily apply to a
resource. One example would be
OPTIONS * RTSP/1.0
6 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version
define some HTTP codes. The valid response codes and the methods they
can be used with are defined in the table 1.
After receiving and interpreting a request message, the recipient
responds with an RTSP response message.
Response = Status-Line ; Section
*( general-header ; Section
| response-header ; Section
| entity-header ) ; Section
CRLF
[ message-body ] ; Section
H. Schulzrinne, A. Rao, R. Lanphier [Page 16]
Internet Draft RTSP March 27, 1997
6.1 Status-Line
The first line of a Response message is the Status-Line , consisting
of the protocol version followed by a numeric status code, the
sequence number of the corresponding request 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.
Note that the addition of a
Status-Line = RTSP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF
6.1.1 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 section10. The Reason-Phrase is intended to give a short
textual description of the Status-Code. The Status-Code is intended
for use by automata and the Reason-Phrase is intended for the human
user. The client is not required to examine or display the Reason-
Phrase
The first digit of the Status-Code defines the class of response. The
last two digits do not have any categorization role. There are 5
values for the first digit:
o 1xx: Informational - Request received, continuing process
o 2xx: Success - The action was successfully received,
understood, and accepted
o 3xx: Redirection - Further action must be taken in order to
complete the request
o 4xx: Client Error - The request contains bad syntax or cannot
be fulfilled
o 5xx: Server Error - The server failed to fulfill an apparently
valid request
The individual values of the numeric status codes defined for
RTSP/1.0, and an example set of corresponding Reason-Phrase 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 status codes and adds RTSP-specific
status codes in the starting at 450 to avoid conflicts with newly
H. Schulzrinne, A. Rao, R. Lanphier [Page 17]
Internet Draft RTSP March 27, 1997
defined HTTP status codes.
Status-Code = "100" ; Continue
| "200" ; OK
| "201" ; Created
| "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "304" ; Not Modified
| "305" ; Use Proxy
| "400" ; Bad Request
| "401" ; Unauthorized
| "402" ; Payment Required
| "403" ; Forbidden
| "404" ; Not Found
| "405" ; Method Not Allowed
| "406" ; Not Acceptable
| "407" ; Proxy Authentication Required
| "408" ; Request Time-out
| "409" ; Conflict
| "410" ; Gone
| "411" ; Length Required
| "412" ; Precondition Failed
| "413" ; Request Entity Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "451" ; Parameter Not Understood
| "452" ; Conference Not Found
| "453" ; Not Enough Bandwidth
| "45x" ; Session Not Found
| "45x" ; Method Not Valid in This State
| "45x" ; Header Field Not Valid for Resource
| "45x" ; Invalid Range
| "45x" ; Parameter Is Read-Only
| "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Time-out
| "505" ; HTTP Version not supported
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
H. Schulzrinne, A. Rao, R. Lanphier [Page 18]
Internet Draft RTSP March 27, 1997
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.
6.1.2 Response Header Fields
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
the Status-Line server and about further access to the resource
identified by the Request-URI
response-header = Location ; Section
| Proxy-Authenticate ; Section
| Public ; Section
| Retry-After ; Section
| Server ; Section
| Vary ; Section
| WWW-Authenticate ; Section
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.
7 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
H. Schulzrinne, A. Rao, R. Lanphier [Page 19]
Internet Draft RTSP March 27, 1997
Code reason
_______________________________________________________________
100 Continue all
_______________________________________________________________
200 OK all
201 Created RECORD
_______________________________________________________________
300 Multiple Choices all
301 Moved Permanently all
302 Moved Temporarily all
303 See Other all
305 Use Proxy all
_______________________________________________________________
400 Bad Request all
401 Unauthorized all
402 Payment Required all
403 Forbidden all
404 Not Found all
405 Method Not Allowed all
406 Not Acceptable all
407 Proxy Authentication Required all
408 Request Timeout all
409 Conflict
410 Gone all
411 Length Required SETUP
412 Precondition Failed
413 Request Entity Too Large SETUP
414 Request-URI Too Long all
415 Unsupported Media Type SETUP
45x Only Valid for Stream SETUP
45x Invalid parameter SETUP
45x Not Enough Bandwidth SETUP
45x Illegal Conference Identifier SETUP
45x Illegal Session Identifier PLAY, RECORD, TEARDOWN
45x Parameter Is Read-Only SET_PARAMETER
45x Header Field Not Valid 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
Table 1: Status codes and their usage with RTSP methods
H. Schulzrinne, A. Rao, R. Lanphier [Page 20]
Internet Draft RTSP March 27, 1997
or the server, depending on who sends and who receives the entity.
7.1 Entity Header Fields
Entity-header fields define optional metainformation about the
entity-body or, if no body is present, about the resource identified
by the request.
entity-header = Allow ; Section 14.7
| Content-Encoding ; Section 14.12
| Content-Language ; Section 14.13
| Content-Length ; Section 14.14
| Content-Type ; Section 14.18
| Expires ; Section 14.21
| Last-Modified ; Section 14.29
| 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.
7.2 Entity Body
See [H7.2]
8 Connections
RTSP requests can be transmitted in several different ways:
o persistent transport connections used for several request-
response transactions;
o one connection per request/response transaction;
o connectionless mode.
The type of transport connection is defined by the RTSP URI (Section
3.2). For the scheme "rtsp", a persistent connection is assumed,
while the scheme "rtspu" calls for RTSP requests to be send without
setting up a connection.
Unlike HTTP, RTSP allows the media server to send requests to the
H. Schulzrinne, A. Rao, R. Lanphier [Page 21]
Internet Draft RTSP March 27, 1997
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.
8.1 Pipelining
A client that supports persistent connections or connectionless mode
MAY "pipeline" its requests (i.e., send multiple requests without
waiting for each response). A server MUST send its responses to those
requests in the same order that the requests were received.
8.2 Reliability and Acknowledgements
Requests are acknowledged by the receiver unless they are sent to a
multicast group. If there is no acknowledgement, the sender may
resend the same message after a timeout of one round-trip time (RTT).
The round-trip time is estimated as in TCP (RFC TBD), with an initial
round-trip value of 500 ms. An implementation MAY cache the last RTT
measurement as the initial value for future connections. If a
reliable transport protocol is used to carry RTSP, the timeout value
MAY be set to an arbitrarily large value.
This can greatly increase responsiveness for proxies
operating in local-area networks with small RTTs. The
mechanism is defined such that the client implementation
does not have be aware of whether a reliable or unreliable
transport protocol is being used. It is probably a bad idea
to have two reliability mechanisms on top of each other,
although the RTSP RTT estimate is likely to be larger than
the TCP estimate.
Each request carries a sequence number, which is incremented by one
for each request transmitted. If a request is repeated because of
lack of acknowledgement, the sequence number is incremented.
This avoids ambiguities when computing round-trip time
estimates. [TBD: An initial sequence number negotiation
needs to be added for UDP; otherwise, a new stream
connection may see a request be acknowledged by a delayed
response from an earlier "connection". This handshake can
be avoided with a sequence number containing a timestamp of
sufficiently high resolution.]
The reliability mechanism described here does not protect against
reordering. This may cause problems in some instances. For example, a
TEARDOWN followed by a PLAY has quite a different effect than the
reverse. Similarly, if a PLAY request arrives before all parameters
H. Schulzrinne, A. Rao, R. Lanphier [Page 22]
Internet Draft RTSP March 27, 1997
are set due to reordering, the media server would have to issue an
error indication. Since sequence numbers for retransmissions are
incremented (to allow easy RTT estimation), the receiver cannot just
ignore out-of-order packets. [TBD: This problem could be fixed by
including both a sequence number that stays the same for
retransmissions and a timestamp for RTT estimation.]
Systems implementing RTSP MUST support carrying RTSP over TCP and MAY
support UDP. The default port for the RTSP server is 554 for both UDP
and TCP.
A number of RTSP packets destined for the same control end point may
be packed into a single lower-layer PDU or encapsulated into a TCP
stream. RTSP data MAY be interleaved with RTP and RTCP packets.
Unlike HTTP, an RTSP method header MUST contain a Content-Length
whenever that method contains a payload. Otherwise, an RTSP packet is
terminated with an empty line immediately following the method
header.
9 Method Definitions
The method token indicates the method to be performed on the resource
identified by the Request-URI case-sensitive. New methods may be
defined in the future. Method names may not start with a $ character
(decimal 24) and must be a token
method direction object requirement
________________________________________________________
DESCRIBE C -> S, S -> C P,S recommended
GET_PARAMETER C -> S, S -> C P,S optional
OPTIONS C -> S P,S required
PAUSE C -> S P,S recommended
PLAY C -> S P,S required
RECORD C -> S P,S optional
REDIRECT S -> C P,S optional
SETUP C -> S S required
SET_PARAMETER C -> S, S -> C P,S optional
TEARDOWN C -> S P,S required
Table 2: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommend, 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
H. Schulzrinne, A. Rao, R. Lanphier [Page 23]
Internet Draft RTSP March 27, 1997
SHOULD not try this method again for this server.
9.1 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 non-standard request. It does not influence server state.
In addition, if the optional Require header is present, option tags
within the header indicate features needed by the requestor that are
not required at the version level of the protocol.
Example 1:
C->S: OPTIONS * RTSP/1.0 1
Require: implicit-play, record-feature
Transport-Require: switch-to-udp-control, gzipped-messages
Note that these are fictional features (though we may want to make
them real one day).
Example 2 (using RFC2069-style authentication only as an example):
S->C: OPTIONS * RTSP/1.0 1
Authenticate: Digest realm="testrealm@host.com",
nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
opaque="5ccc069c403ebaf9f0171e9517f40e41"
S->C: RTSP/1.0 200 1 OK
Date: 23 Jan 1997 15:35:06 GMT
Nack-Transport-Require: switch-to-udp-control
Note that these are fictional features (though we may want to make
them real one day).
Example 2 (using RFC2069-style authentication only as an example):
C->S: RTSP/1.0 401 1 Unauthorized
Authorization: Digest username="Mufasa",
realm="testrealm@host.com",
nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
uri="/dir/index.html",
H. Schulzrinne, A. Rao, R. Lanphier [Page 24]
Internet Draft RTSP March 27, 1997
response="e966c932a9242554e42c8ee200cec7f6",
opaque="5ccc069c403ebaf9f0171e9517f40e41"
9.2 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. Alternatively, the server may "push" a new description to
the client, for example, if a new stream has become available. If a
new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent
again, rather than just the additional components, so that components
can be deleted.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 312
Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 312 OK
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 376
v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
e=mjh@isi.edu (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
m=whiteboard 32416 UDP WB
a=orient:portrait
or
S->C: RTSP/1.0 200 312 OK
Date: 23 Jan 1997 15:35:06 GMT
H. Schulzrinne, A. Rao, R. Lanphier [Page 25]
Internet Draft RTSP March 27, 1997
Content-Type: application/rtsl
Content-Length: 2782
<2782 octets of data containing stream description>
Server to client example:
S->C: DESCRIBE /twister RTSP/1.0 902
Session: 1234
Content-Type: application/rtsl
new RTSL presentation description
9.3 SETUP
The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. A client can issue a SETUP request for
a stream that is already playing to change transport parameters. For
the benefit of any intervening firewalls, a client must indicate the
transport parameters even if it has no influence over these
parameters, for example, where the server advertises a fixed
multicast address.
This avoids having firewall to parse numerous different
presentation description formats, for information which is
irrelevant.
If the optional Require header is present, option tags within the
header indicate features needed by the requestor that are not
required at the version level of the protocol. The Transport-Require
header is used to indicate proxy-sensitive features that MUST be
stripped by the proxy to the server if not supported. Furthermore,
any Transport-Require header features that are not supported by the
proxy MUST be negatively acknowledged by the proxy to the client if
not supported.
HS: In my opinion, the Require header should be replaced by
PEP since PEP is standards-track, has more functionality
and somebody already did the work.
The Transport header specifies the transport parameters acceptable
to the client for data transmission; the response will contain the
transport parameters selected by the server.
H. Schulzrinne, A. Rao, R. Lanphier [Page 26]
Internet Draft RTSP March 27, 1997
C->S: SETUP foo/bar/baz.rm RTSP/1.0 302
Transport: rtp/udp;port=458
S->C: RTSP/1.0 200 302 OK
Date: 23 Jan 1997 15:35:06 GMT
Transport: cush/udp;port=458
9.4 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.
The PLAY request positions the normal play time to the beginning of
the range specified and delivers stream data until the end of the
range is reached. PLAY requests may be pipelined (queued); a server
MUST queue PLAY requests to be executed in order. That is, a PLAY
request arriving while a previous PLAY request is still active is
delayed until the first has been completed.
This allows precise editing. For example, regardless of
how closely spaced the two PLAY commands in the example
below arrive, the server will play first second 10 through
15 and then, immediately following, seconds 20 to 25 and
finally seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 835
Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 836
Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 837
Range: npt=30-
See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It starts playing a
stream from the beginning unless the stream has been paused. If a
stream has been paused via PAUSE, stream delivery resumes at the
pause point. If a stream is playing, such a PLAY request causes no
further action and can be used by the client to test server liveness.
H. Schulzrinne, A. Rao, R. Lanphier [Page 27]
Internet Draft RTSP March 27, 1997
The Range header may also contain a time parameter. This parameter
specifies a time in UTC at which the playback should start. If the
message is received after the specified time, playback is started
immediately. The time parameter may be used to aid in
synchronisation of streams obtained from different sources.
For a on-demand stream, the server replies back with the actual range
that will be played back. This may differ from the requested range if
alignment of the requested range to valid frame boundaries is
required for the media source. If no range is specified in the
request, the current position is returned in the reply. The unit of
the range in the reply is the same as that in the request.
After playing the desired range, the presentation is automatically
paused, as if a PAUSE request had been issued.
The following example plays the whole presentation starting at SMPTE
time code 0:10:20 until the end of the clip. The playback is to start
at 15:36 on 23 Jan 1997.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 833
Range: smpte=0:10:20-;time=19970123T153600Z
S->C: RTSP/1.0 200 833 OK
Date: 23 Jan 1997 15:35:06 GMT
Range: smpte=0:10:22-;time=19970123T153600Z
For playing back a recording of a live presentation, it may be
desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0 835
Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 833 OK
Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the smpte format
and MAY support the clock format.
9.5 PAUSE
H. Schulzrinne, A. Rao, R. Lanphier [Page 28]
Internet Draft RTSP March 27, 1997
The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only
playback and recording of that stream is halted. For example, for
audio, this is equivalent to muting. If the request URL names a
presentation or group of streams, delivery of all currently active
streams within the presentation or group is halted. After resuming
playback or recording, synchronization of the tracks MUST be
maintained. Any server resources are kept.
The PAUSE request may contain a Range header specifying when the
stream or presentation is to be halted. The header must contain
exactly one value rather than a time range. The normal play time for
the stream is set to that value. The pause request becomes effective
the first time the server is encountering the time point specified.
If this header is missing, stream delivery is interrupted immediately
on receipt of the message.
For example, if the server has play requests for ranges 10 to 15 and
20 to 29 pending and then receives a pause request for NPT 21, it
would start playing the second range and stop at NPT 21. If the pause
request is for NPT 12 and the server is playing at NPT 13 serving the
first play request, it stops immediately. If the pause request is for
NPT 16, it stops after completing the first play request and discards
the second play request.
As another example, if a server has received requests to play ranges
10 to 15 and then 13 to 20, that is, overlapping ranges, the PAUSE
request for NPT=14 would take effect while playing the first range,
with the second PLAY request effectively being ignored, assuming the
PAUSE request arrives before the server has started playing the
second, overlapping range. Regardless of when the PAUSE request
arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the
Range header, a PLAY would still resume at that point in time, as it
is assumed that the client has discarded data after that point. This
ensures continuous pause/play cycling without gaps.
Example:
C->S: PAUSE /fizzle/foo RTSP/1.0 834
S->C: RTSP/1.0 200 834 OK
Date: 23 Jan 1997 15:35:06 GMT
H. Schulzrinne, A. Rao, R. Lanphier [Page 29]
Internet Draft RTSP March 27, 1997
9.6 TEARDOWN
Stop the stream delivery for the given URI, freeing the resources
associated with it. If the URI is the root node for this
presentation, any RTSP session identifier associated with the session
is no longer valid. Unless all transport parameters are defined by
the session description, a SETUP request has to be issued before the
session can be played again.
Example:
C->S: TEARDOWN /fizzle/foo RTSP/1.0 892
S->C: RTSP/1.0 200 892 OK
9.7 GET_PARAMETER
The requests retrieves the value of a parameter of a presentation or
stream specified in the URI. Multiple parameters can be requested in
the message body using the content type text/rtsp-parameters Note
that parameters include server and client statistics. IANA registers
parameter names for statistics and other purposes. GET_PARAMETER with
no entity body may be used to test client or server liveness
("ping").
Example:
S->C: GET_PARAMETER /fizzle/foo RTSP/1.0 431
Content-Type: text/rtsp-parameters
Session: 1234
Content-Length: 15
packets_received
jitter
C->S: RTSP/1.0 200 431 OK
Content-Length: 46
Content-Type: text/rtsp-parameters
packets_received: 10
jitter: 0.3838
H. Schulzrinne, A. Rao, R. Lanphier [Page 30]
Internet Draft RTSP March 27, 1997
9.8 SET_PARAMETER
This method requests to set the value of a parameter for a
presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client
to determine why a particular request failed. A server MUST allow a
parameter to be set repeatedly to the same value, but it MAY disallow
changing parameter values.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
Restricting setting transport parameters to SETUP is for
the benefit of firewalls.
The parameters are split in a fine-grained fashion so that
there can be more meaningful error indications. However, it
may make sense to allow the setting of several parameters
if an atomic setting is desirable. Imagine device control
where the client does not want the camera to pan unless it
can also tilt to the right angle at the same time.
A SET_PARAMETER request without parameters can be used as a way to
detect client or server liveness.
Example:
C->S: SET_PARAMETER /fizzle/foo RTSP/1.0 421
Content-type: text/rtsp-parameters
fooparam: foostuff
barparam: barstuff
S->C: RTSP/1.0 450 421 Invalid Parameter
Content-Length: 6
barparam
9.9 REDIRECT
A redirect request informs the client that it must connect to another
server location. It contains the mandatory header Location, which
indicates that the client should issue a DESCRIBE for that URL. It
H. Schulzrinne, A. Rao, R. Lanphier [Page 31]
Internet Draft RTSP March 27, 1997
may contain the parameter Range, which indicates when the
redirection takes effect.
This example request redirects traffic for this URI to the new server
at the given play time:
S->C: REDIRECT /fizzle/foo RTSP/1.0 732
Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z-
9.10 RECORD
This method initiates recording a range of media data according to
the presentation description. The timestamp reflects start and end
time (UTC). If no time range is given, use the start or end time
provided in the presentation description. If the session has already
started, commence recording immediately. The Conference header is
mandatory.
The server decides whether to store the recorded data under the
request-URI or another URI. If the server does not use the request-
URI, the response SHOULD be 201 (Created) and contain an entity which
describes the status of the request and refers to the new resource,
and a Location header.
A media server supporting recording of live presentations MUST
support the clock range format; the smpte format does not make sense.
In this example, the media server was previously invited to the
conference indicated.
C->S: RECORD /meeting/audio.en RTSP/1.0 954
Session: 1234
Conference: 128.16.64.19/32492374
9.11 Embedded Binary Data
Binary packets such as RTP data are encapsulated by an ASCII dollar
sign (24 decimal), followed by a one-byte session identifier,
followed by the length of the encapsulated binary data as a binary,
two-byte integer in network byte order. The binary data follows
immediately afterwards, without a CRLF.
H. Schulzrinne, A. Rao, R. Lanphier [Page 32]
Internet Draft RTSP March 27, 1997
10 Status Code Definitions
Where applicable, HTTP status [H10] codes are re-used. 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 request.
10.1 Redirection 3xx
See [H10.3].
Within RTSP, redirection may be used for load balancing or
redirecting stream requests to a server topologically closer to the
client. Mechanisms to determine topological proximity are beyond the
scope of this specification.
10.2 Client Error 4xx
10.2.1 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
contained in the request.
10.2.2 452 Conference Not Found
The conference indicated by a Conference header field is unknown to
the media server.
10.2.3 453 Not Enough Bandwidth
The request was refused since there was insufficient bandwidth. This
may, for example, be the result of a resource reservation failure.
10.2.4 45x Session Not Found
The RTSP session identifier is invalid or has timed out.
10.2.5 45x Method Not Valid in This State
The client or server cannot process this request in its current
state.
10.2.6 45x 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.
10.2.7 45x Invalid Range
H. Schulzrinne, A. Rao, R. Lanphier [Page 33]
Internet Draft RTSP March 27, 1997
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
10.2.8 45x Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can only be read, but not
modified.
11 Header Field Definitions
HTTP/1.1 or other, non-standard header fields not listed here
currently have no well-defined meaning and SHOULD be ignored by the
recipient.
Tables 3 summarizes the header fields used by RTSP. Type "R"
designates request headers, type "r" response headers. Fields marked
with "req." in the column labeled "support" MUST be implemented by
the recipient for a particular method, while fields marked "opt." are
optional. Note that not all fields marked 'r' will be send in every
request of this type; merely, that client (for response headers) and
server (for request headers) MUST implement them. The last column
lists the method for which this header field is meaningful; the
designation "entity" refers to all methods that return a message
body. Within this specification, DESCRIBE and GET_PARAMETER fall
into this class.
If the field content does not apply to the particular resource, the
server MUST return status 45x (Header Field Not Valid for Resource).
11.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
The "level" parameter for presentation descriptions is
properly defined as part of the MIME type registration, not
here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl, application/sdp;level=2
H. Schulzrinne, A. Rao, R. Lanphier [Page 34]
Internet Draft RTSP March 27, 1997
Header type support methods
_________________________________________________________________
Accept R opt. entity
Accept-Encoding R opt. entity
Accept-Language R opt. all
Authorization R opt. all
Bandwidth R opt. SETUP
Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control Rr opt. SETUP
Conference R opt. SETUP
Connection Rr req. all
Content-Encoding R req. SET_PARAMETER
Content-Encoding r req. DESCRIBE
Content-Length R req. SET_PARAMETER
Content-Length r req. entity
Content-Type R req. SET_PARAMETER
Content-Type r req. entity
Date Rr opt. all
Expires r opt. DESCRIBE
If-Modified-Since R opt. DESCRIBE, SETUP
Last-Modified r opt. entity
Public r opt. all
Range R opt. PLAY, PAUSE, RECORD
Range r opt. PLAY, PAUSE, RECORD
Referer R opt. all
Require R req. all
Retry-After r opt. all
Scale Rr opt. PLAY, RECORD
Session Rr req. all but SETUP, OPTIONS
Server r opt. all
Speed Rr opt. PLAY
Transport Rr req. SETUP
Transport-Require R xeq. all
User-Agent R opt. all
Via Rr opt. all
WWW-Authenticate r opt. all
Table 3: Overview of RTSP header fields
11.2 Accept-Encoding
See [H14.3]
11.3 Accept-Language
See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media
H. Schulzrinne, A. Rao, R. Lanphier [Page 35]
Internet Draft RTSP March 27, 1997
content.
11.4 Allow
The Allow response header field lists the methods supported by the
resource identified by the request-URI. The purpose of this field is
to strictly inform the recipient of valid methods associated with the
resource. An Allow header field must be present in a 405 (Method not
allowed) response.
Example of use:
Allow: SETUP, PLAY, RECORD, SET_PARAMETER
11.5 Authorization
See [H14.8]
11.6 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.
Bandwidth = "Bandwidth" ":" 1*DIGIT
Example:
Bandwidth: 4000
11.7 Blocksize
This 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 overrides it
with the media specific size if necessary. The block size is a
strictly positive decimal number and measured in octets. The server
only returns an error (416) if the value is syntactically invalid.
H. Schulzrinne, A. Rao, R. Lanphier [Page 36]
Internet Draft RTSP March 27, 1997
11.8 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.
[HS: Should there be an exception for DESCRIBE?]
Cache-Control = "Cache-Control" ":" 1#cache-directive
cache-directive = cache-request-directive
| cache-response-directive
cache-request-directive =
"no-cache"
| "max-stale"
| "min-fresh"
| "only-if-cached"
| cache-extension
cache-response-directive =
"public"
| "private"
| "no-cache"
| "no-transform"
| "must-revalidate"
| "proxy-revalidate"
| "max-age" "=" delta-seconds
| cache-extension
cache-extension = token [ "=" ( token | quoted-string ) ]
no-cache: Indicates that the media stream MUST NOT be cached
anywhere. This allows an origin server to prevent caching even
H. Schulzrinne, A. Rao, R. Lanphier [Page 37]
Internet Draft RTSP March 27, 1997
by caches that have been configured to return stale responses to
client requests.
public: Indicates that the media stream is cachable 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 certain stream. A proxy might, for
example, convert between video formats to save cache space or to
reduce the amount of traffic on a slow link. Serious operational
problems may occur, however, when these transformations have
been applied to streams intended for certain kinds of
applications. For example, applications for medical imaging,
scientific data analysis and those using end-to-end
authentication, all depend on receiving a stream that is bit for
bit identical to the original entity-body. Therefore, if a
response includes the no-transform directive, an intermediate
cache or proxy MUST NOT change the encoding of the stream.
Unlike HTTP, RTSP does not provide for partial transformation at
this point, e.g., allowing translation into a different
language.
only-if-cached: In some cases, such as times of extremely poor
network connectivity, a client may want a cache to return only
those media streams that it currently has stored, and not to
receive these from the origin server. To do this, the client may
include the only-if-cached directive in a request. If it
receives this directive, a cache SHOULD either respond using a
cached media stream that is consistent with the other
constraints of the request, or respond with a 504 (Gateway
Timeout) status. However, if a group of caches is being operated
as a unified system with good internal connectivity, such a
request MAY be forwarded within that group of caches.
max-stale: Indicates that the client is willing to accept a media
stream that has exceeded its expiration time. If max-stale is
assigned a value, then the client is willing to accept a
response that has exceeded its expiration time by no more than
the specified number of seconds. If no value is assigned to
max-stale, then the client is willing to accept a stale response
of any age.
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
H. Schulzrinne, A. Rao, R. Lanphier [Page 38]
Internet Draft RTSP March 27, 1997
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. (I.e., 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.)
11.9 Conference
This request header field establishes a logical connection between a
conference, established using non-RTSP means, and an RTSP stream. The
conference-id must not be changed for the same RTSP session.
Conference = "Conference" ":" conference-id
Example:
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu
11.10 Connection
See [H14.10].
11.11 Content-Encoding
See [H14.12]
11.12 Content-Length
This 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. It is interpreted according to [H14.14].
11.13 Content-Type
See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and
parameter-value types.
H. Schulzrinne, A. Rao, R. Lanphier [Page 39]
Internet Draft RTSP March 27, 1997
11.14 Date
See [H14.19].
11.15 Expires
The Expires entity-header field gives the date/time after which the
media-stream should be considered stale. A stale cache entry may not
normally be returned by a cache (either a proxy cache or an user
agent cache) unless it is first validated with the origin server (or
with an intermediate cache that has a fresh copy of the entity). See
section 13.2 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 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 cachable, unless
indicated otherwise by a Cache-Control header field (Section 11.8.
H. Schulzrinne, A. Rao, R. Lanphier [Page 40]
Internet Draft RTSP March 27, 1997
11.16 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 setup ( 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
11.17 Last-modified
The Last-Modified entity-header field indicates the date and time at
which the origin server believes the variant was last modified. See
[H14.29]. If the request URI refers to an aggregate, the field
indicates the last modification time across all leave nodes of that
aggregate.
11.18 Location
See [H14.30].
11.19 Nack-Transport-Require
Negative acknowledgement of features not supported by the server. If
there is a proxy on the path between the client and the server, the
proxy MUST insert a message reply with an error message 506 (Feature
not supported).
HS: Same caveat as for Require applies.
11.20 Range
This request header field specifies a range of time. The range can be
specified in a number of units. This specification defines the smpte
(see Section 3.4) and clock (see Section 3.6) range units. Within
H. Schulzrinne, A. Rao, R. Lanphier [Page 41]
Internet Draft RTSP March 27, 1997
RTSP, byte ranges [H14.36.1] are not meaningful and MUST NOT be used.
The header may also contain a time parameter in UTC, specifying the
time at which the operation is to be made effective.
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
header. It allows to select a clip from the media object,
to play from a given point to the end and from the current
location to a given point.
11.21 Require
The Require header is used by clients to query the server about
features that it may or may not support. The server MUST respond to
this header by negatively acknowledging those features which are NOT
supported in the Unsupported header.
HS: Naming of features -- yet another name space. I believe
this header field to be redundant. PEP should be used
instead.
For example
C->S: SETUP /foo/bar/baz.rm RTSP/1.0 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/1.0 200 506 Option not supported
Unsupported: funky-feature
C->S: SETUP /foo/bar/baz.rm RTSP/1.0 303
S->C: RTSP/1.0 200 303 OK
H. Schulzrinne, A. Rao, R. Lanphier [Page 42]
Internet Draft RTSP March 27, 1997
This is to make sure that the client-server interaction will proceed
optimally when all options are understood by both sides, and only
slow down if options aren't understood (as in the case above). For a
well-matched client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation mechanisms. In
addition, it also removes state ambiguity when the client requires
features that the server doesn't understand.
11.22 Retry-After
See [H14.38].
11.23 Scale
A scale value of 1 indicates normal play or record at the normal
forward viewing rate. If not 1, the value corresponds to the rate
with respect to normal viewing rate. For example, a ratio of 2
indicates twice the normal viewing rate ("fast forward") and a ratio
of 0.5 indicates half the normal viewing rate. In other words, a
ratio of 2 has normal play time increase at twice the wallclock rate.
For every second of elapsed (wallclock) time, 2 seconds of content
will be delivered. A negative value indicates reverse direction.
Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected key frames. For audio, it may time-scale
the audio while preserving pitch or, less desirably, deliver
fragments of audio.
The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response
MUST contain the actual scale value chosen by the server.
If the request contains a Range parameter, the new scale value will
take effect at that time.
Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
H. Schulzrinne, A. Rao, R. Lanphier [Page 43]
Internet Draft RTSP March 27, 1997
11.24 Speed
This request header fields parameter 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. A negative value indicates that the stream
is to be played back in reverse direction.
HS: With 'Scale', the negative value is redundant and
should probably be removed since it only leads to possible
conflicts when Scale is positive and Speed negative.
If the request contains a Range parameter, the new speed value will
take effect at that time.
Speed = "Speed" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
Example:
Speed: 2.5
11.25 Server
See [H14.39]
11.26 Session
This request and response header field identifies an RTSP session,
started by the media server in a SETUP response and concluded by
TEARDOWN on the presentation URL. The session identifier is chosen by
the media server and has the same syntax as a conference identifier.
Once a client receives a Session identifier, it MUST return it for
any request related to that session.
HS: This may be redundant with the standards-track HTTP
state maintenance mechanism [2]. The equivalent way of
H. Schulzrinne, A. Rao, R. Lanphier [Page 44]
Internet Draft RTSP March 27, 1997
doing this would be for the server to send Set-Cookie:
Session="123"; Version=1; Path = "/twister" and for the
client to return later Cookie: Session = "123"; $Version=1;
$Path = "/twister" response to the TEARDOWN message, the
server would simply send Set-Cookie: Session="123";
Version=1; Max-Age=0 to get rid of the cookie on the client
side. Cookies also have a time-out, so that a server may
limit the lifetime of a session at will. Unlike a web
browser, a client would not store these states on disk. To
avoid privacy issues, we should prohibit the Host
parameter.
11.27 Transport
This request header indicates which transport protocol is to be used
and configures its parameters such as multicast, compression,
multicast time-to-live and destination port for a single stream. It
sets those values not already determined by a presentation
description. In some cases, the presentation description contains all
necessary information. In those cases, a Transport header field
(and the SETUP request containing it) are not needed.
in whatever protocol is being used by the control stream. Currently,
the next-layer protocols RTP is defined. Parameters may be added to
each protocol, separated by a semicolon. For RTP, the boolean
parameter compressed is defined, indicating compressed RTP according
to RFC XXXX. For multicast UDP, the integer parameter ttl defines
the time-to-live value to be used. The client may specify the
multicast address with the multicast parameter. A server SHOULD
authenticate the client before allowing the client to direct a media
stream to a multicast address not chosen by the server to avoid
becoming the unwitting perpetrator of a denial-of-service attack. For
UDP and TCP, the parameter port defines the port data is to be sent
to.
The SSRC parameter indicates the RTP SSRC value that should be
(request) or will be (response) used by the media server. This
parameter is only valid for unicast transmission. It identifies the
synchronization source to be associated with the media stream.
The Transport header 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 in the response to
indicate the values actually chosen.
A Transport request header field may contain a list of transport
H. Schulzrinne, A. Rao, R. Lanphier [Page 45]
Internet Draft RTSP March 27, 1997
options acceptable to the client. In that case, the server MUST
return a single option which was actually chosen. The Transport
header field makes sense only for an individual media stream, not a
presentation.
Transport = "Transport" ":"
1#transport-protocol/upper-layer *parameter
transport-protocol = "UDP" | "TCP"
upper-layer = "RTP"
parameters = ";" "multicast" [ "=" mca ]
| ";" "compressed"
| ";" "interleaved"
| ";" "ttl" "=" ttl
| ";" "port" "=" port
| ";" "ssrc" "=" ssrc
ttl = 1*3(DIGIT)
port = 1*5(DIGIT)
ssrc = 8*8(HEX)
mca = host
Example:
Transport: udp/rtp;compressed;ttl=127;port=3456
11.28 Transport-Require
The Transport-Require header is used to indicate proxy-sensitive
features that MUST be stripped by the proxy to the server if not
supported. Furthermore, any Transport-Require header features that
are not supported by the proxy MUST be negatively acknowledged by the
proxy to the client if not supported.
See Section 11.21 for more details on the mechanics of this message
and a usage example.
HS: Same caveat as for Require applies.
11.29 Unsupported
See Section 11.21 for a usage example.
HS: same caveat as for Require applies.
H. Schulzrinne, A. Rao, R. Lanphier [Page 46]
Internet Draft RTSP March 27, 1997
11.30 User-Agent
See [H14.42]
11.31 Via
See [H14.44].
11.32 WWW-Authenticate
See [H14.46].
12 Caching
In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cachable, with the
exception of the stream 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.
On receiving a SETUP or PLAY request, the proxy would ascertain as
to whether it has an up-to-date copy of the continuous media content.
If not, it would modify the SETUP transport parameters as
appropriate and forward the request to the origin server. Subsequent
control commands such as PLAY or PAUSE would pass the proxy
unmodified. The proxy would then pass the continuous media data to
the client, while possibly making a local copy for later re-use. The
exact behavior allowed to the cache is given by the cache-response
directives described in Section 11.8. 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 would appear like a regular media
server, to the media origin server like a client. Just like an HTTP
cache has to store the content type, content language, etc. for the
objects it caches, a media cache has to store the presentation
description. Typically, a cache would eliminate all transport-
references (that is, multicast information) from the presentation
description, since these are independent of the data delivery from
H. Schulzrinne, A. Rao, R. Lanphier [Page 47]
Internet Draft RTSP March 27, 1997
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.
13 Examples
The following examples reference stream description formats that are
not finalized, such as RTSL and SDP. Please do not use these examples
as a reference for those formats.
13.1 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
give an indication about the time line of the movie.
In our example, the client is only interested in the last part of the
movie. The server requires authentication for this movie. The audio
track can be dynamically switched between between two sets of
encodings. The URL with scheme rtpsu indicates the media servers
want to use UDP for exchanging RTSP messages.
C->W: DESCRIBE /twister HTTP/1.1
Host: www.example.com
Accept: application/rtsl; application/sdp
W->C: 200 OK
Content-Type: application/rtsl
<session>
<group language=en lipsync>
<switch>
<track type=audio
e="PCMU/8000/1"
src="rtsp://audio.example.com/twister/audio.en/lofi">
<track type=audio
e="DVI4/16000/2" pt="90 DVI4/8000/1"
src="rtsp://audio.example.com/twister/audio.en/hifi">
</switch>
<track type="video/jpeg"
src="rtspu://video.example.com/twister/video">
</group>
H. Schulzrinne, A. Rao, R. Lanphier [Page 48]
Internet Draft RTSP March 27, 1997
</session>
C->A: SETUP rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 1
Transport: rtp/udp;compression;port=3056
A->C: RTSP/1.0 200 1 OK
Session: 1234
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 1
Transport: rtp/udp;compression;port=3058
V->C: RTSP/1.0 200 1 OK
Session: 1235
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 2
Session: 1235
Range: smpte=0:10:00-
V->C: RTSP/1.0 200 2 OK
C->A: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 2
Session: 1234
Range: smpte=0:10:00-
A->C: 200 2 OK
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 3
Session: 1234
A->C: 200 3 OK
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 3
Session: 1235
V->C: 200 3 OK
Even though the audio and video track are on two different servers,
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.
13.2 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
H. Schulzrinne, A. Rao, R. Lanphier [Page 49]
Internet Draft RTSP March 27, 1997
description, while the media server M maintains the full description.
During the RTSP session, a new subtitling stream is added.
C->W: GET /concert HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: application/rtsl
<session>
<track id=17 src="rtsp://live.example.com/concert/audio">
</session>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0 1
M->C: RTSP/1.0 200 1 OK
Content-Type: application/rtsl
<track id=17 type=audio address=224.2.0.1 port=3456 ttl=16>
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 2
Transport: multicast=224.2.0.1; port=3456; ttl=16
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 3
Range: smpte 1:12:0
M->C: RTSP/1.0 405 3 No positioning possible
M->C: DESCRIBE concert RTSP/1.0
Content-Type: application/rtsl
<session>
<track id=17
media=audio/g.728 src="rtsp://live.example.com/concert/audio">
<track id=18
media=text/html src="rtsp://live.example.com/concert/lyrics">
</session>
C->M: PLAY rtsp://live.example.com/concert/lyrics RTSP/1.0
The attempt to position the stream fails since this is a live
presentation.
13.3 Playing media into an existing session
H. Schulzrinne, A. Rao, R. Lanphier [Page 50]
Internet Draft RTSP March 27, 1997
A conference participant C wants to have the media server M play back
a demo tape into an existing conference. When retrieving the
presentation description, C indicates to the media server that the
network addresses and encryption keys are already given by the
conference, so they should not be chosen by the server. The example
omits the simple ACK responses.
C->M: GET /demo HTTP/1.1
Host: www.example.com
Accept: application/rtsl, application/sdp
M->C: HTTP/1.1 200 1 OK
Content-type: application/rtsl
<session>
<track type=audio/g.723.1
src="rtsp://server.example.com/demo/548/sound">
</session>
C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0 2
Conference: 218kadjk
13.4 Recording
The conference participant C asks the media server M to record a
meeting. If the presentation description contains any alternatives,
the server records them all.
C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 89
Content-Type: application/sdp
v=0
s=Mbone Audio
i=Discussion of Mbone Engineering Issues
M->C: 415 89 Unsupported Media Type
Accept: application/rtsl
C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 90
Content-Type: application/rtsl
M->C: 200 90 OK
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 91
H. Schulzrinne, A. Rao, R. Lanphier [Page 51]
Internet Draft RTSP March 27, 1997
Range: clock 19961110T1925-19961110T2015
14 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 (HTTP/1.1).
14.1 Base Syntax
OCTET = <any 8-bit sequence of data>
CHAR = <any US-ASCII character (octets 0 - 127)>
UPALPHA = <any US-ASCII uppercase letter "A".."Z">
LOALPHA = <any US-ASCII lowercase letter "a".."z">
ALPHA = UPALPHA | LOALPHA
DIGIT = <any US-ASCII digit "0".."9">
CTL = <any US-ASCII control character
(octets 0 - 31) and DEL (127)>
CR = <US-ASCII CR, carriage return (13)>
LF = <US-ASCII LF, linefeed (10)>
SP = <US-ASCII SP, space (32)>
HT = <US-ASCII HT, horizontal-tab (9)>
<"> = <US-ASCII double-quote mark (34)>
CRLF = CR LF
LWS = [CRLF] 1*( SP | HT )
TEXT = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | "
| "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">>
quoted-pair = "
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>
15 Security Considerations
H. Schulzrinne, A. Rao, R. Lanphier [Page 52]
Internet Draft RTSP March 27, 1997
The protocol offers the opportunity for a remote-control denial-of-
service attack. The attacker, using a forged source IP address, can
ask for a stream to be played back to that forged IP address.
Since there is no relation between a transport layer connection and
an RTSP session, it is possible for a malicious client to issue
requests with random session identifiers which would affect
unsuspecting clients. This does not require spoofing of network
packet addresses. The server SHOULD use a large random session
identifier to make this attack more difficult.
Both problems can be be prevented by appropriate authentication.
In addition, the security considerations outlined in [H15] apply.
A RTSP Protocol State Machines
The RTSP client and server state machines describe the behavior of
the protocol from RTSP session initialization through RTSP session
termination.
[TBD: should we allow for the trivial case of a server that only
implements the PLAY message, with no control.]
State is defined on a per object basis. An object is uniquely
identified by the stream URL and the RTSP session identifier. (A
server may choose to generate dynamic presentation descriptions where
the URL is unique for a particular RTSP session and thus may not need
an explicit RTSP session identifier in the request header.) Any
request/reply using URLs denoting an RTSP session comprised of
multiple streams will have an effect on the individual states of all
the substreams. For example, if the stream /movie contains two
substreams /movie/audio and /movie/video, then the following command:
PLAY /movie RTSP/1.0 559
Session: 12345
will have an effect on the states of movie/audio and movie/video.
This example does not imply a standard way to represent
substreams in URLs or a relation to the filesystem. See
Section 3.2.
The requests OPTIONS, DESCRIBE, GET_PARAMETER, SET_PARAMETER do
H. Schulzrinne, A. Rao, R. Lanphier [Page 53]
Internet Draft RTSP March 27, 1997
not have any effect on client or server state and are therefore not
listed in the state tables.
Client and servers MUST disregard messages with a sequence number
less than the last one. If no message has been received, the first
received message's sequence number will be the starting point.
A.1 Client State Machine
The client can assume the following states:
Init: SETUP has been sent, waiting for reply.
Ready: SETUP reply received OR after playing, PAUSE reply received.
Playing: PLAY reply received
Recording: RECORD reply received
The client changes state on receipt of replies to requests. If no
explicit SETUP is required for the object (for example, it is
available via a multicast group), state begins at READY. In this
case, there are only two states, READY and PLAYING.
The "next state" column indicates the state assumed after receiving a
success response (2xx). If a request yields a status code greater or
equal to 300, the client state becomes Init, with the exception of
status codes 401 (Unauthorized) and 402 (Payment Required), where the
state remains unchanged and the request should be re-issued with the
appropriate authentication or payment information. Messages not
listed for each state MUST NOT be issued by the client in that state,
with the exception of messages not affecting state, as listed above.
Receiving a REDIRECT from the server is equivalent to receiving a 3xx
redirect status from the server.
HS: Depends on allowing PLAY without SETUP. After 4xx or
5xx error, do we go back to Init?
state message next state
_______________________________________________________
Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
RECORD Recording
TEARDOWN Init
Playing PAUSE Ready
TEARDOWN Init
H. Schulzrinne, A. Rao, R. Lanphier [Page 54]
Internet Draft RTSP March 27, 1997
PLAY Playing
RECORD Recording
SETUP Playing (changed transport)
Recording PAUSE Init
TEARDOWN Init
PLAY Playing
RECORD Recording
SETUP Recording (changed transport)
A.2 Server State Machine
The server can assume the following states:
Init: The initial state, no valid SETUP received.
Ready: Last SETUP received was successful, reply sent or after
playing, last PAUSE received was successful, reply sent.
Playing: Last PLAY received was successful, reply sent. Data is
being sent.
Recording: The server is recording media data.
The server changes state on receiving requests. If the server is in
state Playing or Recording and in unicast mode, it MAY revert to Init
and tear down the RTSP session if it has not received "wellness"
information, such as RTCP reports, from the client for a defined
interval, with a default of one minute. If the server is in state
Ready, it MAY revert to Init if it does not receive an RTSP request
for an interval of more than one minute.
The REDIRECT message, when sent, is effective immediately. If a
similar change of location occurs at a certain time in the future,
this is assumed to be indicated by the presentation description.
SETUP is valid in states Init and Ready only. An error message should
be returned in other cases. If no explicit SETUP is required for the
object, state starts at READY, there are only two states READY and
PLAYING.
state message next state
___________________________________
Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
SETUP Ready
TEARDOWN Ready
H. Schulzrinne, A. Rao, R. Lanphier [Page 55]
Internet Draft RTSP March 27, 1997
Playing PLAY Playing
PAUSE Ready
TEARDOWN Ready
RECORD Recording
SETUP Playing
Recording RECORD Recording
PAUSE Ready
TEARDOWN Ready
PLAY Playing
SETUP Recording
B Open Issues
o Define text/rtsp-parameter MIME type.
o HS believes that RTSP should only control individual media
objects rather than aggregates. This avoids disconnects between
presentation descriptions and streams and avoids having to deal
separately with single-host and multi-host case. Cost: several
PLAY/PAUSE/RECORD in one packet, one for each stream.
o Allow changing of transport for a stream that's playing? May
not be a great idea since the same can be accomplished by tear
down and re-setup.
o Allow fragment (#) identifiers for controlling substreams in
Quicktime, AVI and ASF files?
o How does the server get back to the client unless a persistent
connection is used? Probably cannot, in general.
o Cache and proxy behavior?
o Session: or Set-Cookie: ?
o When do relative RTSP URLs make sense?
o Nack-require, etc. are dubious. This is getting awfully close
to the HTTP extension mechanisms [19] in complexity, but is
different.
o Use HTTP absolute path + Host field or do the right thing and
carry full URL, including host in request?
C Changes
Since the February 1997 version, the following changes were made:
H. Schulzrinne, A. Rao, R. Lanphier [Page 56]
Internet Draft RTSP March 27, 1997
o Various editorial changes and clarifications.
o Removed references to SDF and replaced by RTSL.
o Added Scale general header.
o Clarify behavior of PLAY.
o Rename GET to DESCRIBE.
o Removed SESSION since it is just DESCRIBE in the other
direction.
o Rename CLOSE to TEARDOWN, in symmetry with SETUP.
o Terminology adjusted to "presentation" and "stream".
o Redundant syntax BNF in appendix removed since it just
duplicates HTTP spec.
o Beginnings of cache control.
Changes are marked by changebars in the margins of the PostScript
version.
D 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
Netscape Communications Corp.
USA
electronic mail: anup@netscape.com
Robert Lanphier
Progressive Networks
1111 Third Avenue Suite 2900
Seattle, WA 98101
USA
electronic mail: robla@prognet.com
E Acknowledgements
H. Schulzrinne, A. Rao, R. Lanphier [Page 57]
Internet Draft RTSP March 27, 1997
This draft is based on the functionality of the RTSP draft. 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 Eduardo F. Llach
Bruce Butterfield Rob McCool
Martin Dunsmuir Sujal Patel
Eric Fleischman
Mark Handley Igor Plotnikov
Peter Haight Pinaki Shah
Brad Hefta-Gaub Jeff Smith
John K. Ho Alexander Sokolsky
Ruth Lang Dale Stammen
Stephanie Leif John Francis Stracke
F Bibliography
[1] H. Schulzrinne, "RTP profile for audio and video conferences with
minimal control," RFC 1890, Internet Engineering Task Force, Jan.
1996.
[2] D. Kristol and L. Montulli, "HTTP state management mechanism,"
RFC 2109, Internet Engineering Task Force, Feb. 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.
[4] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," Internet Draft, Internet Engineering Task Force, Jan. 1997.
Work in progress.
[5] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-Lee,
"Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet
Engineering Task Force, Jan. 1997.
[6] M. Handley, "SDP: Session description protocol," Internet Draft,
Internet Engineering Task Force, Nov. 1996. Work in progress.
[7] A. Freier, P. Karlton, and P. Kocher, "The TLS protocol,"
Internet Draft, Internet Engineering Task Force, Dec. 1996. Work in
progress.
H. Schulzrinne, A. Rao, R. Lanphier [Page 58]
Internet Draft RTSP March 27, 1997
[8] J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and E.
L. Stewart, "An extension to HTTP: digest access authentication,"
RFC 2069, Internet Engineering Task Force, Jan. 1997.
[9] J. Postel, "User datagram protocol," STD 6, RFC 768, Internet
Engineering Task Force, Aug. 1980.
[10] R. Hinden and C. Partridge, "Version 2 of the reliable data
protocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr.
1990.
[11] J. Postel, "Transmission control protocol," STD 7, RFC 793,
Internet Engineering Task Force, Sept. 1981.
[12] M. Handley, H. Schulzrinne, and E. Schooler, "SIP: Session
initiation protocol," Internet Draft, Internet Engineering Task
Force, Dec. 1996. Work in progress.
[13] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
RFC 1961, Internet Engineering Task Force, June 1996.
[14] D. Crocker, "Augmented BNF for syntax specifications: ABNF,"
Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in
progress.
[15] R. Elz, "A compact representation of IPv6 addresses," RFC 1924,
Internet Engineering Task Force, Apr. 1996.
[16] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource
locators (URL)," RFC 1738, Internet Engineering Task Force, Dec.
1994.
[17] International Telecommunication Union, "Visual telephone systems
and equipment for local area networks which provide a non-guaranteed
quality of service," Recommendation H.323, Telecommunication
Standardization Sector of ITU, Geneva, Switzerland, May 1996.
[18] 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.
[19] D. Connolly, "PEP: an extension mechanism for http," Internet
Draft, Internet Engineering Task Force, Jan. 1997. Work in progress.
H. Schulzrinne, A. Rao, R. Lanphier [Page 59]
Internet Draft RTSP March 27, 1997
Table of Contents
1 Introduction ........................................ 1
1.1 Purpose ............................................. 1
1.2 Requirements ........................................ 3
1.3 Terminology ......................................... 3
1.4 Protocol Properties ................................. 5
1.5 Extending RTSP ...................................... 6
1.6 Overall Operation ................................... 7
1.7 RTSP States ......................................... 8
1.8 Relationship with Other Protocols ................... 9
2 Notational Conventions .............................. 10
3 Protocol Parameters ................................. 10
3.1 H3.1 ................................................ 10
3.2 RTSP URL ............................................ 10
3.3 Conference Identifiers .............................. 11
3.4 SMPTE Relative Timestamps ........................... 12
3.5 Normal Play Time .................................... 13
3.6 Absolute Time ....................................... 13
4 RTSP Message ........................................ 13
4.1 Message Types ....................................... 14
4.2 Message Headers ..................................... 14
4.3 Message Body ........................................ 14
4.4 Message Length ...................................... 14
5 Request ............................................. 15
6 Response ............................................ 16
6.1 Status-Line ......................................... 17
6.1.1 Status Code and Reason Phrase ....................... 17
6.1.2 Response Header Fields .............................. 19
7 Entity .............................................. 19
7.1 Entity Header Fields ................................ 21
7.2 Entity Body ......................................... 21
8 Connections ......................................... 21
8.1 Pipelining .......................................... 22
8.2 Reliability and Acknowledgements .................... 22
9 Method Definitions .................................. 23
9.1 OPTIONS ............................................. 24
9.2 DESCRIBE ........................................... 25
9.3 SETUP .............................................. 26
9.4 PLAY ............................................... 27
9.5 PAUSE .............................................. 28
9.6 TEARDOWN ........................................... 30
9.7 GET_PARAMETER ...................................... 30
9.8 SET_PARAMETER ...................................... 31
9.9 REDIRECT ........................................... 31
9.10 RECORD ............................................. 32
H. Schulzrinne, A. Rao, R. Lanphier [Page 60]
Internet Draft RTSP March 27, 1997
9.11 Embedded Binary Data ................................ 32
10 Status Code Definitions ............................. 33
10.1 Redirection 3xx ..................................... 33
10.2 Client Error 4xx .................................... 33
10.2.1 451 Parameter Not Understood ........................ 33
10.2.2 452 Conference Not Found ............................ 33
10.2.3 453 Not Enough Bandwidth ............................ 33
10.2.4 45x Session Not Found ............................... 33
10.2.5 45x Method Not Valid in This State .................. 33
10.2.6 45x Header Field Not Valid for Resource ............. 33
10.2.7 45x Invalid Range ................................... 33
10.2.8 45x Parameter Is Read-Only .......................... 34
11 Header Field Definitions ............................ 34
11.1 Accept .............................................. 34
11.2 Accept-Encoding ..................................... 35
11.3 Accept-Language ..................................... 35
11.4 Allow ............................................... 36
11.5 Authorization ....................................... 36
11.6 Bandwidth ........................................... 36
11.7 Blocksize ........................................... 36
11.8 Cache-Control ....................................... 37
11.9 Conference .......................................... 39
11.10 Connection .......................................... 39
11.11 Content-Encoding .................................... 39
11.12 Content-Length ...................................... 39
11.13 Content-Type ........................................ 39
11.14 Date ................................................ 40
11.15 Expires ............................................. 40
11.16 If-Modified-Since ................................... 41
11.17 Last-modified ....................................... 41
11.18 Location ............................................ 41
11.19 Nack-Transport-Require .............................. 41
11.20 Range ............................................... 41
11.21 Require ............................................. 42
11.22 Retry-After ......................................... 43
11.23 Scale ............................................... 43
11.24 Speed ............................................... 44
11.25 Server .............................................. 44
11.26 Session ............................................. 44
11.27 Transport ........................................... 45
11.28 Transport-Require ................................... 46
11.29 Unsupported ......................................... 46
11.30 User-Agent .......................................... 47
11.31 Via ................................................. 47
11.32 WWW-Authenticate .................................... 47
12 Caching ............................................. 47
13 Examples ............................................ 48
13.1 Media on Demand (Unicast) ........................... 48
H. Schulzrinne, A. Rao, R. Lanphier [Page 61]
Internet Draft RTSP March 27, 1997
13.2 Live Media Presentation Using Multicast ............. 49
13.3 Playing media into an existing session .............. 50
13.4 Recording ........................................... 51
14 Syntax .............................................. 52
14.1 Base Syntax ......................................... 52
15 Security Considerations ............................. 52
A RTSP Protocol State Machines ........................ 53
A.1 Client State Machine ................................ 54
A.2 Server State Machine ................................ 55
B Open Issues ......................................... 56
C Changes ............................................. 56
D Author Addresses .................................... 57
E Acknowledgements .................................... 57
F Bibliography ........................................ 58
H. Schulzrinne, A. Rao, R. Lanphier [Page 62]
