RFC2326
Network Working Group H. Schulzrinne Request for Comments: 2326 Columbia U. Category: Standards Track A. Rao
Netscape
R. Lanphier
RealNetworks
April 1998
Real Time Streaming Protocol (RTSP)
Status of this Memo
This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (1998). All Rights Reserved.
Abstract
The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 1889).
1 Introduction
Contents
- 1 Purpose
- 2 Requirements
- 3 Terminology
- 4 Protocol Properties
- 5 Extending RTSP
- 6 Overall Operation
- 7 RTSP States
- 8 Relationship with Other Protocols
- 9 RTSP Version
- 10 RTSP URL
- 11 Conference Identifiers
- 12 Session Identifiers
- 13 SMPTE Relative Timestamps
- 14 Normal Play Time
- 15 Absolute Time
- 16 Option Tags
- 17 Message Types
- 18 Message Headers
- 19 Message Body
- 20 Message Length
- 21 Request Line
- 22 Request Header Fields
- 23 Status-Line
- 24 Entity Header Fields
- 25 Entity Body
- 26 Pipelining
- 27 Reliability and Acknowledgements
Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls either a single or several time-synchronized streams of continuous media such as audio and video. It does not typically deliver the continuous streams itself, although interleaving of the continuous media stream with the control stream is possible (see Section 10.12). In other words, RTSP acts as a "network remote control" for multimedia servers.
The set of streams to be controlled is defined by a presentation description. This memorandum does not define a format for a presentation description.
There is no notion of an RTSP connection; instead, a server maintains a session labeled by an identifier. An RTSP session is in no way tied to a transport-level connection such as a TCP connection. During an RTSP session, an RTSP client may open and close many reliable transport connections to the server to issue RTSP requests. Alternatively, it may use a connectionless transport protocol such as UDP.
The streams controlled by RTSP may use RTP [1], but the operation of RTSP does not depend on the transport mechanism used to carry continuous media. The protocol is intentionally similar in syntax and operation to HTTP/1.1 [2] so that extension mechanisms to HTTP can in most cases also be added to RTSP. However, RTSP differs in a number of important aspects from HTTP:
* RTSP introduces a number of new methods and has a different protocol identifier. * An RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. * Both an RTSP server and client can issue requests. * Data is carried out-of-band by a different protocol. (There is an exception to this.) * RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, consistent with current HTML internationalization efforts [3]. * The Request-URI always contains the absolute URI. Because of backward compatibility with a historical blunder, HTTP/1.1 [2] carries only the absolute path in the request and puts the host name in a separate header field.
This makes "virtual hosting" easier, where a single host with one IP address hosts several document trees.
The protocol supports the following operations:
Retrieval of media from media server:
The client can request a presentation description via HTTP or
some other method. If the presentation is being multicast, the
presentation description contains the multicast addresses and
ports to be used for the continuous media. If the presentation
is to be sent only to the client via unicast, the client
provides the destination for security reasons.
Invitation of a media server to a conference:
A media server can be "invited" to join an existing
conference, either to play back media into the presentation or
to record all or a subset of the media in a presentation. This
mode is useful for distributed teaching applications. Several
parties in the conference may take turns "pushing the remote
control buttons."
Addition of media to an existing presentation:
Particularly for live presentations, it is useful if the
server can tell the client about additional media becoming
available.
RTSP requests may be handled by proxies, tunnels and caches as in HTTP/1.1 [2].
Requirements
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [4].
Terminology
Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not listed here are defined as in HTTP/1.1.
Aggregate control:
The control of the multiple streams using a single timeline by
the server. For audio/video feeds, this means that the client
may issue a single play or pause message to control both the
audio and video feeds.
Conference:
a multiparty, multimedia presentation, where "multi" implies
greater than or equal to one.
Client:
The client requests continuous media data from the media
server.
Connection:
A transport layer virtual circuit established between two
programs for the purpose of communication.
Container file:
A file which may contain multiple media streams which often
comprise a presentation when played together. RTSP servers may
offer aggregate control on these files, though the concept of
a container file is not embedded in the protocol.
Continuous media:
Data where there is a timing relationship between source and
sink; that is, the sink must reproduce the timing relationship
that existed at the source. The most common examples of
continuous media are audio and motion video. Continuous media
can be real-time (interactive), where there is a "tight"
timing relationship between source and sink, or streaming
(playback), where the relationship is less strict.
Entity:
The information transferred as the payload of a request or
response. An entity consists of metainformation in the form of
entity-header fields and content in the form of an entity-
body, as described in Section 8.
Media initialization:
Datatype/codec specific initialization. This includes such
things as clockrates, color tables, etc. Any transport-
independent information which is required by a client for
playback of a media stream occurs in the media initialization
phase of stream setup.
Media parameter:
Parameter specific to a media type that may be changed before
or during stream playback.
Media server:
The server providing playback or recording services for one or
more media streams. Different media streams within a
presentation may originate from different media servers. A
media server may reside on the same or a different host as the
web server the presentation is invoked from.
Media server indirection:
Redirection of a media client to a different media server.
(Media) stream:
A single media instance, e.g., an audio stream or a video
stream as well as a single whiteboard or shared application
group. When using RTP, a stream consists of all RTP and RTCP
packets created by a source within an RTP session. This is
equivalent to the definition of a DSM-CC stream([5]).
Message:
The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined in
Section 15 and transmitted via a connection or a
connectionless protocol.
Participant:
Member of a conference. A participant may be a machine, e.g.,
a media record or playback server.
Presentation:
A set of one or more streams presented to the client as a
complete media feed, using a presentation description as
defined below. In most cases in the RTSP context, this implies
aggregate control of those streams, but does not have to.
Presentation description:
A presentation description contains information about one or
more media streams within a presentation, such as the set of
encodings, network addresses and information about the
content. Other IETF protocols such as SDP (RFC 2327 [6]) use
the term "session" for a live presentation. The presentation
description may take several different formats, including but
not limited to the session description format SDP.
Response:
An RTSP response. If an HTTP response is meant, that is
indicated explicitly.
Request:
An RTSP request. If an HTTP request is meant, that is
indicated explicitly.
RTSP session:
A complete RTSP "transaction", e.g., the viewing of a movie.
A session typically consists of a client setting up a
transport mechanism for the continuous media stream (SETUP),
starting the stream with PLAY or RECORD, and closing the
stream with TEARDOWN.
Transport initialization:
The negotiation of transport information (e.g., port numbers,
transport protocols) between the client and the server.
Protocol Properties
RTSP has the following properties:
Extendable:
New methods and parameters can be easily added to RTSP.
Easy to parse:
RTSP can be parsed by standard HTTP or MIME parsers.
Secure:
RTSP re-uses web security mechanisms. All HTTP authentication
mechanisms such as basic (RFC 2068 [2, Section 11.1]) and
digest authentication (RFC 2069 [8]) are directly applicable.
One may also reuse transport or network layer security
mechanisms.
Transport-independent:
RTSP may use either an unreliable datagram protocol (UDP) (RFC
768 [9]), a reliable datagram protocol (RDP, RFC 1151, not
widely used [10]) or a reliable stream protocol such as TCP
(RFC 793 [11]) as it implements application-level reliability.
Multi-server capable:
Each media stream within a presentation can reside on a
different server. The client automatically establishes several
concurrent control sessions with the different media servers.
Media synchronization is performed at the transport level.
Control of recording devices:
The protocol can control both recording and playback devices,
as well as devices that can alternate between the two modes
("VCR").
Separation of stream control and conference initiation:
Stream control is divorced from inviting a media server to a
conference. The only requirement is that the conference
initiation protocol either provides or can be used to create a
unique conference identifier. In particular, SIP [12] or H.323
[13] may be used to invite a server to a conference.
Suitable for professional applications:
RTSP supports frame-level accuracy through SMPTE time stamps
to allow remote digital editing.
Presentation description neutral:
The protocol does not impose a particular presentation
description or metafile format and can convey the type of
format to be used. However, the presentation description must
contain at least one RTSP URI.
Proxy and firewall friendly:
The protocol should be readily handled by both application and
transport-layer (SOCKS [14]) firewalls. A firewall may need to
understand the SETUP method to open a "hole" for the UDP media
stream.
HTTP-friendly:
Where sensible, RTSP reuses HTTP concepts, so that the
existing infrastructure can be reused. This infrastructure
includes PICS (Platform for Internet Content Selection
[15,16]) for associating labels with content. However, RTSP
does not just add methods to HTTP since the controlling
continuous media requires server state in most cases.
Appropriate server control:
If a client can start a stream, it must be able to stop a
stream. Servers should not start streaming to clients in such
a way that clients cannot stop the stream.
Transport negotiation:
The client can negotiate the transport method prior to
actually needing to process a continuous media stream.
Capability negotiation:
If basic features are disabled, there must be some clean
mechanism for the client to determine which methods are not
going to be implemented. This allows clients to present the
appropriate user interface. For example, if seeking is not
allowed, the user interface must be able to disallow moving a
sliding position indicator.
An earlier requirement in RTSP was multi-client capability. However, it was determined that a better approach was to make sure that the protocol is easily extensible to the multi-client scenario. Stream identifiers can be used by several control streams, so that "passing the remote" would be possible. The protocol would not address how several clients negotiate access; this is left to either a "social protocol" or some other floor
control mechanism.
Extending RTSP
Since not all media servers have the same functionality, media servers by necessity will support different sets of requests. For example:
* A server may only be capable of playback thus has no need to support the RECORD request. * A server may not be capable of seeking (absolute positioning) if it is to support live events only. * Some servers may not support setting stream parameters and thus not support GET_PARAMETER and SET_PARAMETER.
A server SHOULD implement all header fields described in Section 12.
It is up to the creators of presentation descriptions not to ask the impossible of a server. This situation is similar in HTTP/1.1 [2], where the methods described in [H19.6] are not likely to be supported across all servers.
RTSP can be extended in three ways, listed here in order of the magnitude of changes supported:
* Existing methods can be extended with new parameters, as long as these parameters can be safely ignored by the recipient. (This is equivalent to adding new parameters to an HTML tag.) If the client needs negative acknowledgement when a method extension is not supported, a tag corresponding to the extension may be added in the Require: field (see Section 12.32). * New methods can be added. If the recipient of the message does not understand the request, it responds with error code 501 (Not implemented) and the sender should not attempt to use this method again. A client may also use the OPTIONS method to inquire about methods supported by the server. The server SHOULD list the methods it supports using the Public response header. * A new version of the protocol can be defined, allowing almost all aspects (except the position of the protocol version number) to change.
Overall Operation
Each presentation and media stream may be identified by an RTSP URL. The overall presentation and the properties of the media the presentation is made up of are defined by a presentation description file, the format of which is outside the scope of this specification. The presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored on the media server.
For the purposes of this specification, a presentation description is assumed to describe one or more presentations, each of which maintains a common time axis. For simplicity of exposition and without loss of generality, it is assumed that the presentation description contains exactly one such presentation. A presentation may contain several media streams.
The presentation description file contains a description of the media streams making up the presentation, including their encodings, language, and other parameters that enable the client to choose the most appropriate combination of media. In this presentation description, each media stream that is individually controllable by RTSP is identified by an RTSP URL, which points to the media server handling that particular media stream and names the stream stored on that server. Several media streams can be located on different servers; for example, audio and video streams can be split across servers for load sharing. The description also enumerates which transport methods the server is capable of.
Besides the media parameters, the network destination address and port need to be determined. Several modes of operation can be distinguished:
Unicast:
The media is transmitted to the source of the RTSP request,
with the port number chosen by the client. Alternatively, the
media is transmitted on the same reliable stream as RTSP.
Multicast, server chooses address:
The media server picks the multicast address and port. This is
the typical case for a live or near-media-on-demand
transmission.
Multicast, client chooses address:
If the server is to participate in an existing multicast
conference, the multicast address, port and encryption key are
given by the conference description, established by means
outside the scope of this specification.
RTSP States
RTSP controls a stream which may be sent via a separate protocol, independent of the control channel. For example, RTSP control may occur on a TCP connection while the data flows via UDP. Thus, data delivery continues even if no RTSP requests are received by the media
server. Also, during its lifetime, a single media stream may be controlled by RTSP requests issued sequentially on different TCP connections. Therefore, the server needs to maintain "session state" to be able to correlate RTSP requests with a stream. The state transitions are described in Section A.
Many methods in RTSP do not contribute to state. However, the following play a central role in defining the allocation and usage of stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and TEARDOWN.
SETUP:
Causes the server to allocate resources for a stream and start
an RTSP session.
PLAY and RECORD:
Starts data transmission on a stream allocated via SETUP.
PAUSE:
Temporarily halts a stream without freeing server resources.
TEARDOWN:
Frees resources associated with the stream. The RTSP session
ceases to exist on the server.
RTSP methods that contribute to state use the Session header
field (Section 12.37) to identify the RTSP session whose state
is being manipulated. The server generates session identifiers
in response to SETUP requests (Section 10.4).
Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may interact with HTTP in that the initial contact with streaming content is often to be made through a web page. The current protocol specification aims to allow different hand-off points between a web server and the media server implementing RTSP. For example, the presentation description can be retrieved using HTTP or RTSP, which reduces roundtrips in web-browser-based scenarios, yet also allows for standalone RTSP servers and clients which do not rely on HTTP at all.
However, RTSP differs fundamentally from HTTP in that data delivery takes place out-of-band in a different protocol. HTTP is an asymmetric protocol where the client issues requests and the server responds. In RTSP, both the media client and media server can issue requests. RTSP requests are also not stateless; they may set parameters and continue to control a media stream long after the
request has been acknowledged.
Re-using HTTP functionality has advantages in at least two areas, namely security and proxies. The requirements are very similar, so having the ability to adopt HTTP work on caches, proxies and authentication is valuable.
While most real-time media will use RTP as a transport protocol, RTSP is not tied to RTP.
RTSP assumes the existence of a presentation description format that can express both static and temporal properties of a presentation containing several media streams.
2 Notational Conventions
Since many of the definitions and syntax are identical to HTTP/1.1, this specification only points to the section where they are defined rather than copying it. For brevity, [HX.Y] is to be taken to refer to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]).
All the mechanisms specified in this document are described in both prose and an augmented Backus-Naur form (BNF) similar to that used in [H2.1]. It is described in detail in RFC 2234 [17], with the difference that this RTSP specification maintains the "1#" notation for comma-separated lists.
In this memo, we use indented and smaller-type paragraphs to provide background and motivation. This is intended to give readers who were not involved with the formulation of the specification an understanding of why things are the way that they are in RTSP.
3 Protocol Parameters
RTSP Version
[H3.1] applies, with HTTP replaced by RTSP.
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 \cite{rfc1123}>
port = *DIGIT
abs_path is defined in [H3.2.1].
Note that fragment and query identifiers do not have a well-defined meaning at this time, with the interpretation left to the RTSP server.
The scheme rtsp requires that commands are issued via a reliable protocol (within the Internet, TCP), while the scheme rtspu identifies an unreliable protocol (within the Internet, UDP).
If the port is empty or not given, port 554 is assumed. The semantics are that the identified resource can be controlled by RTSP at the server listening for TCP (scheme "rtsp") connections or UDP (scheme "rtspu") packets on that port of host, and the Request-URI for the resource is rtsp_URL.
The use of IP addresses in URLs SHOULD be avoided whenever possible (see RFC 1924 [19]).
A presentation or a stream is identified by a textual media identifier, using the character set and escape conventions [H3.2] of URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of streams, i.e., a presentation. Accordingly, requests described in Section 10 can apply to either the whole presentation or an individual stream within the presentation. Note that some request methods can only be applied to streams, not presentations and vice versa.
For example, the RTSP URL:
rtsp://media.example.com:554/twister/audiotrack
identifies the audio stream within the presentation "twister", which can be controlled via RTSP requests issued over a TCP connection to port 554 of host media.example.com.
Also, the RTSP URL:
rtsp://media.example.com:554/twister
identifies the presentation "twister", which may be composed of audio and video streams.
This does not imply a standard way to reference streams in URLs. The presentation description defines the hierarchical relationships in the presentation and the URLs for the individual streams. A presentation description may name a stream "a.mov" and the whole presentation "b.mov".
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.
Conference Identifiers
Conference identifiers are opaque to RTSP and are encoded using standard URI encoding methods (i.e., LWS is escaped with %). They can contain any octet value. The conference identifier MUST be globally unique. For H.323, the conferenceID value is to be used.
conference-id = 1*xchar
Conference identifiers are used to allow RTSP sessions to obtain parameters from multimedia conferences the media server is participating in. These conferences are created by protocols outside the scope of this specification, e.g., H.323 [13] or SIP [12]. Instead of the RTSP client explicitly providing transport information, for example, it asks the media server to use the values in the conference description instead.
Session Identifiers
Session identifiers are opaque strings of arbitrary length. Linear white space must be URL-escaped. A session identifier MUST be chosen randomly and MUST be at least eight octets long to make guessing it more difficult. (See Section 16.)
session-id = 1*( ALPHA | DIGIT | safe )
SMPTE Relative Timestamps
A SMPTE relative timestamp expresses time relative to the start of the clip. Relative timestamps are expressed as SMPTE time codes for frame-level access accuracy. The time code has the format hours:minutes:seconds:frames.subframes, with the origin at the start of the clip. The default smpte format is "SMPTE 30 drop" format, with frame rate is 29.97 frames per second. Other SMPTE codes MAY be supported (such as "SMPTE 25") through the use of alternative use of "smpte time". For the "frames" field in the time value can assume the values 0 through 29. The difference between 30 and 29.97 frames per second is handled by dropping the first two frame indices (values 00 and 01) of every minute, except every tenth minute. If the frame value is zero, it may be omitted. Subframes are measured in one-hundredth of a frame.
smpte-range = smpte-type "=" smpte-time "-" [ smpte-time ] smpte-type = "smpte" | "smpte-30-drop" | "smpte-25"
; other timecodes may be added
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ]
[ "." 1*2DIGIT ]
Examples:
smpte=10:12:33:20- smpte=10:07:33- smpte=10:07:00-10:07:33:05.01 smpte-25=10:07:00-10:07:33:05.01
Normal Play Time
Normal play time (NPT) indicates the stream absolute position relative to the beginning of the presentation. The timestamp consists of a decimal fraction. The part left of the decimal may be expressed in either seconds or hours, minutes, and seconds. The part right of the decimal point measures fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative values are not defined. The special constant now is defined as the current instant of a live event. It may be used only for live events.
NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the viewer associates with a program. It is often digitally displayed on a VCR. NPT advances normally when in normal play mode (scale = 1), advances at a faster rate when in fast scan forward (high positive scale ratio), decrements when in scan reverse (high negative scale ratio) and is fixed in pause mode. NPT is (logically) equivalent to SMPTE time codes." [5]
npt-range = ( npt-time "-" [ npt-time ] ) | ( "-" npt-time ) npt-time = "now" | npt-sec | npt-hhmmss npt-sec = 1*DIGIT [ "." *DIGIT ] npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ] npt-hh = 1*DIGIT ; any positive number npt-mm = 1*2DIGIT ; 0-59 npt-ss = 1*2DIGIT ; 0-59
Examples:
npt=123.45-125 npt=12:05:35.3- npt=now-
The syntax conforms to ISO 8601. The npt-sec notation is optimized for automatic generation, the ntp-hhmmss notation for consumption by human readers. The "now" constant allows clients to request to
receive the live feed rather than the stored or time-delayed version. This is needed since neither absolute time nor zero time are appropriate for this case.
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
Option Tags
Option tags are unique identifiers used to designate new options in RTSP. These tags are used in Require (Section 12.32) and Proxy- Require (Section 12.27) header fields.
Syntax:
option-tag = 1*xchar
The creator of a new RTSP option should either prefix the option with a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name for a feature whose inventor can be reached at "foo.com"), or register the new option with the Internet Assigned Numbers Authority (IANA).
Registering New Option Tags with IANA
When registering a new RTSP option, the following information should be provided:
* Name and description of option. The name may be of any length, but SHOULD be no more than twenty characters long. The name MUST not contain any spaces, control characters or periods. * Indication of who has change control over the option (for example, IETF, ISO, ITU-T, other international standardization bodies, a consortium or a particular company or group of companies);
* A reference to a further description, if available, for example (in order of preference) an RFC, a published paper, a patent filing, a technical report, documented source code or a computer manual; * For proprietary options, contact information (postal and email address);
4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2279 [21]). Lines are terminated by CRLF, but receivers should be prepared to also interpret CR and LF by themselves as line terminators.
Text-based protocols make it easier to add optional parameters in a self-describing manner. Since the number of parameters and the frequency of commands is low, processing efficiency is not a concern. Text-based protocols, if done carefully, also allow easy implementation of research prototypes in scripting languages such as Tcl, Visual Basic and Perl.
The 10646 character set avoids tricky character set switching, but is invisible to the application as long as US-ASCII is being used. This is also the encoding used for RTCP. ISO 8859-1 translates directly into Unicode with a high-order octet of zero. ISO 8859-1 characters with the most-significant bit set are represented as 1100001x 10xxxxxx. (See RFC 2279 [21])
RTSP messages can be carried over any lower-layer transport protocol that is 8-bit clean.
Requests contain methods, the object the method is operating upon and parameters to further describe the method. Methods are idempotent, unless otherwise noted. Methods are also designed to require little or no state maintenance at the media server.
Message Types
See [H4.1]
Message Headers
See [H4.2]
Message Body
See [H4.3]
Message Length
When a message body is included with a message, the length of that body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message body
(such as the 1xx, 204, and 304 responses) is always terminated
by the first empty line after the header fields, regardless of
the entity-header fields present in the message. (Note: An
empty line consists of only CRLF.)
2. If a Content-Length header field (section 12.14) is present,
its value in bytes represents the length of the message-body.
If this header field is not present, a value of zero is
assumed.
3. By the server closing the connection. (Closing the connection
cannot be used to indicate the end of a request body, since
that would leave no possibility for the server to send back a
response.)
Note that RTSP does not (at present) support the HTTP/1.1 "chunked" transfer coding(see [H3.6]) and requires the presence of the Content-Length header field.
Given the moderate length of presentation descriptions returned, the server should always be able to determine its length, even if it is generated dynamically, making the chunked transfer encoding unnecessary. Even though Content-Length must be present if there is any entity body, the rules ensure reasonable behavior even if the length is not given explicitly.
5 General Header Fields
See [H4.5], except that Pragma, Transfer-Encoding and Upgrade headers are not defined:
general-header = Cache-Control ; Section 12.8
| Connection ; Section 12.10
| Date ; Section 12.18
| Via ; Section 12.43
6 Request
A request message from a client to a server or vice versa includes, within the first line of that message, the method to be applied to the resource, the identifier of the resource, and the protocol version in use.
Request = Request-Line ; Section 6.1
*( general-header ; Section 5
| request-header ; Section 6.2
| entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
Request Line
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Method = "DESCRIBE" ; Section 10.2
| "ANNOUNCE" ; Section 10.3
| "GET_PARAMETER" ; Section 10.8
| "OPTIONS" ; Section 10.1
| "PAUSE" ; Section 10.6
| "PLAY" ; Section 10.5
| "RECORD" ; Section 10.11
| "REDIRECT" ; Section 10.10
| "SETUP" ; Section 10.4
| "SET_PARAMETER" ; Section 10.9
| "TEARDOWN" ; Section 10.7
| extension-method
extension-method = token
Request-URI = "*" | absolute_URI
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
Request Header Fields
request-header = Accept ; Section 12.1
| Accept-Encoding ; Section 12.2
| Accept-Language ; Section 12.3
| Authorization ; Section 12.5
| From ; Section 12.20
| If-Modified-Since ; Section 12.23
| Range ; Section 12.29
| Referer ; Section 12.30
| User-Agent ; Section 12.41
Note that in contrast to HTTP/1.1 [2], RTSP requests always contain the absolute URL (that is, including the scheme, host and port) rather than just the absolute path.
HTTP/1.1 requires servers to understand the absolute URL, but clients are supposed to use the Host request header. This is purely needed for backward-compatibility with HTTP/1.0 servers, a consideration that does not apply to RTSP.
The asterisk "*" in the Request-URI means that the request does not apply to a particular resource, but to the server itself, and is only allowed when the method used does not necessarily apply to a resource. One example would be:
OPTIONS * RTSP/1.0
7 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version. Also, RTSP defines additional status codes and does not define some HTTP codes. The valid response codes and the methods they can be used with are defined in Table 1.
After receiving and interpreting a request message, the recipient responds with an RTSP response message.
Response = Status-Line ; Section 7.1
*( general-header ; Section 5
| response-header ; Section 7.1.2
| entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
Status-Line
The first line of a Response message is the Status-Line, consisting of the protocol version followed by a numeric status code, and the textual phrase associated with the status code, with each element separated by SP characters. No CR or LF is allowed except in the final CRLF sequence.
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the attempt to understand and satisfy the request. These codes are fully defined in Section 11. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata and the Reason-Phrase is intended for the human user. The client is not required to examine or display the Reason- Phrase.
The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. There are 5 values for the first digit:
* 1xx: Informational - Request received, continuing process * 2xx: Success - The action was successfully received, understood, and accepted * 3xx: Redirection - Further action must be taken in order to complete the request * 4xx: Client Error - The request contains bad syntax or cannot be fulfilled * 5xx: Server Error - The server failed to fulfill an apparently valid request
The individual values of the numeric status codes defined for RTSP/1.0, and an example set of corresponding Reason-Phrase's, are presented below. The reason phrases listed here are only recommended - they may be replaced by local equivalents without affecting the protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and adds RTSP-specific status codes starting at x50 to avoid conflicts with newly defined HTTP status codes.
Status-Code = "100" ; Continue
| "200" ; OK
| "201" ; Created
| "250" ; Low on Storage Space
| "300" ; Multiple Choices
| "301" ; Moved Permanently
| "302" ; Moved Temporarily
| "303" ; See Other
| "304" ; Not Modified
| "305" ; Use Proxy
| "400" ; Bad Request
| "401" ; Unauthorized
| "402" ; Payment Required
| "403" ; Forbidden
| "404" ; Not Found
| "405" ; Method Not Allowed
| "406" ; Not Acceptable
| "407" ; Proxy Authentication Required
| "408" ; Request Time-out
| "410" ; Gone
| "411" ; Length Required
| "412" ; Precondition Failed
| "413" ; Request Entity Too Large
| "414" ; Request-URI Too Large
| "415" ; Unsupported Media Type
| "451" ; Parameter Not Understood
| "452" ; Conference Not Found
| "453" ; Not Enough Bandwidth
| "454" ; Session Not Found
| "455" ; Method Not Valid in This State
| "456" ; Header Field Not Valid for Resource
| "457" ; Invalid Range
| "458" ; Parameter Is Read-Only
| "459" ; Aggregate operation not allowed
| "460" ; Only aggregate operation allowed
| "461" ; Unsupported transport
| "462" ; Destination unreachable
| "500" ; Internal Server Error
| "501" ; Not Implemented
| "502" ; Bad Gateway
| "503" ; Service Unavailable
| "504" ; Gateway Time-out
| "505" ; RTSP Version not supported
| "551" ; Option not supported
| extension-code
extension-code = 3DIGIT
Reason-Phrase = *<TEXT, excluding CR, LF>
RTSP status codes are extensible. RTSP applications are not required to understand the meaning of all registered status codes, though such understanding is obviously desirable. However, applications MUST understand the class of any status code, as indicated by the first digit, and treat any unrecognized response as being equivalent to the x00 status code of that class, with the exception that an unrecognized response MUST NOT be cached. For example, if an unrecognized status code of 431 is received by the client, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 status code. In such cases, user agents SHOULD present to the user the entity returned with the response, since that entity is likely to include human- readable information which will explain the unusual status.
Code reason
100 Continue all
200 OK all 201 Created RECORD 250 Low on Storage Space RECORD
300 Multiple Choices all 301 Moved Permanently all 302 Moved Temporarily all 303 See Other all 305 Use Proxy all
400 Bad Request all 401 Unauthorized all 402 Payment Required all 403 Forbidden all 404 Not Found all 405 Method Not Allowed all 406 Not Acceptable all 407 Proxy Authentication Required all 408 Request Timeout all 410 Gone all 411 Length Required all 412 Precondition Failed DESCRIBE, SETUP 413 Request Entity Too Large all 414 Request-URI Too Long all 415 Unsupported Media Type all 451 Invalid parameter SETUP 452 Illegal Conference Identifier SETUP 453 Not Enough Bandwidth SETUP 454 Session Not Found all 455 Method Not Valid In This State all 456 Header Field Not Valid all 457 Invalid Range PLAY 458 Parameter Is Read-Only SET_PARAMETER 459 Aggregate Operation Not Allowed all 460 Only Aggregate Operation Allowed all 461 Unsupported Transport all 462 Destination Unreachable all
500 Internal Server Error all 501 Not Implemented all 502 Bad Gateway all 503 Service Unavailable all 504 Gateway Timeout all 505 RTSP Version Not Supported all 551 Option not support all
Table 1: Status codes and their usage with RTSP methods
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. These header fields give information about the server and about further access to the resource identified by the Request-URI.
response-header = Location ; Section 12.25
| Proxy-Authenticate ; Section 12.26
| Public ; Section 12.28
| Retry-After ; Section 12.31
| Server ; Section 12.36
| Vary ; Section 12.42
| WWW-Authenticate ; Section 12.44
Response-header field names can be extended reliably only in combination with a change in the protocol version. However, new or experimental header fields MAY be given the semantics of response- header fields if all parties in the communication recognize them to be response-header fields. Unrecognized header fields are treated as entity-header fields.
8 Entity
Request and Response messages MAY transfer an entity if not otherwise restricted by the request method or response status code. An entity consists of entity-header fields and an entity-body, although some responses will only include the entity-headers.
In this section, both sender and recipient refer to either the client or the server, depending on who sends and who receives the entity.
Entity Header Fields
Entity-header fields define optional metainformation about the entity-body or, if no body is present, about the resource identified by the request.
entity-header = Allow ; Section 12.4
| Content-Base ; Section 12.11
| Content-Encoding ; Section 12.12
| Content-Language ; Section 12.13
| Content-Length ; Section 12.14
| Content-Location ; Section 12.15
| Content-Type ; Section 12.16
| Expires ; Section 12.19
| Last-Modified ; Section 12.24
| extension-header
extension-header = message-header
The extension-header mechanism allows additional entity-header fields to be defined without changing the protocol, but these fields cannot be assumed to be recognizable by the recipient. Unrecognized header fields SHOULD be ignored by the recipient and forwarded by proxies.
Entity Body
See [H7.2]
9 Connections
RTSP requests can be transmitted in several different ways:
* persistent transport connections used for several request-response transactions; * one connection per request/response transaction; * connectionless mode.
The type of transport connection is defined by the RTSP URI (Section 3.2). For the scheme "rtsp", a persistent connection is assumed, while the scheme "rtspu" calls for RTSP requests to be sent without setting up a connection.
Unlike HTTP, RTSP allows the media server to send requests to the media client. However, this is only supported for persistent 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.
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.
Reliability and Acknowledgements
Requests are acknowledged by the receiver unless they are sent to a multicast group. If there is no acknowledgement, the sender may resend the same message after a timeout of one round-trip time (RTT). The round-trip time is estimated as in TCP (RFC 1123) [18], with an initial round-trip value of 500 ms. An implementation MAY cache the last RTT measurement as the initial value for future connections.
If a reliable transport protocol is used to carry RTSP, requests MUST NOT be retransmitted; the RTSP application MUST instead rely on the underlying transport to provide reliability.
If both the underlying reliable transport such as TCP and the RTSP application retransmit requests, it is possible that each packet loss results in two retransmissions. The receiver cannot typically take advantage of the application-layer retransmission since the
transport stack will not deliver the application-layer retransmission before the first attempt has reached the receiver. If the packet loss is caused by congestion, multiple retransmissions at different layers will exacerbate the congestion.
If RTSP is used over a small-RTT LAN, standard procedures for optimizing initial TCP round trip estimates, such as those used in T/TCP (RFC 1644) [22], can be beneficial.
The Timestamp header (Section 12.38) is used to avoid the retransmission ambiguity problem [23, p. 301] and obviates the need for Karn's algorithm.
Each request carries a sequence number in the CSeq header (Section 12.17), which is incremented by one for each distinct request transmitted. If a request is repeated because of lack of acknowledgement, the request MUST carry the original sequence number (i.e., the sequence number is not incremented).
Systems implementing RTSP MUST support carrying RTSP over TCP and MAY support UDP. The default port for the RTSP server is 554 for both UDP and TCP.
A number of RTSP packets destined for the same control end point may be packed into a single lower-layer PDU or encapsulated into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP packets. Unlike HTTP, an RTSP message MUST contain a Content-Length header whenever that message contains a payload. Otherwise, an RTSP packet is terminated with an empty line immediately following the last message header.
10 Method Definitions
The method token indicates the method to be performed on the resource identified by the Request-URI. The method is case-sensitive. New methods may be defined in the future. Method names may not start with a $ character (decimal 24) and must be a token. Methods are summarized in Table 2.
method direction object requirement
DESCRIBE C->S P,S recommended
ANNOUNCE C->S, S->C P,S optional
GET_PARAMETER C->S, S->C P,S optional
OPTIONS C->S, S->C P,S required
(S->C: optional)
PAUSE C->S P,S recommended
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 recommended, but not required in that a fully functional server can be built that does not support this method, for example, for live feeds. If a server does not support a particular method, it MUST return "501 Not Implemented" and a client SHOULD not try this method again for this server.
10.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An OPTIONS request may be issued at any time, e.g., if the client is about to try a nonstandard request. It does not influence server state.
Example:
C->S: OPTIONS * RTSP/1.0
CSeq: 1
Require: implicit-play
Proxy-Require: gzipped-messages
S->C: RTSP/1.0 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are necessarily fictional features (one would hope that we would not purposefully overlook a truly useful feature just so that we could have a strong example in this section).
10.2 DESCRIBE
The DESCRIBE method retrieves the description of a presentation or media object identified by the request URL from a server. It may use the Accept header to specify the description formats that the client understands. The server responds with a description of the requested resource. The DESCRIBE reply-response pair constitutes the media initialization phase of RTSP.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 OK
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 376
v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
[email protected] (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
m=whiteboard 32416 UDP WB
a=orient:portrait
The DESCRIBE response MUST contain all media initialization information for the resource(s) that it describes. If a media client obtains a presentation description from a source other than DESCRIBE and that description contains a complete set of media initialization parameters, the client SHOULD use those parameters and not then request a description for the same media via RTSP.
Additionally, servers SHOULD NOT use the DESCRIBE response as a means of media indirection.
Clear ground rules need to be established so that clients have an unambiguous means of knowing when to request media initialization information via DESCRIBE, and when not to. By forcing a DESCRIBE
response to contain all media initialization for the set of streams that it describes, and discouraging use of DESCRIBE for media indirection, we avoid looping problems that might result from other approaches.
Media initialization is a requirement for any RTSP-based system, but the RTSP specification does not dictate that this must be done via the DESCRIBE method. There are three ways that an RTSP client may receive initialization information:
* via RTSP's DESCRIBE method; * via some other protocol (HTTP, email attachment, etc.); * via the command line or standard input (thus working as a browser helper application launched with an SDP file or other media initialization format).
In the interest of practical interoperability, it is highly recommended that minimal servers support the DESCRIBE method, and highly recommended that minimal clients support the ability to act as a "helper application" that accepts a media initialization file from standard input, command line, and/or other means that are appropriate to the operating environment of the client.
10.3 ANNOUNCE
The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a presentation or media object identified by the request URL to a server. When sent from server to client, ANNOUNCE updates the session description in real-time.
If a new media stream is added to a presentation (e.g., during a live presentation), the whole presentation description should be sent again, rather than just the additional components, so that components can be deleted.
Example:
C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Content-Type: application/sdp
Content-Length: 332
v=0
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
[email protected] (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
S->C: RTSP/1.0 200 OK
CSeq: 312
10.4 SETUP
The SETUP request for a URI specifies the transport mechanism to be used for the streamed media. A client can issue a SETUP request for a stream that is already playing to change transport parameters, which a server MAY allow. If it does not allow this, it MUST respond with error "455 Method Not Valid In This State". For the benefit of any intervening firewalls, a client must indicate the transport parameters even if it has no influence over these parameters, for example, where the server advertises a fixed multicast address.
Since SETUP includes all transport initialization information, firewalls and other intermediate network devices (which need this information) are spared the more arduous task of parsing the DESCRIBE response, which has been reserved for media initialization.
The Transport header specifies the transport parameters acceptable to the client for data transmission; the response will contain the transport parameters selected by the server.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Transport: RTP/AVP;unicast;client_port=4588-4589
S->C: RTSP/1.0 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;
client_port=4588-4589;server_port=6256-6257
The server generates session identifiers in response to SETUP requests. If a SETUP request to a server includes a session identifier, the server MUST bundle this setup request into the
existing session or return error "459 Aggregate Operation Not Allowed" (see Section 11.3.10).
10.5 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 requests in the example below arrive, the server will first play seconds 10 through 15, then, immediately following, seconds 20 to 25, and finally seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 835
Session: 12345678
Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 836
Session: 12345678
Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 837
Session: 12345678
Range: npt=30-
See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It starts playing a stream from the beginning unless the stream has been paused. If a stream has been paused via PAUSE, stream delivery resumes at the 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.
The Range header may also contain a time parameter. This parameter specifies a time in UTC at which the playback should start. If the message is received after the specified time, playback is started immediately. The time parameter may be used to aid in synchronization of streams obtained from different sources.
For a on-demand stream, the server replies with the actual range that will be played back. This may differ from the requested range if alignment of the requested range to valid frame boundaries is required for the media source. If no range is specified in the request, the current position is returned in the reply. The unit of the range in the reply is the same as that in the request.
After playing the desired range, the presentation is automatically paused, as if a PAUSE request had been issued.
The following example plays the whole presentation starting at SMPTE time code 0:10:20 until the end of the clip. The playback is to start at 15:36 on 23 Jan 1997.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
CSeq: 833
Session: 12345678
Range: smpte=0:10:20-;time=19970123T153600Z
S->C: RTSP/1.0 200 OK
CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT
Range: smpte=0:10:22-;time=19970123T153600Z
For playing back a recording of a live presentation, it may be desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
CSeq: 835
Session: 12345678
Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the npt format and MAY support the clock and smpte formats.
10.6 PAUSE
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, though servers MAY close the session and free resources after being paused for the duration specified with the timeout parameter of the Session header in the SETUP message.
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
The PAUSE request may contain a Range header specifying when the stream or presentation is to be halted. We refer to this point as the "pause point". The header must contain exactly one value rather than a time range. The normal play time for the stream is set to the pause point. The pause request becomes effective the first time the server is encountering the time point specified in any of the currently pending PLAY requests. If the Range header specifies a time outside any currently pending PLAY requests, the error "457 Invalid Range" is returned. If a media unit (such as an audio or video frame) starts presentation at exactly the pause point, it is not played or recorded. If the Range header is missing, stream delivery is interrupted immediately on receipt of the message and the pause point is set to the current normal play time.
A PAUSE request discards all queued PLAY requests. However, the pause point in the media stream MUST be maintained. A subsequent PLAY request without Range header resumes from the pause point.
For example, if the server has play requests for ranges 10 to 15 and 20 to 29 pending and then receives a pause request for NPT 21, it would start playing the second range and stop at NPT 21. If the pause request is for NPT 12 and the server is playing at NPT 13 serving the first play request, the server stops immediately. If the pause request is for NPT 16, the server stops after completing the first
play request and discards the second play request.
As another example, if a server has received requests to play ranges 10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE request for NPT=14 would take effect while the server plays the first range, with the second PLAY request effectively being ignored, 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.
10.7 TEARDOWN
The TEARDOWN request stops the stream delivery for the given URI, freeing the resources associated with it. If the URI is the presentation URI 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 rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 892
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 892
10.8 GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a presentation or stream specified in the URI. The content of the reply and response is left to the implementation. GET_PARAMETER with no entity body may be used to test client or server liveness ("ping").
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 431
Content-Type: text/parameters
Session: 12345678
Content-Length: 15
packets_received
jitter
C->S: RTSP/1.0 200 OK
CSeq: 431
Content-Length: 46
Content-Type: text/parameters
packets_received: 10
jitter: 0.3838
The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation.
10.9 SET_PARAMETER
This method requests to set the value of a parameter for a 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. If the request contains several parameters, the server MUST only act on the request if all of the parameters can be set successfully. A server MUST allow a parameter to be set repeatedly to the same value, but it 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.
Example:
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 421
Content-length: 20
Content-type: text/parameters
barparam: barstuff
S->C: RTSP/1.0 451 Invalid Parameter
CSeq: 421
Content-length: 10
Content-type: text/parameters
barparam
The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation.
10.10 REDIRECT
A redirect request informs the client that it must connect to another server location. It contains the mandatory header Location, which indicates that the client should issue requests for that URL. It may contain the parameter Range, which indicates when the redirection takes effect. If the client wants to continue to send or receive media for this URI, the client MUST issue a TEARDOWN request for the current session and a SETUP for the new session at the designated host.
This example request redirects traffic for this URI to the new server at the given play time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 732
Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z-
10.11 RECORD
This method initiates recording a range of media data according to the presentation description. The timestamp reflects start and end time (UTC). If no time range is given, use the start or end time provided in the presentation description. If the session has already started, commence recording immediately.
The server decides whether to store the recorded data under the request-URI or another URI. If the server does not use the request- URI, the response SHOULD be 201 (Created) and contain an entity which describes the status of the request and refers to the new resource, and a Location header.
A media server supporting recording of live presentations MUST 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 rtsp://example.com/meeting/audio.en RTSP/1.0
CSeq: 954
Session: 12345678
Conference: 128.16.64.19/32492374
10.12 Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server to interleave RTSP methods and stream data. This interleaving should generally be avoided unless necessary since it complicates client and server operation and imposes additional overhead. Interleaved binary data SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar sign (24 hexadecimal), followed by a one-byte channel identifier, followed by the length of the encapsulated binary data as a binary, two-byte integer in network byte order. The stream data follows immediately afterwards, without a CRLF, but including the upper-layer protocol headers. Each $ block contains exactly one upper-layer protocol data unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header with the interleaved parameter(Section 12.39).
When the transport choice is RTP, RTCP messages are also interleaved by the server over the TCP connection. As a default, RTCP packets are sent on the first available channel higher than the RTP channel. The client MAY explicitly request RTCP packets on another channel. This is done by specifying two channels in the interleaved parameter of the Transport header(Section 12.39).
RTCP is needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a convenient way to tunnel RTP/RTCP packets through the TCP control connection when required by the network configuration and transfer them onto UDP when possible.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
CSeq: 2
Transport: RTP/AVP/TCP;interleaved=0-1
S->C: RTSP/1.0 200 OK
CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;interleaved=0-1
Session: 12345678
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
CSeq: 3
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 3
Session: 12345678
Date: 05 Jun 1997 18:59:15 GMT
RTP-Info: url=rtsp://foo.com/bar.file;
seq=232433;rtptime=972948234
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\001{2 byte length}{"length" bytes RTCP packet}
11 Status Code Definitions
Where applicable, HTTP status [H10] codes are reused. Status codes that have the same meaning are not repeated here. See Table 1 for a listing of which status codes may be returned by which requests.
11.1 Success 2xx
11.1.1 250 Low on Storage Space
The server returns this warning after receiving a RECORD request that it may not be able to fulfill completely due to insufficient storage space. If possible, the server should use the Range header to indicate what time period it may still be able to record. Since other processes on the server may be consuming storage space simultaneously, a client should take this only as an estimate.
11.2 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.
11.3 Client Error 4xx
11.3.1 405 Method Not Allowed
The method specified in the request is not allowed for the resource identified by the request URI. The response MUST include an Allow header containing a list of valid methods for the requested resource. This status code is also to be used if a request attempts to use a method not indicated during SETUP, e.g., if a RECORD request is issued even though the mode parameter in the Transport header only specified PLAY.
11.3.2 451 Parameter Not Understood
The recipient of the request does not support one or more parameters contained in the request.
11.3.3 452 Conference Not Found
The conference indicated by a Conference header field is unknown to the media server.
11.3.4 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth. This may, for example, be the result of a resource reservation failure.
11.3.5 454 Session Not Found
The RTSP session identifier in the Session header is missing, invalid, or has timed out.
11.3.6 455 Method Not Valid in This State
The client or server cannot process this request in its current state. The response SHOULD contain an Allow header to make error recovery easier.
11.3.7 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example, if PLAY contains the Range header field but the stream does not allow seeking.
11.3.8 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the presentation.
11.3.9 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not modified.
11.3.10 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URL in question since it is an aggregate (presentation) URL. The method may be applied on a stream URL.
11.3.11 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URL in question since it is not an aggregate (presentation) URL. The method may be applied on the presentation URL.
11.3.12 461 Unsupported Transport
The Transport field did not contain a supported transport specification.
11.3.13 462 Destination Unreachable
The data transmission channel could not be established because the client address could not be reached. This error will most likely be the result of a client attempt to place an invalid Destination parameter in the Transport field.
11.3.14 551 Option not supported
An option given in the Require or the Proxy-Require fields was not supported. The Unsupported header should be returned stating the option for which there is no support.
12 Header Field Definitions
HTTP/1.1 [2] or other, non-standard header fields not listed here currently have no well-defined meaning and SHOULD be ignored by the recipient.
Table 3 summarizes the header fields used by RTSP. Type "g" designates general request headers to be found in both requests and responses, type "R" designates request headers, type "r" designates response headers, and type "e" designates entity header fields. 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 "req." will be sent in every request of this type. The "req." means only that client (for response headers) and server (for request headers) MUST implement the fields. 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.
Header type support methods Accept R opt. entity Accept-Encoding R opt. entity Accept-Language R opt. all Allow r opt. all Authorization R opt. all Bandwidth R opt. all Blocksize R opt. all but OPTIONS, TEARDOWN Cache-Control g opt. SETUP Conference R opt. SETUP Connection g req. all Content-Base e opt. entity Content-Encoding e req. SET_PARAMETER Content-Encoding e req. DESCRIBE, ANNOUNCE Content-Language e req. DESCRIBE, ANNOUNCE Content-Length e req. SET_PARAMETER, ANNOUNCE Content-Length e req. entity Content-Location e opt. entity Content-Type e req. SET_PARAMETER, ANNOUNCE Content-Type r req. entity CSeq g req. all Date g opt. all Expires e opt. DESCRIBE, ANNOUNCE From R opt. all If-Modified-Since R opt. DESCRIBE, SETUP Last-Modified e opt. entity Proxy-Authenticate Proxy-Require R req. all 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 RTP-Info r req. PLAY Scale Rr opt. PLAY, RECORD Session Rr req. all but SETUP, OPTIONS Server r opt. all Speed Rr opt. PLAY Transport Rr req. SETUP Unsupported r req. all User-Agent R opt. all Via g opt. all WWW-Authenticate r opt. all
Overview of RTSP header fields
12.1 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
12.2 Accept-Encoding
See [H14.3]
12.3 Accept-Language
See [H14.4]. Note that the language specified applies to the presentation description and any reason phrases, not the media content.
12.4 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
12.5 Authorization
See [H14.8]
12.6 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. The bandwidth available to the client may change during an RTSP session, e.g., due to modem retraining.
Bandwidth = "Bandwidth" ":" 1*DIGIT
Example:
Bandwidth: 4000
12.7 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 override it with the media-specific size if necessary. The block size MUST be a positive decimal number, measured in octets. The server only returns an error (416) if the value is syntactically invalid.
12.8 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, except for responses to DESCRIBE.
Cache-Control = "Cache-Control" ":" 1#cache-directive cache-directive = cache-request-directive
| cache-response-directive
cache-request-directive = "no-cache"
| "max-stale"
| "min-fresh"
| "only-if-cached"
| cache-extension
cache-response-directive = "public"
| "private"
| "no-cache"
| "no-transform"
| "must-revalidate"
| "proxy-revalidate"
| "max-age" "=" delta-seconds
| cache-extension
cache-extension = token [ "=" ( token | quoted-string ) ]
no-cache:
Indicates that the media stream MUST NOT be cached anywhere.
This allows an origin server to prevent caching even by caches
that have been configured to return stale responses to client
requests.
public:
Indicates that the media stream is cacheable by any cache.
private:
Indicates that the media stream is intended for a single user
and MUST NOT be cached by a shared cache. A private (non-
shared) cache may cache the media stream.
no-transform:
An intermediate cache (proxy) may find it useful to convert
the media type of a certain stream. A proxy might, for
example, convert between video formats to save cache space or
to reduce the amount of traffic on a slow link. Serious
operational problems may occur, however, when these
transformations have been applied to streams intended for
certain kinds of applications. For example, applications for
medical imaging, scientific data analysis and those using
end-to-end authentication all depend on receiving a stream
that is bit-for-bit identical to the original entity-body.
Therefore, if a response includes the no-transform directive,
an intermediate cache or proxy MUST NOT change the encoding of
the stream. Unlike HTTP, RTSP does not provide for partial
transformation at this point, e.g., allowing translation into
a different language.
only-if-cached:
In some cases, such as times of extremely poor network
connectivity, a client may want a cache to return only those
media streams that it currently has stored, and not to receive
these from the origin server. To do this, the client may
include the only-if-cached directive in a request. If it
receives this directive, a cache SHOULD either respond using a
cached media stream that is consistent with the other
constraints of the request, or respond with a 504 (Gateway
Timeout) status. However, if a group of caches is being
operated as a unified system with good internal connectivity,
such a request MAY be forwarded within that group of caches.
max-stale:
Indicates that the client is willing to accept a media stream
that has exceeded its expiration time. If max-stale is
assigned a value, then the client is willing to accept a
response that has exceeded its expiration time by no more than
the specified number of seconds. If no value is assigned to
max-stale, then the client is willing to accept a stale
response of any age.
min-fresh:
Indicates that the client is willing to accept a media stream
whose freshness lifetime is no less than its current age plus
the specified time in seconds. That is, the client wants a
response that will still be fresh for at least the specified
number of seconds.
must-revalidate:
When the must-revalidate directive is present in a SETUP
response received by a cache, that cache MUST NOT use the
entry after it becomes stale to respond to a subsequent
request without first revalidating it with the origin server.
That is, the cache must do an end-to-end revalidation every
time, if, based solely on the origin server's Expires, the
cached response is stale.)
12.9 Conference
This request header field establishes a logical connection between a pre-established conference and an RTSP stream. The conference-id must not be changed for the same RTSP session.
Conference = "Conference" ":" conference-id Example:
Conference: [email protected]%20Starr
A response code of 452 (452 Conference Not Found) is returned if the conference-id is not valid.
12.10 Connection
See [H14.10]
12.11 Content-Base
See [H14.11]
12.12 Content-Encoding
See [H14.12]
12.13 Content-Language
See [H14.13]
12.14 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. If it is missing, a default value of zero is assumed. It is interpreted according to [H14.14].
12.15 Content-Location
See [H14.15]
12.16 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.
12.17 CSeq
The CSeq field specifies the sequence number for an RTSP request- response pair. This field MUST be present in all requests and responses. For every RTSP request containing the given sequence number, there will be a corresponding response having the same number. Any retransmitted request must contain the same sequence number as the original (i.e. the sequence number is not incremented for retransmissions of the same request).
12.18 Date
See [H14.19].
12.19 Expires
The Expires entity-header field gives a date and time after which the description or media-stream should be considered stale. The interpretation depends on the method:
DESCRIBE response:
The Expires header indicates a date and time after which the
description should be considered stale.
A stale cache entry may not normally be returned by a cache (either a proxy cache or an user agent cache) unless it is first validated with the origin server (or with an intermediate cache that has a fresh copy of the entity). See section 13 for further discussion of the expiration model.
The presence of an Expires field does not imply that the original resource will change or cease to exist at, before, or after that time.
The format is an absolute date and time as defined by HTTP-date in [H3.3]; it MUST be in RFC1123-date format:
Expires = "Expires" ":" HTTP-date
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats, especially including the value "0", as having occurred in the past (i.e., "already expired").
To mark a response as "already expired," an origin server should use an Expires date that is equal to the Date header value. To mark a response as "never expires," an origin server should use an Expires date approximately one year from the time the response is sent. RTSP/1.0 servers should not send Expires dates more than one year in the future.
The presence of an Expires header field with a date value of some time in the future on a media stream that otherwise would by default be non-cacheable indicates that the media stream is cacheable, unless indicated otherwise by a Cache-Control header field (Section 12.8).
12.20 From
See [H14.22].
12.21 Host
This HTTP request header field is not needed for RTSP. It should be silently ignored if sent.
12.22 If-Match
See [H14.25].
This field is especially useful for ensuring the integrity of the presentation description, in both the case where it is fetched via means external to RTSP (such as HTTP), or in the case where the server implementation is guaranteeing the integrity of the description between the time of the DESCRIBE message and the SETUP message.
The identifier is an opaque identifier, and thus is not specific to any particular session description language.
12.23 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE and SETUP methods to make them conditional. If the requested variant has not been modified since the time specified in this field, a description will not be returned from the server (DESCRIBE) or a stream will not be set up (SETUP). Instead, a 304 (not modified) response will be returned without any message-body.
If-Modified-Since = "If-Modified-Since" ":" HTTP-date
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
12.24 Last-Modified
The Last-Modified entity-header field indicates the date and time at which the origin server believes the presentation description or media stream was last modified. See [H14.29]. For the methods DESCRIBE or ANNOUNCE, the header field indicates the last modification date and time of the description, for SETUP that of the media stream.
12.25 Location
See [H14.30].
12.26 Proxy-Authenticate
See [H14.33].
12.27 Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive features that MUST be supported by the proxy. Any Proxy-Require header features that are not supported by the proxy MUST be negatively acknowledged by the proxy to the client if not supported. Servers
should treat this field identically to the Require field.
See Section 12.32 for more details on the mechanics of this message and a usage example.
12.28 Public
See [H14.35].
12.29 Range
This request and response header field specifies a range of time. The range can be specified in a number of units. This specification defines the smpte (Section 3.5), npt (Section 3.6), and clock (Section 3.7) range units. Within 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. Servers supporting the Range header MUST understand the NPT range format and SHOULD understand the SMPTE range format. The Range response header indicates what range of time is actually being played or recorded. If the Range header is given in a time format that is not understood, the recipient should return "501 Not Implemented".
Ranges are half-open intervals, including the lower point, but excluding the upper point. In other words, a range of a-b starts exactly at time a, but stops just before b. Only the start time of a media unit such as a video or audio frame is relevant. As an example, assume that video frames are generated every 40 ms. A range of 10.0- 10.1 would include a video frame starting at 10.0 or later time and would include a video frame starting at 10.08, even though it lasted beyond the interval. A range of 10.0-10.08, on the other hand, would exclude the frame at 10.08.
Range = "Range" ":" 1\#ranges-specifier
[ ";" "time" "=" utc-time ]
ranges-specifier = npt-range | utc-range | smpte-range
Example:
Range: clock=19960213T143205Z-;time=19970123T143720Z
The notation is similar to that used for the HTTP/1.1 [2] byte- range header. It allows clients to select an excerpt from the media object, and to play from a given point to the end as well as from the current location to a given point. The start of playback can be scheduled for any time in the future, although a server may refuse to keep server resources for extended idle periods.
12.30 Referer
See [H14.37]. The URL refers to that of the presentation description, typically retrieved via HTTP.
12.31 Retry-After
See [H14.38].
12.32 Require
The Require header is used by clients to query the server about options that it may or may not support. The server MUST respond to this header by using the Unsupported header to negatively acknowledge those options which are NOT supported.
This is to make sure that the client-server interaction will proceed without delay when all options are understood by both sides, and only slow down if options are not understood (as in the case above). For a well-matched client-server pair, the interaction proceeds quickly, saving a round-trip often required by negotiation mechanisms. In addition, it also removes state ambiguity when the client requires features that the server does not understand.
Require = "Require" ":" 1#option-tag
Example:
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/1.0 551 Option not supported
CSeq: 302
Unsupported: funky-feature
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 303
S->C: RTSP/1.0 200 OK
CSeq: 303
In this example, "funky-feature" is the feature tag which indicates to the client that the fictional Funky-Parameter field is required. The relationship between "funky-feature" and Funky-Parameter is not communicated via the RTSP exchange, since that relationship is an immutable property of "funky-feature" and thus should not be transmitted with every exchange.
Proxies and other intermediary devices SHOULD ignore features that are not understood in this field. If a particular extension requires that intermediate devices support it, the extension should be tagged in the Proxy-Require field instead (see Section 12.27).
12.33 RTP-Info
This field is used to set RTP-specific parameters in the PLAY response.
url:
Indicates the stream URL which for which the following RTP
parameters correspond.
seq:
Indicates the sequence number of the first packet of the
stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that
originated after the seek.
rtptime:
Indicates the RTP timestamp corresponding to the time value in
the Range response header. (Note: For aggregate control, a
particular stream may not actually generate a packet for the
Range time value returned or implied. Thus, there is no
guarantee that the packet with the sequence number indicated
by seq actually has the timestamp indicated by rtptime.) The
client uses this value to calculate the mapping of RTP time to
NPT.
A mapping from RTP timestamps to NTP timestamps (wall clock) is available via RTCP. However, this information is not sufficient to generate a mapping from RTP timestamps to NPT. Furthermore, in order to ensure that this information is available at the necessary time (immediately at startup or after a seek), and that it is delivered reliably, this mapping is placed in the RTSP control channel.
In order to compensate for drift for long, uninterrupted presentations, RTSP clients should additionally map NPT to NTP, using initial RTCP sender reports to do the mapping, and later reports to check drift against the mapping.
Syntax:
RTP-Info = "RTP-Info" ":" 1#stream-url 1*parameter stream-url = "url" "=" url parameter = ";" "seq" "=" 1*DIGIT
| ";" "rtptime" "=" 1*DIGIT
Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
12.34 Scale
A scale value of 1 indicates normal play or record at the normal 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
12.35 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. If the request contains a Range parameter, the new speed value will take effect at that time.
Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]
Example:
Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It is meant for use in specific circumstances where preview of the presentation at a higher or lower rate is necessary. Implementors should keep in mind that bandwidth for the session may be negotiated beforehand (by means other than RTSP), and therefore re-negotiation may be necessary. When data is delivered over UDP, it is highly recommended that means such as RTCP be used to track packet loss rates.
12.36 Server
See [H14.39]
12.37 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 (see Section 3.4). Once a client receives a Session identifier, it MUST return it for any request related to that session. A server does not have to set up a session identifier if it has other means of identifying a session, such as dynamically generated URLs.
Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
The timeout parameter is only allowed in a response header. The server uses it to indicate to the client how long the server is prepared to wait between RTSP commands before closing the session due to lack of activity (see Section A). The timeout is measured in
seconds, with a default of 60 seconds (1 minute).
Note that a session identifier identifies a RTSP session across transport sessions or connections. Control messages for more than one RTSP URL may be sent within a single RTSP session. Hence, it is possible that clients use the same session for controlling many streams constituting a presentation, as long as all the streams come from the same server. (See example in Section 14). However, multiple "user" sessions for the same URL from the same client MUST use different session identifiers.
The session identifier is needed to distinguish several delivery requests for the same URL coming from the same client.
The response 454 (Session Not Found) is returned if the session identifier is invalid.
12.38 Timestamp
The timestamp general header describes when the client sent the request to the server. The value of the timestamp is of significance only to the client and may use any timescale. The server MUST echo the exact same value and MAY, if it has accurate information about this, add a floating point number indicating the number of seconds that has elapsed since it has received the request. The timestamp is used by the client to compute the round-trip time to the server so that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ] delay = *(DIGIT) [ "." *(DIGIT) ]
12.39 Transport
This request header indicates which transport protocol is to be used and configures its parameters such as destination address, compression, multicast time-to-live and destination port for a single stream. It sets those values not already determined by a presentation description.
Transports are comma separated, listed in order of preference. Parameters may be added to each transport, separated by a semicolon.
The Transport header 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 options acceptable to the client. In that case, the server MUST return a single option which was actually chosen.
The syntax for the transport specifier is
transport/profile/lower-transport.
The default value for the "lower-transport" parameters is specific to the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport:
General parameters:
unicast | multicast:
mutually exclusive indication of whether unicast or multicast
delivery will be attempted. Default value is multicast.
Clients that are capable of handling both unicast and
multicast transmission MUST indicate such capability by
including two full transport-specs with separate parameters
for each.
destination:
The address to which a stream will be sent. The client may
specify the multicast address with the destination parameter.
To avoid becoming the unwitting perpetrator of a remote-
controlled denial-of-service attack, a server SHOULD
authenticate the client and SHOULD log such attempts before
allowing the client to direct a media stream to an address not
chosen by the server. This is particularly important if RTSP
commands are issued via UDP, but implementations cannot rely
on TCP as reliable means of client identification by itself. A
server SHOULD not allow a client to direct media streams to an
address that differs from the address commands are coming
from.
source:
If the source address for the stream is different than can be
derived from the RTSP endpoint address (the server in playback
or the client in recording), the source MAY be specified.
This information may also be available through SDP. However, since this is more a feature of transport than media initialization, the authoritative source for this information should be in the SETUP response.
layers:
The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses starting
at the destination address.
mode:
The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY.
append:
If the mode parameter includes RECORD, the append parameter
indicates that the media data should append to the existing
resource rather than overwrite it. If appending is requested
and the server does not support this, it MUST refuse the
request rather than overwrite the resource identified by the
URI. The append parameter is ignored if the mode parameter
does not contain RECORD.
interleaved:
The interleaved parameter implies mixing the media stream with
the control stream in whatever protocol is being used by the
control stream, using the mechanism defined in Section 10.12.
The argument provides the channel number to be used in the $
statement. This parameter may be specified as a range, e.g.,
interleaved=4-5 in cases where the transport choice for the
media stream requires it.
This allows RTP/RTCP to be handled similarly to the way that it is done with UDP, i.e., one channel for RTP and the other for RTCP.
Multicast specific:
ttl:
multicast time-to-live
RTP Specific:
port:
This parameter provides the RTP/RTCP port pair for a multicast
session. It is specified as a range, e.g., port=3456-3457.
client_port:
This parameter provides the unicast RTP/RTCP port pair on
which the client has chosen to receive media data and control
information. It is specified as a range, e.g.,
client_port=3456-3457.
server_port:
This parameter provides the unicast RTP/RTCP port pair on
which the server has chosen to receive media data and control
information. It is specified as a range, e.g.,
server_port=3456-3457.
ssrc:
The ssrc parameter indicates the RTP SSRC [24, Sec. 3] 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.
Transport = "Transport" ":"
1\#transport-spec
transport-spec = transport-protocol/profile[/lower-transport]
*parameter
transport-protocol = "RTP" profile = "AVP" lower-transport = "TCP" | "UDP" parameter = ( "unicast" | "multicast" )
| ";" "destination" [ "=" address ]
| ";" "interleaved" "=" channel [ "-" channel ]
| ";" "append"
| ";" "ttl" "=" ttl
| ";" "layers" "=" 1*DIGIT
| ";" "port" "=" port [ "-" port ]
| ";" "client_port" "=" port [ "-" port ]
| ";" "server_port" "=" port [ "-" port ]
| ";" "ssrc" "=" ssrc
| ";" "mode" = <"> 1\#mode <">
ttl = 1*3(DIGIT) port = 1*5(DIGIT) ssrc = 8*8(HEX) channel = 1*3(DIGIT) address = host mode = <"> *Method <"> | Method
Example:
Transport: RTP/AVP;multicast;ttl=127;mode="PLAY",
RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"
The Transport header is restricted to describing a single RTP stream. (RTSP can also control multiple streams as a single entity.) Making it part of RTSP rather than relying on a multitude of session description formats greatly simplifies designs of firewalls.
12.40 Unsupported
The Unsupported response header lists the features not supported by the server. In the case where the feature was specified via the Proxy-Require field (Section 12.32), if there is a proxy on the path between the client and the server, the proxy MUST insert a message reply with an error message "551 Option Not Supported".
See Section 12.32 for a usage example.
12.41 User-Agent
See [H14.42]
12.42 Vary
See [H14.43]
12.43 Via
See [H14.44].
12.44 WWW-Authentica
See [H14.46].
13 Caching
In HTTP, response-request pairs are cached. RTSP differs significantly in that respect. Responses are not cacheable, with the exception of the presentation description returned by DESCRIBE or included with ANNOUNCE. (Since the responses for anything but DESCRIBE and GET_PARAMETER do not return any data, caching is not really an issue for these requests.) However, it is desirable for the continuous media data, typically delivered out-of-band with respect to RTSP, to be cached, as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it has an up-to-date copy of the continuous media content and its description. It can determine whether the copy is up-to-date by issuing a SETUP or DESCRIBE request, respectively, and comparing the Last-Modified header with that of the cached copy. If the copy is not up-to-date, it modifies the SETUP transport parameters as appropriate and forwards the request to the origin server. Subsequent control commands such as PLAY or PAUSE then pass the proxy unmodified. The proxy delivers the continuous media data to the client, while possibly making a local copy for later reuse. The exact behavior allowed to the cache is given by the cache-response directives
described in Section 12.8. 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 appears like a regular media server, to the media origin server like a client. Just as an HTTP cache has to store the content type, content language, and so on for the objects it caches, a media cache has to store the presentation description. Typically, a cache eliminates all transport-references (that is, multicast information) from the presentation description, since these are independent of the data delivery from the cache to the client. Information on the encodings remains the same. If the cache is able to translate the cached media data, it would create a new presentation description with all the encoding possibilities it can offer.
14 Examples
The following examples refer to stream description formats that are not standards, such as RTSL. The following examples are not to be used as a reference for those formats.
14.1 Media on Demand (Unicast)
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 timeline of the movie.
In this example, the client is only interested in the last part of the movie.
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.0 200 OK
Content-Type: application/sdp
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
A->C: RTSP/1.0 200 OK
CSeq: 1
Session: 12345678
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
server_port=5000-5001
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
V->C: RTSP/1.0 200 OK
CSeq: 1
Session: 23456789
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
server_port=5002-5003
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-
V->C: RTSP/1.0 200 OK
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://video.example.com/twister/video;
seq=12312232;rtptime=78712811
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/1.0 200 OK
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
seq=876655;rtptime=1032181
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 3
Session: 12345678
A->C: RTSP/1.0 200 OK
CSeq: 3
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 3
Session: 23456789
V->C: RTSP/1.0 200 OK
CSeq: 3
Even though the audio and video track are on two different servers, and may start at slightly different times and may drift with respect to each other, the client can synchronize the two using standard RTP methods, in particular the time scale contained in the RTCP sender reports.
14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in which multiple continuous media types pertaining to the same end-user presentation are present. In effect, the container file represents an RTSP presentation, with each of its components being RTSP streams. Container files are a widely used means to store such presentations. While the components are transported as independent streams, it is desirable to maintain a common context for those streams at the server end.
This enables the server to keep a single storage handle open easily. It also allows treating all the streams equally in case of any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent selective retrieval of the streams by the client in order to preserve the artistic effect of the combined media presentation. Similarly, in such a tightly bound presentation, it is desirable to be able to control all the streams via a single control message using an aggregate URL.
The following is an example of using a single RTSP session to control multiple streams. It also illustrates the use of aggregate URLs.
Client C requests a presentation from media server M . The movie is stored in a container file. The client has obtained an RTSP URL to the container file.
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 164
v=0
o=- 2890844256 2890842807 IN IP4 172.16.2.93
s=RTSP Session
i=An Example of RTSP Session Usage
a=control:rtsp://foo/twister
t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://foo/twister/audio
m=video 0 RTP/AVP 26
a=control:rtsp://foo/twister/video
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001;
server_port=9000-9001
Session: 12345678
C->M: SETUP rtsp://foo/twister/video RTSP/1.0
CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003;
server_port=9004-9005
Session: 12345678
C->M: PLAY rtsp://foo/twister RTSP/1.0
CSeq: 4
Range: npt=0-
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 4
Session: 12345678
RTP-Info: url=rtsp://foo/twister/video;
seq=9810092;rtptime=3450012
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
CSeq: 5
Session: 12345678
M->C: RTSP/1.0 460 Only aggregate operation allowed
CSeq: 5
C->M: PAUSE rtsp://foo/twister RTSP/1.0
CSeq: 6
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 6
Session: 12345678
C->M: SETUP rtsp://foo/twister RTSP/1.0
CSeq: 7
Transport: RTP/AVP;unicast;client_port=10000
M->C: RTSP/1.0 459 Aggregate operation not allowed
CSeq: 7
In the first instance of failure, the client tries to pause one stream (in this case video) of the presentation. This is disallowed for that presentation by the server. In the second instance, the aggregate URL may not be used for SETUP and one control message is required per stream to set up transport parameters.
This keeps the syntax of the Transport header simple and allows easy parsing of transport information by firewalls.
14.3 Single Stream Container Files
Some RTSP servers may treat all files as though they are "container files", yet other servers may not support such a concept. Because of this, clients SHOULD use the rules set forth in the session description for request URLs, rather than assuming that a consistent URL may always be used throughout. Here's an example of how a multi- stream server might expect a single-stream file to be served:
Accept: application/x-rtsp-mh, application/sdp
CSeq: 1
S->C RTSP/1.0 200 OK
CSeq: 1
Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp
Content-length: 48
v=0
o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file
i=audio test
t=0 0
m=audio 0 RTP/AVP 0
a=control:streamid=0
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast;
client_port=6970-6971;mode=play
CSeq: 2
S->C RTSP/1.0 200 OK
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
server_port=6970-6971;mode=play
CSeq: 2
Session: 2034820394
C->S PLAY rtsp://foo.com/test.wav RTSP/1.0
CSeq: 3
Session: 2034820394
S->C RTSP/1.0 200 OK
CSeq: 3
Session: 2034820394
RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
seq=981888;rtptime=3781123
Note the different URL in the SETUP command, and then the switch back to the aggregate URL in the PLAY command. This makes complete sense when there are multiple streams with aggregate control, but is less than intuitive in the special case where the number of streams is one.
In this special case, it is recommended that servers be forgiving of implementations that send:
C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
CSeq: 3
In the worst case, servers should send back:
S->C RTSP/1.0 460 Only aggregate operation allowed
CSeq: 3
One would also hope that server implementations are also forgiving of the following:
C->S SETUP rtsp://foo.com/test.wav RTSP/1.0
Transport: rtp/avp/udp;client_port=6970-6971;mode=play
CSeq: 2
Since there is only a single stream in this file, it's not ambiguous what this means.
14.4 Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, we assume that the web server only contains a pointer to the full description, while the media server M maintains the full description.
C->W: GET /concert.sdp HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: application/x-rtsl
<session>
<track src="rtsp://live.example.com/concert/audio">
</session>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 44
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 3456 RTP/AVP 0
a=control:rtsp://live.example.com/concert/audio
c=IN IP4 224.2.0.1/16
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 2
Transport: RTP/AVP;multicast
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=224.2.0.1;
port=3456-3457;ttl=16
Session: 0456804596
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 3
Session: 0456804596
M->C: RTSP/1.0 200 OK
CSeq: 3
Session: 0456804596
14.5 Playing media into an existing session
A conference participant C wants to have the media server M play back a demo tape into an existing conference. 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: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 1
Accept: application/sdp
M->C: RTSP/1.0 200 1 OK
Content-type: application/sdp
Content-Length: 44
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
i=See above
t=0 0
m=audio 0 RTP/AVP 0
C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
port=7000-7001;ttl=127
Conference: [email protected]%20Starr
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
port=7000-7001;ttl=127
Session: 91389234234
Conference: [email protected]%20Starr
C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 3
Session: 91389234234
M->C: RTSP/1.0 200 OK
CSeq: 3
14.6 Recording
The conference participant client C asks the media server M to record the audio and video portions of a meeting. The client uses the ANNOUNCE method to provide meta-information about the recorded session to the server.
C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
CSeq: 90
Content-Type: application/sdp
Content-Length: 121
v=0
o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
s=IETF Meeting, Munich - 1
i=The thirty-ninth IETF meeting will be held in Munich, Germany
u=http://www.ietf.org/meetings/Munich.html
e=IETF Channel 1 <[email protected]>
p=IETF Channel 1 +49-172-2312 451
c=IN IP4 224.0.1.11/127
t=3080271600 3080703600
a=tool:sdr v2.4a6
a=type:test
m=audio 21010 RTP/AVP 5
c=IN IP4 224.0.1.11/127
a=ptime:40
m=video 61010 RTP/AVP 31
c=IN IP4 224.0.1.12/127
M->C: RTSP/1.0 200 OK
CSeq: 90
C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
CSeq: 91
Transport: RTP/AVP;multicast;destination=224.0.1.11;
port=21010-21011;mode=record;ttl=127
M->C: RTSP/1.0 200 OK
CSeq: 91
Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.11;
port=21010-21011;mode=record;ttl=127
C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
CSeq: 92
Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127
M->C: RTSP/1.0 200 OK
CSeq: 92
Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
CSeq: 93
Session: 50887676
Range: clock=19961110T1925-19961110T2015
M->C: RTSP/1.0 200 OK
CSeq: 93
15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF) as used in RFC 2068 [2].
15.1 Base Syntax
OCTET = <any 8-bit sequence of data> CHAR = <any US-ASCII character (octets 0 - 127)> UPALPHA = <any US-ASCII uppercase letter "A".."Z"> LOALPHA = <any US-ASCII lowercase letter "a".."z"> ALPHA = UPALPHA | LOALPHA
DIGIT = <any US-ASCII digit "0".."9"> CTL = <any US-ASCII control character
(octets 0 - 31) and DEL (127)>
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 = "\" CHAR
message-header = field-name ":" [ field-value ] CRLF field-name = token field-value = *( field-content | LWS ) field-content = <the OCTETs making up the field-value and
consisting of either *TEXT or
combinations of token, tspecials, and
quoted-string>
safe = "\$" | "-" | "_" | "." | "+" extra = "!" | "*" | "$'$" | "(" | ")" | ","
hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
"a" | "b" | "c" | "d" | "e" | "f"
escape = "\%" hex hex reserved = ";" | "/" | "?" | ":" | "@" | "&" | "="
unreserved = alpha | digit | safe | extra xchar = unreserved | reserved | escape
16 Security Considerations
Because of the similarity in syntax and usage between RTSP servers and HTTP servers, the security considerations outlined in [H15] apply. Specifically, please note the following:
Authentication Mechanisms:
RTSP and HTTP share common authentication schemes, and thus
should follow the same prescriptions with regards to
authentication. See [H15.1] for client authentication issues,
and [H15.2] for issues regarding support for multiple
authentication mechanisms.
Abuse of Server Log Information:
RTSP and HTTP servers will presumably have similar logging
mechanisms, and thus should be equally guarded in protecting
the contents of those logs, thus protecting the privacy of the
users of the servers. See [H15.3] for HTTP server
recommendations regarding server logs.
Transfer of Sensitive Information:
There is no reason to believe that information transferred via
RTSP may be any less sensitive than that normally transmitted
via HTTP. Therefore, all of the precautions regarding the
protection of data privacy and user privacy apply to
implementors of RTSP clients, servers, and proxies. See
[H15.4] for further details.
Attacks Based On File and Path Names:
Though RTSP URLs are opaque handles that do not necessarily
have file system semantics, it is anticipated that many
implementations will translate portions of the request URLs
directly to file system calls. In such cases, file systems
SHOULD follow the precautions outlined in [H15.5], such as
checking for ".." in path components.
Personal Information:
RTSP clients are often privy to the same information that HTTP
clients are (user name, location, etc.) and thus should be
equally. See [H15.6] for further recommendations.
Privacy Issues Connected to Accept Headers:
Since may of the same "Accept" headers exist in RTSP as in
HTTP, the same caveats outlined in [H15.7] with regards to
their use should be followed.
DNS Spoofing:
Presumably, given the longer connection times typically
associated to RTSP sessions relative to HTTP sessions, RTSP
client DNS optimizations should be less prevalent.
Nonetheless, the recommendations provided in [H15.8] are still
relevant to any implementation which attempts to rely on a
DNS-to-IP mapping to hold beyond a single use of the mapping.
Location Headers and Spoofing:
If a single server supports multiple organizations that do not
trust one another, then it must check the values of Location
and Content-Location headers in responses that are generated
under control of said organizations to make sure that they do
not attempt to invalidate resources over which they have no
authority. ([H15.9])
In addition to the recommendations in the current HTTP specification (RFC 2068 [2], as of this writing), future HTTP specifications may provide additional guidance on security issues.
The following are added considerations for RTSP implementations.
Concentrated denial-of-service attack:
The protocol offers the opportunity for a remote-controlled
denial-of-service attack. The attacker may initiate traffic
flows to one or more IP addresses by specifying them as the
destination in SETUP requests. While the attacker's IP address
may be known in this case, this is not always useful in
prevention of more attacks or ascertaining the attackers
identity. Thus, an RTSP server SHOULD only allow client-
specified destinations for RTSP-initiated traffic flows if the
server has verified the client's identity, either against a
database of known users using RTSP authentication mechanisms
(preferably digest authentication or stronger), or other
secure means.
Session hijacking:
Since there is no relation between a transport layer
connection and an RTSP session, it is possible for a malicious
client to issue requests with random session identifiers which
would affect unsuspecting clients. The server SHOULD use a
large, random and non-sequential session identifier to
minimize the possibility of this kind of attack.
Authentication:
Servers SHOULD implement both basic and digest [8]
authentication. In environments requiring tighter security for
the control messages, the RTSP control stream may be
encrypted.
Stream issues:
RTSP only provides for stream control. Stream delivery issues
are not covered in this section, nor in the rest of this memo.
RTSP implementations will most likely rely on other protocols
such as RTP, IP multicast, RSVP and IGMP, and should address
security considerations brought up in those and other
applicable specifications.
Persistently suspicious behavior:
RTSP servers SHOULD return error code 403 (Forbidden) upon
receiving a single instance of behavior which is deemed a
security risk. RTSP servers SHOULD also be aware of attempts
to probe the server for weaknesses and entry points and MAY
arbitrarily disconnect and ignore further requests clients
which are deemed to be in violation of local security policy.
Appendix 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.
State is defined on a per object basis. An object is uniquely identified by the stream URL and the RTSP session identifier. Any request/reply using aggregate URLs denoting RTSP presentations composed of multiple streams will have an effect on the individual states of all the streams. For example, if the presentation /movie contains two streams, /movie/audio and /movie/video, then the following command:
PLAY rtsp://foo.com/movie RTSP/1.0 CSeq: 559 Session: 12345678
will have an effect on the states of movie/audio and movie/video.
This example does not imply a standard way to represent streams in URLs or a relation to the filesystem. See Section 3.2.
The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER, SET_PARAMETER do not have any effect on client or server state and are therefore not listed in the state tables.
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 PAUSE reply received while in Playing
state.
Playing:
PLAY reply received
Recording:
RECORD reply received
In general, the client changes state on receipt of replies to requests. Note that some requests are effective at a future time or position (such as a PAUSE), and state also changes accordingly. 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 client also changes state from Playing/Recording to Ready when the end of the requested range is reached.
The "next state" column indicates the state assumed after receiving a success response (2xx). If a request yields a status code of 3xx, the state becomes Init, and a status code of 4xx yields no change in state. 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.
state message sent next state after response Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
RECORD Recording
TEARDOWN Init
SETUP Ready
Playing PAUSE Ready
TEARDOWN Init
PLAY Playing
SETUP Playing (changed transport)
Recording PAUSE Ready
TEARDOWN Init
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 has been received yet.
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.
In general, 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 or RTSP commands, from the client for a defined interval, with a default of one minute. The server can declare another timeout value in the Session response header (Section 12.37). 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. Note that some requests (such as PAUSE) may be effective at a future time or position, and server state changes at the appropriate time. The server reverts from state Playing or Recording to state Ready at the end of the range requested by the client.
The REDIRECT message, when sent, is effective immediately unless it has a Range header specifying when the redirect is effective. In such a case, server state will also change at the appropriate time.
If no explicit SETUP is required for the object, the state starts at Ready and there are only two states, Ready and Playing.
The "next state" column indicates the state assumed after sending a success response (2xx). If a request results in a status code of 3xx, the state becomes Init. A status code of 4xx results in no change.
state message received next state
Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
SETUP Ready
TEARDOWN Init
RECORD Recording
Playing PLAY Playing
PAUSE Ready
TEARDOWN Init
SETUP Playing
Recording RECORD Recording
PAUSE Ready
TEARDOWN Init
SETUP Recording
Appendix B: Interaction with RTP
RTSP allows media clients to control selected, non-contiguous sections of media presentations, rendering those streams with an RTP media layer[24]. The media layer rendering the RTP stream should not be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be continuous and monotonic across jumps of NPT.
As an example, assume a clock frequency of 8000 Hz, a packetization interval of 100 ms and an initial sequence number and timestamp of zero. First we play NPT 10 through 15, then skip ahead and play NPT 18 through 20. The first segment is presented as RTP packets with sequence numbers 0 through 49 and timestamp 0 through 39,200. The second segment consists of RTP packets with sequence number 50 through 69, with timestamps 40,000 through 55,200.
We cannot assume that the RTSP client can communicate with the RTP media agent, as the two may be independent processes. If the RTP timestamp shows the same gap as the NPT, the media agent will assume that there is a pause in the presentation. If the jump in NPT is large enough, the RTP timestamp may roll over and the media agent may believe later packets to be duplicates of packets just played out.
For certain datatypes, tight integration between the RTSP layer and the RTP layer will be necessary. This by no means precludes the above restriction. Combined RTSP/RTP media clients should use the RTP-Info field to determine whether incoming RTP packets were sent before or after a seek.
For continuous audio, the server SHOULD set the RTP marker bit at the beginning of serving a new PLAY request. This allows the client to perform playout delay adaptation.
For scaling (see Section 12.34), RTP timestamps should correspond to the playback timing. For example, when playing video recorded at 30 frames/second at a scale of two and speed (Section 12.35) of one, the server would drop every second frame to maintain and deliver video packets with the normal timestamp spacing of 3,000 per frame, but NPT would increase by 1/15 second for each video frame.
The client can maintain a correct display of NPT by noting the RTP timestamp value of the first packet arriving after repositioning. The sequence parameter of the RTP-Info (Section 12.33) header provides the first sequence number of the next segment.
Appendix C: Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, RFC 2327 [6]) may be used to describe streams or presentations in RTSP. Such usage is limited to specifying means of access and encoding(s) for:
aggregate control:
A presentation composed of streams from one or more servers
that are not available for aggregate control. Such a
description is typically retrieved by HTTP or other non-RTSP
means. However, they may be received with ANNOUNCE methods.
non-aggregate control:
A presentation composed of multiple streams from a single
server that are available for aggregate control. Such a
description is typically returned in reply to a DESCRIBE
request on a URL, or received in an ANNOUNCE method.
This appendix describes how an SDP file, retrieved, for example, through HTTP, determines the operation of an RTSP session. It also describes how a client should interpret SDP content returned in reply to a DESCRIBE request. SDP provides no mechanism by which a client can distinguish, without human guidance, between several media streams to be rendered simultaneously and a set of alternatives (e.g., two audio streams spoken in different languages).
C.1 Definitions
The terms "session-level", "media-level" and other key/attribute names and values used in this appendix are to be used as defined in SDP (RFC 2327 [6]):
C.1.1 Control URL
The "a=control:" attribute is used to convey the control URL. This attribute is used both for the session and media descriptions. If used for individual media, it indicates the URL to be used for controlling that particular media stream. If found at the session level, the attribute indicates the URL for aggregate control.
Example:
a=control:rtsp://example.com/foo
This attribute may contain either relative and absolute URLs, following the rules and conventions set out in RFC 1808 [25]. Implementations should look for a base URL in the following order:
1. The RTSP Content-Base field 2. The RTSP Content-Location field 3. The RTSP request URL
If this attribute contains only an asterisk (*), then the URL is treated as if it were an empty embedded URL, and thus inherits the entire base URL.
C.1.2 Media streams
The "m=" field is used to enumerate the streams. It is expected that all the specified streams will be rendered with appropriate synchronization. If the session is unicast, the port number serves as a recommendation from the server to the client; the client still has to include it in its SETUP request and may ignore this recommendation. If the server has no preference, it SHOULD set the port number value to zero.
Example:
m=audio 0 RTP/AVP 31
C.1.3 Payload type(s)
The payload type(s) are specified in the "m=" field. In case the payload type is a static payload type from RFC 1890 [1], no other information is required. In case it is a dynamic payload type, the media attribute "rtpmap" is used to specify what the media is. The "encoding name" within the "rtpmap" attribute may be one of those specified in RFC 1890 (Sections 5 and 6), or an experimental encoding with a "X-" prefix as specified in SDP (RFC 2327 [6]). Codec- specific parameters are not specified in this field, but rather in the "fmtp" attribute described below. Implementors seeking to register new encodings should follow the procedure in RFC 1890 [1]. If the media type is not suited to the RTP AV profile, then it is recommended that a new profile be created and the appropriate profile name be used in lieu of "RTP/AVP" in the "m=" field.
C.1.4 Format-specific parameters
Format-specific parameters are conveyed using the "fmtp" media attribute. The syntax of the "fmtp" attribute is specific to the encoding(s) that the attribute refers to. Note that the packetization interval is conveyed using the "ptime" attribute.
C.1.5 Range of presentation
The "a=range" attribute defines the total time range of the stored session. (The length of live sessions can be deduced from the "t" and "r" parameters.) Unless the presentation contains media streams of different durations, the range attribute is a session-level attribute. The unit is specified first, followed by the value range. The units and their values are as defined in Section 3.5, 3.6 and 3.7.
Examples:
a=range:npt=0-34.4368 a=range:clock=19971113T2115-19971113T2203
C.1.6 Time of availability
The "t=" field MUST contain suitable values for the start and stop times for both aggregate and non-aggregate stream control. With aggregate control, the server SHOULD indicate a stop time value for which it guarantees the description to be valid, and a start time that is equal to or before the time at which the DESCRIBE request was received. It MAY also indicate start and stop times of 0, meaning that the session is always available. With non-aggregate control, the values should reflect the actual period for which the session is available in keeping with SDP semantics, and not depend on other means (such as the life of the web page containing the description) for this purpose.
C.1.7 Connection Information
In SDP, the "c=" field contains the destination address for the media stream. However, for on-demand unicast streams and some multicast streams, the destination address is specified by the client via the SETUP request. Unless the media content has a fixed destination address, the "c=" field is to be set to a suitable null value. For addresses of type "IP4", this value is "0.0.0.0".
C.1.8 Entity Tag
The optional "a=etag" attribute identifies a version of the session description. It is opaque to the client. SETUP requests may include this identifier in the If-Match field (see section 12.22) to only allow session establishment if this attribute value still corresponds to that of the current description. The attribute value is opaque and may contain any character allowed within SDP attribute values.
Example:
a=etag:158bb3e7c7fd62ce67f12b533f06b83a
One could argue that the "o=" field provides identical functionality. However, it does so in a manner that would put constraints on servers that need to support multiple session description types other than SDP for the same piece of media content.
C.2 Aggregate Control Not Available
If a presentation does not support aggregate control and multiple media sections are specified, each section MUST have the control URL specified via the "a=control:" attribute.
Example:
v=0 o=- 2890844256 2890842807 IN IP4 204.34.34.32 s=I came from a web page t=0 0 c=IN IP4 0.0.0.0 m=video 8002 RTP/AVP 31 a=control:rtsp://audio.com/movie.aud m=audio 8004 RTP/AVP 3 a=control:rtsp://video.com/movie.vid
Note that the position of the control URL in the description implies that the client establishes separate RTSP control sessions to the servers audio.com and video.com.
It is recommended that an SDP file contains the complete media initialization information even if it is delivered to the media client through non-RTSP means. This is necessary as there is no mechanism to indicate that the client should request more detailed media stream information via DESCRIBE.
C.3 Aggregate Control Available
In this scenario, the server has multiple streams that can be controlled as a whole. In this case, there are both media-level "a=control:" attributes, which are used to specify the stream URLs, and a session-level "a=control:" attribute which is used as the request URL for aggregate control. If the media-level URL is relative, it is resolved to absolute URLs according to Section C.1.1 above.
If the presentation comprises only a single stream, the media-level "a=control:" attribute may be omitted altogether. However, if the presentation contains more than one stream, each media stream section MUST contain its own "a=control" attribute.
Example:
v=0 o=- 2890844256 2890842807 IN IP4 204.34.34.32 s=I contain i=<more info> t=0 0 c=IN IP4 0.0.0.0 a=control:rtsp://example.com/movie/ m=video 8002 RTP/AVP 31 a=control:trackID=1 m=audio 8004 RTP/AVP 3 a=control:trackID=2
In this example, the client is required to establish a single RTSP session to the server, and uses the URLs rtsp://example.com/movie/trackID=1 and rtsp://example.com/movie/trackID=2 to set up the video and audio streams, respectively. The URL rtsp://example.com/movie/ controls the whole movie.
Appendix D: Minimal RTSP implementation
D.1 Client
A client implementation MUST be able to do the following :
* Generate the following requests: SETUP, TEARDOWN, and one of PLAY (i.e., a minimal playback client) or RECORD (i.e., a minimal recording client). If RECORD is implemented, ANNOUNCE must be implemented as well. * Include the following headers in requests: CSeq, Connection, Session, Transport. If ANNOUNCE is implemented, the capability to include headers Content-Language, Content-Encoding, Content- Length, and Content-Type should be as well. * Parse and understand the following headers in responses: CSeq, Connection, Session, Transport, Content-Language, Content- Encoding, Content-Length, Content-Type. If RECORD is implemented, the Location header must be understood as well. RTP-compliant implementations should also implement RTP-Info. * Understand the class of each error code received and notify the end-user, if one is present, of error codes in classes 4xx and 5xx. The notification requirement may be relaxed if the end-user explicitly does not want it for one or all status codes. * Expect and respond to asynchronous requests from the server, such as ANNOUNCE. This does not necessarily mean that it should implement the ANNOUNCE method, merely that it MUST respond positively or negatively to any request received from the server.
Though not required, the following are highly recommended at the time of publication for practical interoperability with initial implementations and/or to be a "good citizen".
* Implement RTP/AVP/UDP as a valid transport. * Inclusion of the User-Agent header. * Understand SDP session descriptions as defined in Appendix C * Accept media initialization formats (such as SDP) from standard input, command line, or other means appropriate to the operating environment to act as a "helper application" for other applications (such as web browsers).
There may be RTSP applications different from those initially envisioned by the contributors to the RTSP specification for which the requirements above do not make sense. Therefore, the recommendations above serve only as guidelines instead of strict requirements.
D.1.1 Basic Playback
To support on-demand playback of media streams, the client MUST additionally be able to do the following:
* generate the PAUSE request; * implement the REDIRECT method, and the Location header.
D.1.2 Authentication-enabled
In order to access media presentations from RTSP servers that require authentication, the client MUST additionally be able to do the following:
* recognize the 401 status code; * parse and include the WWW-Authenticate header; * implement Basic Authentication and Digest Authentication.
D.2 Server
A minimal server implementation MUST be able to do the following:
* Implement the following methods: SETUP, TEARDOWN, OPTIONS and either PLAY (for a minimal playback server) or RECORD (for a minimal recording server). If RECORD is implemented, ANNOUNCE should be implemented as well. * Include the following headers in responses: Connection, Content-Length, Content-Type, Content-Language, Content-Encoding, Transport, Public. The capability to include the Location header should be implemented if the RECORD method is. RTP-compliant implementations should also implement the RTP-Info field. * Parse and respond appropriately to the following headers in requests: Connection, Session, Transport, Require.
Though not required, the following are highly recommended at the time of publication for practical interoperability with initial implementations and/or to be a "good citizen".
* Implement RTP/AVP/UDP as a valid transport. * Inclusion of the Server header. * Implement the DESCRIBE method. * Generate SDP session descriptions as defined in Appendix C
There may be RTSP applications different from those initially envisioned by the contributors to the RTSP specification for which the requirements above do not make sense. Therefore, the recommendations above serve only as guidelines instead of strict requirements.
D.2.1 Basic Playback
To support on-demand playback of media streams, the server MUST additionally be able to do the following:
* Recognize the Range header, and return an error if seeking is not supported. * Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface features, the following are highly recommended for on-demand media servers:
* Include and parse the Range header, with NPT units. Implementation of SMPTE units is recommended. * Include the length of the media presentation in the media initialization information. * Include mappings from data-specific timestamps to NPT. When RTP is used, the rtptime portion of the RTP-Info field may be used to map RTP timestamps to NPT.
Client implementations may use the presence of length information to determine if the clip is seekable, and visibly disable seeking features for clips for which the length information is unavailable. A common use of the presentation length is to implement a "slider bar" which serves as both a progress indicator and a timeline positioning tool.
Mappings from RTP timestamps to NPT are necessary to ensure correct positioning of the slider bar.
D.2.2 Authentication-enabled
In order to correctly handle client authentication, the server MUST additionally be able to do the following:
* Generate the 401 status code when authentication is required for the resource. * Parse and include the WWW-Authenticate header * Implement Basic Authentication and Digest Authentication
Appendix E: Authors' Addresses
Henning Schulzrinne Dept. of Computer Science Columbia University 1214 Amsterdam Avenue New York, NY 10027 USA
EMail: [email protected]
Anup Rao Netscape Communications Corp. 501 E. Middlefield Road Mountain View, CA 94043 USA
EMail: [email protected]
Robert Lanphier RealNetworks 1111 Third Avenue Suite 2900 Seattle, WA 98101 USA
EMail: [email protected]
Appendix F: Acknowledgements
This memo is based on the functionality of the original RTSP document submitted in October 96. 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, Torsten Braun, Brent Browning, Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka, Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Papadopouli, Sujal Patel, Ema Patki, Alagu Periyannan, Igor Plotnikov, Pinaki Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and John Francis Stracke.
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19 Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform
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20 Yergeau, F., "UTF-8, a transformation format of ISO 10646",
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