7
Network Working Group H. Schulzrinne
8
Request for Comments: 2326 Columbia U.
9
Category: Standards Track A. Rao
15
Real Time Streaming Protocol (RTSP)
19
This document specifies an Internet standards track protocol for the
20
Internet community, and requests discussion and suggestions for
21
improvements. Please refer to the current edition of the "Internet
22
Official Protocol Standards" (STD 1) for the standardization state
23
and status of this protocol. Distribution of this memo is unlimited.
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Copyright (C) The Internet Society (1998). All Rights Reserved.
31
The Real Time Streaming Protocol, or RTSP, is an application-level
32
protocol for control over the delivery of data with real-time
33
properties. RTSP provides an extensible framework to enable
34
controlled, on-demand delivery of real-time data, such as audio and
35
video. Sources of data can include both live data feeds and stored
36
clips. This protocol is intended to control multiple data delivery
37
sessions, provide a means for choosing delivery channels such as UDP,
38
multicast UDP and TCP, and provide a means for choosing delivery
39
mechanisms based upon RTP (RFC 1889).
43
* 1 Introduction ................................................. 5
44
+ 1.1 Purpose ............................................... 5
45
+ 1.2 Requirements .......................................... 6
46
+ 1.3 Terminology ........................................... 6
47
+ 1.4 Protocol Properties ................................... 9
48
+ 1.5 Extending RTSP ........................................ 11
49
+ 1.6 Overall Operation ..................................... 11
50
+ 1.7 RTSP States ........................................... 12
51
+ 1.8 Relationship with Other Protocols ..................... 13
52
* 2 Notational Conventions ....................................... 14
53
* 3 Protocol Parameters .......................................... 14
54
+ 3.1 RTSP Version .......................................... 14
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+ 3.2 RTSP URL .............................................. 14
64
+ 3.3 Conference Identifiers ................................ 16
65
+ 3.4 Session Identifiers ................................... 16
66
+ 3.5 SMPTE Relative Timestamps ............................. 16
67
+ 3.6 Normal Play Time ...................................... 17
68
+ 3.7 Absolute Time ......................................... 18
69
+ 3.8 Option Tags ........................................... 18
70
o 3.8.1 Registering New Option Tags with IANA .......... 18
71
* 4 RTSP Message ................................................. 19
72
+ 4.1 Message Types ......................................... 19
73
+ 4.2 Message Headers ....................................... 19
74
+ 4.3 Message Body .......................................... 19
75
+ 4.4 Message Length ........................................ 20
76
* 5 General Header Fields ........................................ 20
77
* 6 Request ...................................................... 20
78
+ 6.1 Request Line .......................................... 21
79
+ 6.2 Request Header Fields ................................. 21
80
* 7 Response ..................................................... 22
81
+ 7.1 Status-Line ........................................... 22
82
o 7.1.1 Status Code and Reason Phrase .................. 22
83
o 7.1.2 Response Header Fields ......................... 26
84
* 8 Entity ....................................................... 27
85
+ 8.1 Entity Header Fields .................................. 27
86
+ 8.2 Entity Body ........................................... 28
87
* 9 Connections .................................................. 28
88
+ 9.1 Pipelining ............................................ 28
89
+ 9.2 Reliability and Acknowledgements ...................... 28
90
* 10 Method Definitions .......................................... 29
91
+ 10.1 OPTIONS .............................................. 30
92
+ 10.2 DESCRIBE ............................................. 31
93
+ 10.3 ANNOUNCE ............................................. 32
94
+ 10.4 SETUP ................................................ 33
95
+ 10.5 PLAY ................................................. 34
96
+ 10.6 PAUSE ................................................ 36
97
+ 10.7 TEARDOWN ............................................. 37
98
+ 10.8 GET_PARAMETER ........................................ 37
99
+ 10.9 SET_PARAMETER ........................................ 38
100
+ 10.10 REDIRECT ............................................ 39
101
+ 10.11 RECORD .............................................. 39
102
+ 10.12 Embedded (Interleaved) Binary Data .................. 40
103
* 11 Status Code Definitions ..................................... 41
104
+ 11.1 Success 2xx .......................................... 41
105
o 11.1.1 250 Low on Storage Space ...................... 41
106
+ 11.2 Redirection 3xx ...................................... 41
107
+ 11.3 Client Error 4xx ..................................... 42
108
o 11.3.1 405 Method Not Allowed ........................ 42
109
o 11.3.2 451 Parameter Not Understood .................. 42
110
o 11.3.3 452 Conference Not Found ...................... 42
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o 11.3.4 453 Not Enough Bandwidth ...................... 42
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o 11.3.5 454 Session Not Found ......................... 42
121
o 11.3.6 455 Method Not Valid in This State ............ 42
122
o 11.3.7 456 Header Field Not Valid for Resource ....... 42
123
o 11.3.8 457 Invalid Range ............................. 43
124
o 11.3.9 458 Parameter Is Read-Only .................... 43
125
o 11.3.10 459 Aggregate Operation Not Allowed .......... 43
126
o 11.3.11 460 Only Aggregate Operation Allowed ......... 43
127
o 11.3.12 461 Unsupported Transport .................... 43
128
o 11.3.13 462 Destination Unreachable .................. 43
129
o 11.3.14 551 Option not supported ..................... 43
130
* 12 Header Field Definitions .................................... 44
131
+ 12.1 Accept ............................................... 46
132
+ 12.2 Accept-Encoding ...................................... 46
133
+ 12.3 Accept-Language ...................................... 46
134
+ 12.4 Allow ................................................ 46
135
+ 12.5 Authorization ........................................ 46
136
+ 12.6 Bandwidth ............................................ 46
137
+ 12.7 Blocksize ............................................ 47
138
+ 12.8 Cache-Control ........................................ 47
139
+ 12.9 Conference ........................................... 49
140
+ 12.10 Connection .......................................... 49
141
+ 12.11 Content-Base ........................................ 49
142
+ 12.12 Content-Encoding .................................... 49
143
+ 12.13 Content-Language .................................... 50
144
+ 12.14 Content-Length ...................................... 50
145
+ 12.15 Content-Location .................................... 50
146
+ 12.16 Content-Type ........................................ 50
147
+ 12.17 CSeq ................................................ 50
148
+ 12.18 Date ................................................ 50
149
+ 12.19 Expires ............................................. 50
150
+ 12.20 From ................................................ 51
151
+ 12.21 Host ................................................ 51
152
+ 12.22 If-Match ............................................ 51
153
+ 12.23 If-Modified-Since ................................... 52
154
+ 12.24 Last-Modified........................................ 52
155
+ 12.25 Location ............................................ 52
156
+ 12.26 Proxy-Authenticate .................................. 52
157
+ 12.27 Proxy-Require ....................................... 52
158
+ 12.28 Public .............................................. 53
159
+ 12.29 Range ............................................... 53
160
+ 12.30 Referer ............................................. 54
161
+ 12.31 Retry-After ......................................... 54
162
+ 12.32 Require ............................................. 54
163
+ 12.33 RTP-Info ............................................ 55
164
+ 12.34 Scale ............................................... 56
165
+ 12.35 Speed ............................................... 57
166
+ 12.36 Server .............................................. 57
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+ 12.37 Session ............................................. 57
176
+ 12.38 Timestamp ........................................... 58
177
+ 12.39 Transport ........................................... 58
178
+ 12.40 Unsupported ......................................... 62
179
+ 12.41 User-Agent .......................................... 62
180
+ 12.42 Vary ................................................ 62
181
+ 12.43 Via ................................................. 62
182
+ 12.44 WWW-Authenticate .................................... 62
183
* 13 Caching ..................................................... 62
184
* 14 Examples .................................................... 63
185
+ 14.1 Media on Demand (Unicast) ............................ 63
186
+ 14.2 Streaming of a Container file ........................ 65
187
+ 14.3 Single Stream Container Files ........................ 67
188
+ 14.4 Live Media Presentation Using Multicast .............. 69
189
+ 14.5 Playing media into an existing session ............... 70
190
+ 14.6 Recording ............................................ 71
191
* 15 Syntax ...................................................... 72
192
+ 15.1 Base Syntax .......................................... 72
193
* 16 Security Considerations ..................................... 73
194
* A RTSP Protocol State Machines ................................. 76
195
+ A.1 Client State Machine .................................. 76
196
+ A.2 Server State Machine .................................. 77
197
* B Interaction with RTP ......................................... 79
198
* C Use of SDP for RTSP Session Descriptions ..................... 80
199
+ C.1 Definitions ........................................... 80
200
o C.1.1 Control URL .................................... 80
201
o C.1.2 Media streams .................................. 81
202
o C.1.3 Payload type(s) ................................ 81
203
o C.1.4 Format-specific parameters ..................... 81
204
o C.1.5 Range of presentation .......................... 82
205
o C.1.6 Time of availability ........................... 82
206
o C.1.7 Connection Information ......................... 82
207
o C.1.8 Entity Tag ..................................... 82
208
+ C.2 Aggregate Control Not Available ....................... 83
209
+ C.3 Aggregate Control Available ........................... 83
210
* D Minimal RTSP implementation .................................. 85
211
+ D.1 Client ................................................ 85
212
o D.1.1 Basic Playback ................................. 86
213
o D.1.2 Authentication-enabled ......................... 86
214
+ D.2 Server ................................................ 86
215
o D.2.1 Basic Playback ................................. 87
216
o D.2.2 Authentication-enabled ......................... 87
217
* E Authors' Addresses ........................................... 88
218
* F Acknowledgements ............................................. 89
219
* References ..................................................... 90
220
* Full Copyright Statement ....................................... 92
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The Real-Time Streaming Protocol (RTSP) establishes and controls
236
either a single or several time-synchronized streams of continuous
237
media such as audio and video. It does not typically deliver the
238
continuous streams itself, although interleaving of the continuous
239
media stream with the control stream is possible (see Section 10.12).
240
In other words, RTSP acts as a "network remote control" for
243
The set of streams to be controlled is defined by a presentation
244
description. This memorandum does not define a format for a
245
presentation description.
247
There is no notion of an RTSP connection; instead, a server maintains
248
a session labeled by an identifier. An RTSP session is in no way tied
249
to a transport-level connection such as a TCP connection. During an
250
RTSP session, an RTSP client may open and close many reliable
251
transport connections to the server to issue RTSP requests.
252
Alternatively, it may use a connectionless transport protocol such as
255
The streams controlled by RTSP may use RTP [1], but the operation of
256
RTSP does not depend on the transport mechanism used to carry
257
continuous media. The protocol is intentionally similar in syntax
258
and operation to HTTP/1.1 [2] so that extension mechanisms to HTTP
259
can in most cases also be added to RTSP. However, RTSP differs in a
260
number of important aspects from HTTP:
262
* RTSP introduces a number of new methods and has a different
264
* An RTSP server needs to maintain state by default in almost all
265
cases, as opposed to the stateless nature of HTTP.
266
* Both an RTSP server and client can issue requests.
267
* Data is carried out-of-band by a different protocol. (There is an
269
* RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1,
270
consistent with current HTML internationalization efforts [3].
271
* The Request-URI always contains the absolute URI. Because of
272
backward compatibility with a historical blunder, HTTP/1.1 [2]
273
carries only the absolute path in the request and puts the host
274
name in a separate header field.
276
This makes "virtual hosting" easier, where a single host with one
277
IP address hosts several document trees.
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The protocol supports the following operations:
289
Retrieval of media from media server:
290
The client can request a presentation description via HTTP or
291
some other method. If the presentation is being multicast, the
292
presentation description contains the multicast addresses and
293
ports to be used for the continuous media. If the presentation
294
is to be sent only to the client via unicast, the client
295
provides the destination for security reasons.
297
Invitation of a media server to a conference:
298
A media server can be "invited" to join an existing
299
conference, either to play back media into the presentation or
300
to record all or a subset of the media in a presentation. This
301
mode is useful for distributed teaching applications. Several
302
parties in the conference may take turns "pushing the remote
305
Addition of media to an existing presentation:
306
Particularly for live presentations, it is useful if the
307
server can tell the client about additional media becoming
310
RTSP requests may be handled by proxies, tunnels and caches as in
315
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
316
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
317
document are to be interpreted as described in RFC 2119 [4].
321
Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not
322
listed here are defined as in HTTP/1.1.
325
The control of the multiple streams using a single timeline by
326
the server. For audio/video feeds, this means that the client
327
may issue a single play or pause message to control both the
328
audio and video feeds.
331
a multiparty, multimedia presentation, where "multi" implies
332
greater than or equal to one.
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The client requests continuous media data from the media
348
A transport layer virtual circuit established between two
349
programs for the purpose of communication.
352
A file which may contain multiple media streams which often
353
comprise a presentation when played together. RTSP servers may
354
offer aggregate control on these files, though the concept of
355
a container file is not embedded in the protocol.
358
Data where there is a timing relationship between source and
359
sink; that is, the sink must reproduce the timing relationship
360
that existed at the source. The most common examples of
361
continuous media are audio and motion video. Continuous media
362
can be real-time (interactive), where there is a "tight"
363
timing relationship between source and sink, or streaming
364
(playback), where the relationship is less strict.
367
The information transferred as the payload of a request or
368
response. An entity consists of metainformation in the form of
369
entity-header fields and content in the form of an entity-
370
body, as described in Section 8.
372
Media initialization:
373
Datatype/codec specific initialization. This includes such
374
things as clockrates, color tables, etc. Any transport-
375
independent information which is required by a client for
376
playback of a media stream occurs in the media initialization
377
phase of stream setup.
380
Parameter specific to a media type that may be changed before
381
or during stream playback.
384
The server providing playback or recording services for one or
385
more media streams. Different media streams within a
386
presentation may originate from different media servers. A
387
media server may reside on the same or a different host as the
388
web server the presentation is invoked from.
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Media server indirection:
400
Redirection of a media client to a different media server.
403
A single media instance, e.g., an audio stream or a video
404
stream as well as a single whiteboard or shared application
405
group. When using RTP, a stream consists of all RTP and RTCP
406
packets created by a source within an RTP session. This is
407
equivalent to the definition of a DSM-CC stream([5]).
410
The basic unit of RTSP communication, consisting of a
411
structured sequence of octets matching the syntax defined in
412
Section 15 and transmitted via a connection or a
413
connectionless protocol.
416
Member of a conference. A participant may be a machine, e.g.,
417
a media record or playback server.
420
A set of one or more streams presented to the client as a
421
complete media feed, using a presentation description as
422
defined below. In most cases in the RTSP context, this implies
423
aggregate control of those streams, but does not have to.
425
Presentation description:
426
A presentation description contains information about one or
427
more media streams within a presentation, such as the set of
428
encodings, network addresses and information about the
429
content. Other IETF protocols such as SDP (RFC 2327 [6]) use
430
the term "session" for a live presentation. The presentation
431
description may take several different formats, including but
432
not limited to the session description format SDP.
435
An RTSP response. If an HTTP response is meant, that is
436
indicated explicitly.
439
An RTSP request. If an HTTP request is meant, that is
440
indicated explicitly.
443
A complete RTSP "transaction", e.g., the viewing of a movie.
444
A session typically consists of a client setting up a
445
transport mechanism for the continuous media stream (SETUP),
446
starting the stream with PLAY or RECORD, and closing the
450
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stream with TEARDOWN.
457
Transport initialization:
458
The negotiation of transport information (e.g., port numbers,
459
transport protocols) between the client and the server.
461
1.4 Protocol Properties
463
RTSP has the following properties:
466
New methods and parameters can be easily added to RTSP.
469
RTSP can be parsed by standard HTTP or MIME parsers.
472
RTSP re-uses web security mechanisms. All HTTP authentication
473
mechanisms such as basic (RFC 2068 [2, Section 11.1]) and
474
digest authentication (RFC 2069 [8]) are directly applicable.
475
One may also reuse transport or network layer security
478
Transport-independent:
479
RTSP may use either an unreliable datagram protocol (UDP) (RFC
480
768 [9]), a reliable datagram protocol (RDP, RFC 1151, not
481
widely used [10]) or a reliable stream protocol such as TCP
482
(RFC 793 [11]) as it implements application-level reliability.
484
Multi-server capable:
485
Each media stream within a presentation can reside on a
486
different server. The client automatically establishes several
487
concurrent control sessions with the different media servers.
488
Media synchronization is performed at the transport level.
490
Control of recording devices:
491
The protocol can control both recording and playback devices,
492
as well as devices that can alternate between the two modes
495
Separation of stream control and conference initiation:
496
Stream control is divorced from inviting a media server to a
497
conference. The only requirement is that the conference
498
initiation protocol either provides or can be used to create a
499
unique conference identifier. In particular, SIP [12] or H.323
500
[13] may be used to invite a server to a conference.
506
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Suitable for professional applications:
512
RTSP supports frame-level accuracy through SMPTE time stamps
513
to allow remote digital editing.
515
Presentation description neutral:
516
The protocol does not impose a particular presentation
517
description or metafile format and can convey the type of
518
format to be used. However, the presentation description must
519
contain at least one RTSP URI.
521
Proxy and firewall friendly:
522
The protocol should be readily handled by both application and
523
transport-layer (SOCKS [14]) firewalls. A firewall may need to
524
understand the SETUP method to open a "hole" for the UDP media
528
Where sensible, RTSP reuses HTTP concepts, so that the
529
existing infrastructure can be reused. This infrastructure
530
includes PICS (Platform for Internet Content Selection
531
[15,16]) for associating labels with content. However, RTSP
532
does not just add methods to HTTP since the controlling
533
continuous media requires server state in most cases.
535
Appropriate server control:
536
If a client can start a stream, it must be able to stop a
537
stream. Servers should not start streaming to clients in such
538
a way that clients cannot stop the stream.
540
Transport negotiation:
541
The client can negotiate the transport method prior to
542
actually needing to process a continuous media stream.
544
Capability negotiation:
545
If basic features are disabled, there must be some clean
546
mechanism for the client to determine which methods are not
547
going to be implemented. This allows clients to present the
548
appropriate user interface. For example, if seeking is not
549
allowed, the user interface must be able to disallow moving a
550
sliding position indicator.
552
An earlier requirement in RTSP was multi-client capability.
553
However, it was determined that a better approach was to make sure
554
that the protocol is easily extensible to the multi-client
555
scenario. Stream identifiers can be used by several control
556
streams, so that "passing the remote" would be possible. The
557
protocol would not address how several clients negotiate access;
558
this is left to either a "social protocol" or some other floor
562
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Since not all media servers have the same functionality, media
572
servers by necessity will support different sets of requests. For
575
* A server may only be capable of playback thus has no need to
576
support the RECORD request.
577
* A server may not be capable of seeking (absolute positioning) if
578
it is to support live events only.
579
* Some servers may not support setting stream parameters and thus
580
not support GET_PARAMETER and SET_PARAMETER.
582
A server SHOULD implement all header fields described in Section 12.
584
It is up to the creators of presentation descriptions not to ask the
585
impossible of a server. This situation is similar in HTTP/1.1 [2],
586
where the methods described in [H19.6] are not likely to be supported
589
RTSP can be extended in three ways, listed here in order of the
590
magnitude of changes supported:
592
* Existing methods can be extended with new parameters, as long as
593
these parameters can be safely ignored by the recipient. (This is
594
equivalent to adding new parameters to an HTML tag.) If the
595
client needs negative acknowledgement when a method extension is
596
not supported, a tag corresponding to the extension may be added
597
in the Require: field (see Section 12.32).
598
* New methods can be added. If the recipient of the message does
599
not understand the request, it responds with error code 501 (Not
600
implemented) and the sender should not attempt to use this method
601
again. A client may also use the OPTIONS method to inquire about
602
methods supported by the server. The server SHOULD list the
603
methods it supports using the Public response header.
604
* A new version of the protocol can be defined, allowing almost all
605
aspects (except the position of the protocol version number) to
608
1.6 Overall Operation
610
Each presentation and media stream may be identified by an RTSP URL.
611
The overall presentation and the properties of the media the
612
presentation is made up of are defined by a presentation description
613
file, the format of which is outside the scope of this specification.
614
The presentation description file may be obtained by the client using
618
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623
HTTP or other means such as email and may not necessarily be stored
626
For the purposes of this specification, a presentation description is
627
assumed to describe one or more presentations, each of which
628
maintains a common time axis. For simplicity of exposition and
629
without loss of generality, it is assumed that the presentation
630
description contains exactly one such presentation. A presentation
631
may contain several media streams.
633
The presentation description file contains a description of the media
634
streams making up the presentation, including their encodings,
635
language, and other parameters that enable the client to choose the
636
most appropriate combination of media. In this presentation
637
description, each media stream that is individually controllable by
638
RTSP is identified by an RTSP URL, which points to the media server
639
handling that particular media stream and names the stream stored on
640
that server. Several media streams can be located on different
641
servers; for example, audio and video streams can be split across
642
servers for load sharing. The description also enumerates which
643
transport methods the server is capable of.
645
Besides the media parameters, the network destination address and
646
port need to be determined. Several modes of operation can be
650
The media is transmitted to the source of the RTSP request,
651
with the port number chosen by the client. Alternatively, the
652
media is transmitted on the same reliable stream as RTSP.
654
Multicast, server chooses address:
655
The media server picks the multicast address and port. This is
656
the typical case for a live or near-media-on-demand
659
Multicast, client chooses address:
660
If the server is to participate in an existing multicast
661
conference, the multicast address, port and encryption key are
662
given by the conference description, established by means
663
outside the scope of this specification.
667
RTSP controls a stream which may be sent via a separate protocol,
668
independent of the control channel. For example, RTSP control may
669
occur on a TCP connection while the data flows via UDP. Thus, data
670
delivery continues even if no RTSP requests are received by the media
674
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679
server. Also, during its lifetime, a single media stream may be
680
controlled by RTSP requests issued sequentially on different TCP
681
connections. Therefore, the server needs to maintain "session state"
682
to be able to correlate RTSP requests with a stream. The state
683
transitions are described in Section A.
685
Many methods in RTSP do not contribute to state. However, the
686
following play a central role in defining the allocation and usage of
687
stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and
691
Causes the server to allocate resources for a stream and start
695
Starts data transmission on a stream allocated via SETUP.
698
Temporarily halts a stream without freeing server resources.
701
Frees resources associated with the stream. The RTSP session
702
ceases to exist on the server.
704
RTSP methods that contribute to state use the Session header
705
field (Section 12.37) to identify the RTSP session whose state
706
is being manipulated. The server generates session identifiers
707
in response to SETUP requests (Section 10.4).
709
1.8 Relationship with Other Protocols
711
RTSP has some overlap in functionality with HTTP. It also may
712
interact with HTTP in that the initial contact with streaming content
713
is often to be made through a web page. The current protocol
714
specification aims to allow different hand-off points between a web
715
server and the media server implementing RTSP. For example, the
716
presentation description can be retrieved using HTTP or RTSP, which
717
reduces roundtrips in web-browser-based scenarios, yet also allows
718
for standalone RTSP servers and clients which do not rely on HTTP at
721
However, RTSP differs fundamentally from HTTP in that data delivery
722
takes place out-of-band in a different protocol. HTTP is an
723
asymmetric protocol where the client issues requests and the server
724
responds. In RTSP, both the media client and media server can issue
725
requests. RTSP requests are also not stateless; they may set
726
parameters and continue to control a media stream long after the
730
Schulzrinne, et. al. Standards Track [Page 13]
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RFC 2326 Real Time Streaming Protocol April 1998
735
request has been acknowledged.
737
Re-using HTTP functionality has advantages in at least two areas,
738
namely security and proxies. The requirements are very similar, so
739
having the ability to adopt HTTP work on caches, proxies and
740
authentication is valuable.
742
While most real-time media will use RTP as a transport protocol, RTSP
745
RTSP assumes the existence of a presentation description format that
746
can express both static and temporal properties of a presentation
747
containing several media streams.
749
2 Notational Conventions
751
Since many of the definitions and syntax are identical to HTTP/1.1,
752
this specification only points to the section where they are defined
753
rather than copying it. For brevity, [HX.Y] is to be taken to refer
754
to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]).
756
All the mechanisms specified in this document are described in both
757
prose and an augmented Backus-Naur form (BNF) similar to that used in
758
[H2.1]. It is described in detail in RFC 2234 [17], with the
759
difference that this RTSP specification maintains the "1#" notation
760
for comma-separated lists.
762
In this memo, we use indented and smaller-type paragraphs to provide
763
background and motivation. This is intended to give readers who were
764
not involved with the formulation of the specification an
765
understanding of why things are the way that they are in RTSP.
767
3 Protocol Parameters
771
[H3.1] applies, with HTTP replaced by RTSP.
775
The "rtsp" and "rtspu" schemes are used to refer to network resources
776
via the RTSP protocol. This section defines the scheme-specific
777
syntax and semantics for RTSP URLs.
779
rtsp_URL = ( "rtsp:" | "rtspu:" )
780
"//" host [ ":" port ] [ abs_path ]
781
host = <A legal Internet host domain name of IP address
782
(in dotted decimal form), as defined by Section 2.1
786
Schulzrinne, et. al. Standards Track [Page 14]
788
RFC 2326 Real Time Streaming Protocol April 1998
791
of RFC 1123 \cite{rfc1123}>
794
abs_path is defined in [H3.2.1].
796
Note that fragment and query identifiers do not have a well-defined
797
meaning at this time, with the interpretation left to the RTSP
800
The scheme rtsp requires that commands are issued via a reliable
801
protocol (within the Internet, TCP), while the scheme rtspu identifies
802
an unreliable protocol (within the Internet, UDP).
804
If the port is empty or not given, port 554 is assumed. The semantics
805
are that the identified resource can be controlled by RTSP at the
806
server listening for TCP (scheme "rtsp") connections or UDP (scheme
807
"rtspu") packets on that port of host, and the Request-URI for the
808
resource is rtsp_URL.
810
The use of IP addresses in URLs SHOULD be avoided whenever possible
813
A presentation or a stream is identified by a textual media
814
identifier, using the character set and escape conventions [H3.2] of
815
URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of
816
streams, i.e., a presentation. Accordingly, requests described in
817
Section 10 can apply to either the whole presentation or an individual
818
stream within the presentation. Note that some request methods can
819
only be applied to streams, not presentations and vice versa.
821
For example, the RTSP URL:
822
rtsp://media.example.com:554/twister/audiotrack
824
identifies the audio stream within the presentation "twister", which
825
can be controlled via RTSP requests issued over a TCP connection to
826
port 554 of host media.example.com.
829
rtsp://media.example.com:554/twister
831
identifies the presentation "twister", which may be composed of
832
audio and video streams.
834
This does not imply a standard way to reference streams in URLs.
835
The presentation description defines the hierarchical relationships
836
in the presentation and the URLs for the individual streams. A
837
presentation description may name a stream "a.mov" and the whole
838
presentation "b.mov".
842
Schulzrinne, et. al. Standards Track [Page 15]
844
RFC 2326 Real Time Streaming Protocol April 1998
847
The path components of the RTSP URL are opaque to the client and do
848
not imply any particular file system structure for the server.
850
This decoupling also allows presentation descriptions to be used
851
with non-RTSP media control protocols simply by replacing the
854
3.3 Conference Identifiers
856
Conference identifiers are opaque to RTSP and are encoded using
857
standard URI encoding methods (i.e., LWS is escaped with %). They can
858
contain any octet value. The conference identifier MUST be globally
859
unique. For H.323, the conferenceID value is to be used.
861
conference-id = 1*xchar
863
Conference identifiers are used to allow RTSP sessions to obtain
864
parameters from multimedia conferences the media server is
865
participating in. These conferences are created by protocols
866
outside the scope of this specification, e.g., H.323 [13] or SIP
867
[12]. Instead of the RTSP client explicitly providing transport
868
information, for example, it asks the media server to use the
869
values in the conference description instead.
871
3.4 Session Identifiers
873
Session identifiers are opaque strings of arbitrary length. Linear
874
white space must be URL-escaped. A session identifier MUST be chosen
875
randomly and MUST be at least eight octets long to make guessing it
876
more difficult. (See Section 16.)
878
session-id = 1*( ALPHA | DIGIT | safe )
880
3.5 SMPTE Relative Timestamps
882
A SMPTE relative timestamp expresses time relative to the start of
883
the clip. Relative timestamps are expressed as SMPTE time codes for
884
frame-level access accuracy. The time code has the format
885
hours:minutes:seconds:frames.subframes, with the origin at the start
886
of the clip. The default smpte format is "SMPTE 30 drop" format, with
887
frame rate is 29.97 frames per second. Other SMPTE codes MAY be
888
supported (such as "SMPTE 25") through the use of alternative use of
889
"smpte time". For the "frames" field in the time value can assume
890
the values 0 through 29. The difference between 30 and 29.97 frames
891
per second is handled by dropping the first two frame indices (values
892
00 and 01) of every minute, except every tenth minute. If the frame
893
value is zero, it may be omitted. Subframes are measured in
894
one-hundredth of a frame.
898
Schulzrinne, et. al. Standards Track [Page 16]
900
RFC 2326 Real Time Streaming Protocol April 1998
903
smpte-range = smpte-type "=" smpte-time "-" [ smpte-time ]
904
smpte-type = "smpte" | "smpte-30-drop" | "smpte-25"
905
; other timecodes may be added
906
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ]
912
smpte=10:07:00-10:07:33:05.01
913
smpte-25=10:07:00-10:07:33:05.01
917
Normal play time (NPT) indicates the stream absolute position
918
relative to the beginning of the presentation. The timestamp consists
919
of a decimal fraction. The part left of the decimal may be expressed
920
in either seconds or hours, minutes, and seconds. The part right of
921
the decimal point measures fractions of a second.
923
The beginning of a presentation corresponds to 0.0 seconds. Negative
924
values are not defined. The special constant now is defined as the
925
current instant of a live event. It may be used only for live events.
927
NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the
928
viewer associates with a program. It is often digitally displayed on
929
a VCR. NPT advances normally when in normal play mode (scale = 1),
930
advances at a faster rate when in fast scan forward (high positive
931
scale ratio), decrements when in scan reverse (high negative scale
932
ratio) and is fixed in pause mode. NPT is (logically) equivalent to
933
SMPTE time codes." [5]
935
npt-range = ( npt-time "-" [ npt-time ] ) | ( "-" npt-time )
936
npt-time = "now" | npt-sec | npt-hhmmss
937
npt-sec = 1*DIGIT [ "." *DIGIT ]
938
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
939
npt-hh = 1*DIGIT ; any positive number
940
npt-mm = 1*2DIGIT ; 0-59
941
npt-ss = 1*2DIGIT ; 0-59
948
The syntax conforms to ISO 8601. The npt-sec notation is optimized
949
for automatic generation, the ntp-hhmmss notation for consumption
950
by human readers. The "now" constant allows clients to request to
954
Schulzrinne, et. al. Standards Track [Page 17]
956
RFC 2326 Real Time Streaming Protocol April 1998
959
receive the live feed rather than the stored or time-delayed
960
version. This is needed since neither absolute time nor zero time
961
are appropriate for this case.
965
Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT).
966
Fractions of a second may be indicated.
968
utc-range = "clock" "=" utc-time "-" [ utc-time ]
969
utc-time = utc-date "T" utc-time "Z"
970
utc-date = 8DIGIT ; < YYYYMMDD >
971
utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction >
973
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
980
Option tags are unique identifiers used to designate new options in
981
RTSP. These tags are used in Require (Section 12.32) and Proxy-
982
Require (Section 12.27) header fields.
988
The creator of a new RTSP option should either prefix the option with
989
a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name
990
for a feature whose inventor can be reached at "foo.com"), or
991
register the new option with the Internet Assigned Numbers Authority
994
3.8.1 Registering New Option Tags with IANA
996
When registering a new RTSP option, the following information should
999
* Name and description of option. The name may be of any length,
1000
but SHOULD be no more than twenty characters long. The name MUST
1001
not contain any spaces, control characters or periods.
1002
* Indication of who has change control over the option (for
1003
example, IETF, ISO, ITU-T, other international standardization
1004
bodies, a consortium or a particular company or group of
1010
Schulzrinne, et. al. Standards Track [Page 18]
1012
RFC 2326 Real Time Streaming Protocol April 1998
1015
* A reference to a further description, if available, for example
1016
(in order of preference) an RFC, a published paper, a patent
1017
filing, a technical report, documented source code or a computer
1019
* For proprietary options, contact information (postal and email
1024
RTSP is a text-based protocol and uses the ISO 10646 character set in
1025
UTF-8 encoding (RFC 2279 [21]). Lines are terminated by CRLF, but
1026
receivers should be prepared to also interpret CR and LF by
1027
themselves as line terminators.
1029
Text-based protocols make it easier to add optional parameters in a
1030
self-describing manner. Since the number of parameters and the
1031
frequency of commands is low, processing efficiency is not a
1032
concern. Text-based protocols, if done carefully, also allow easy
1033
implementation of research prototypes in scripting languages such
1034
as Tcl, Visual Basic and Perl.
1036
The 10646 character set avoids tricky character set switching, but
1037
is invisible to the application as long as US-ASCII is being used.
1038
This is also the encoding used for RTCP. ISO 8859-1 translates
1039
directly into Unicode with a high-order octet of zero. ISO 8859-1
1040
characters with the most-significant bit set are represented as
1041
1100001x 10xxxxxx. (See RFC 2279 [21])
1043
RTSP messages can be carried over any lower-layer transport protocol
1044
that is 8-bit clean.
1046
Requests contain methods, the object the method is operating upon and
1047
parameters to further describe the method. Methods are idempotent,
1048
unless otherwise noted. Methods are also designed to require little
1049
or no state maintenance at the media server.
1066
Schulzrinne, et. al. Standards Track [Page 19]
1068
RFC 2326 Real Time Streaming Protocol April 1998
1073
When a message body is included with a message, the length of that
1074
body is determined by one of the following (in order of precedence):
1076
1. Any response message which MUST NOT include a message body
1077
(such as the 1xx, 204, and 304 responses) is always terminated
1078
by the first empty line after the header fields, regardless of
1079
the entity-header fields present in the message. (Note: An
1080
empty line consists of only CRLF.)
1082
2. If a Content-Length header field (section 12.14) is present,
1083
its value in bytes represents the length of the message-body.
1084
If this header field is not present, a value of zero is
1087
3. By the server closing the connection. (Closing the connection
1088
cannot be used to indicate the end of a request body, since
1089
that would leave no possibility for the server to send back a
1092
Note that RTSP does not (at present) support the HTTP/1.1 "chunked"
1093
transfer coding(see [H3.6]) and requires the presence of the
1094
Content-Length header field.
1096
Given the moderate length of presentation descriptions returned,
1097
the server should always be able to determine its length, even if
1098
it is generated dynamically, making the chunked transfer encoding
1099
unnecessary. Even though Content-Length must be present if there is
1100
any entity body, the rules ensure reasonable behavior even if the
1101
length is not given explicitly.
1103
5 General Header Fields
1105
See [H4.5], except that Pragma, Transfer-Encoding and Upgrade headers
1108
general-header = Cache-Control ; Section 12.8
1109
| Connection ; Section 12.10
1110
| Date ; Section 12.18
1111
| Via ; Section 12.43
1115
A request message from a client to a server or vice versa includes,
1116
within the first line of that message, the method to be applied to
1117
the resource, the identifier of the resource, and the protocol
1122
Schulzrinne, et. al. Standards Track [Page 20]
1124
RFC 2326 Real Time Streaming Protocol April 1998
1127
Request = Request-Line ; Section 6.1
1128
*( general-header ; Section 5
1129
| request-header ; Section 6.2
1130
| entity-header ) ; Section 8.1
1132
[ message-body ] ; Section 4.3
1136
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
1138
Method = "DESCRIBE" ; Section 10.2
1139
| "ANNOUNCE" ; Section 10.3
1140
| "GET_PARAMETER" ; Section 10.8
1141
| "OPTIONS" ; Section 10.1
1142
| "PAUSE" ; Section 10.6
1143
| "PLAY" ; Section 10.5
1144
| "RECORD" ; Section 10.11
1145
| "REDIRECT" ; Section 10.10
1146
| "SETUP" ; Section 10.4
1147
| "SET_PARAMETER" ; Section 10.9
1148
| "TEARDOWN" ; Section 10.7
1151
extension-method = token
1153
Request-URI = "*" | absolute_URI
1155
RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT
1157
6.2 Request Header Fields
1159
request-header = Accept ; Section 12.1
1160
| Accept-Encoding ; Section 12.2
1161
| Accept-Language ; Section 12.3
1162
| Authorization ; Section 12.5
1163
| From ; Section 12.20
1164
| If-Modified-Since ; Section 12.23
1165
| Range ; Section 12.29
1166
| Referer ; Section 12.30
1167
| User-Agent ; Section 12.41
1169
Note that in contrast to HTTP/1.1 [2], RTSP requests always contain
1170
the absolute URL (that is, including the scheme, host and port)
1171
rather than just the absolute path.
1178
Schulzrinne, et. al. Standards Track [Page 21]
1180
RFC 2326 Real Time Streaming Protocol April 1998
1183
HTTP/1.1 requires servers to understand the absolute URL, but
1184
clients are supposed to use the Host request header. This is purely
1185
needed for backward-compatibility with HTTP/1.0 servers, a
1186
consideration that does not apply to RTSP.
1188
The asterisk "*" in the Request-URI means that the request does not
1189
apply to a particular resource, but to the server itself, and is only
1190
allowed when the method used does not necessarily apply to a
1191
resource. One example would be:
1197
[H6] applies except that HTTP-Version is replaced by RTSP-Version.
1198
Also, RTSP defines additional status codes and does not define some
1199
HTTP codes. The valid response codes and the methods they can be used
1200
with are defined in Table 1.
1202
After receiving and interpreting a request message, the recipient
1203
responds with an RTSP response message.
1205
Response = Status-Line ; Section 7.1
1206
*( general-header ; Section 5
1207
| response-header ; Section 7.1.2
1208
| entity-header ) ; Section 8.1
1210
[ message-body ] ; Section 4.3
1214
The first line of a Response message is the Status-Line, consisting
1215
of the protocol version followed by a numeric status code, and the
1216
textual phrase associated with the status code, with each element
1217
separated by SP characters. No CR or LF is allowed except in the
1218
final CRLF sequence.
1220
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
1222
7.1.1 Status Code and Reason Phrase
1224
The Status-Code element is a 3-digit integer result code of the
1225
attempt to understand and satisfy the request. These codes are fully
1226
defined in Section 11. The Reason-Phrase is intended to give a short
1227
textual description of the Status-Code. The Status-Code is intended
1228
for use by automata and the Reason-Phrase is intended for the human
1229
user. The client is not required to examine or display the Reason-
1234
Schulzrinne, et. al. Standards Track [Page 22]
1236
RFC 2326 Real Time Streaming Protocol April 1998
1239
The first digit of the Status-Code defines the class of response. The
1240
last two digits do not have any categorization role. There are 5
1241
values for the first digit:
1243
* 1xx: Informational - Request received, continuing process
1244
* 2xx: Success - The action was successfully received, understood,
1246
* 3xx: Redirection - Further action must be taken in order to
1247
complete the request
1248
* 4xx: Client Error - The request contains bad syntax or cannot be
1250
* 5xx: Server Error - The server failed to fulfill an apparently
1253
The individual values of the numeric status codes defined for
1254
RTSP/1.0, and an example set of corresponding Reason-Phrase's, are
1255
presented below. The reason phrases listed here are only recommended
1256
- they may be replaced by local equivalents without affecting the
1257
protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and
1258
adds RTSP-specific status codes starting at x50 to avoid conflicts
1259
with newly defined HTTP status codes.
1290
Schulzrinne, et. al. Standards Track [Page 23]
1292
RFC 2326 Real Time Streaming Protocol April 1998
1295
Status-Code = "100" ; Continue
1298
| "250" ; Low on Storage Space
1299
| "300" ; Multiple Choices
1300
| "301" ; Moved Permanently
1301
| "302" ; Moved Temporarily
1303
| "304" ; Not Modified
1305
| "400" ; Bad Request
1306
| "401" ; Unauthorized
1307
| "402" ; Payment Required
1310
| "405" ; Method Not Allowed
1311
| "406" ; Not Acceptable
1312
| "407" ; Proxy Authentication Required
1313
| "408" ; Request Time-out
1315
| "411" ; Length Required
1316
| "412" ; Precondition Failed
1317
| "413" ; Request Entity Too Large
1318
| "414" ; Request-URI Too Large
1319
| "415" ; Unsupported Media Type
1320
| "451" ; Parameter Not Understood
1321
| "452" ; Conference Not Found
1322
| "453" ; Not Enough Bandwidth
1323
| "454" ; Session Not Found
1324
| "455" ; Method Not Valid in This State
1325
| "456" ; Header Field Not Valid for Resource
1326
| "457" ; Invalid Range
1327
| "458" ; Parameter Is Read-Only
1328
| "459" ; Aggregate operation not allowed
1329
| "460" ; Only aggregate operation allowed
1330
| "461" ; Unsupported transport
1331
| "462" ; Destination unreachable
1332
| "500" ; Internal Server Error
1333
| "501" ; Not Implemented
1334
| "502" ; Bad Gateway
1335
| "503" ; Service Unavailable
1336
| "504" ; Gateway Time-out
1337
| "505" ; RTSP Version not supported
1338
| "551" ; Option not supported
1346
Schulzrinne, et. al. Standards Track [Page 24]
1348
RFC 2326 Real Time Streaming Protocol April 1998
1351
extension-code = 3DIGIT
1353
Reason-Phrase = *<TEXT, excluding CR, LF>
1355
RTSP status codes are extensible. RTSP applications are not required
1356
to understand the meaning of all registered status codes, though such
1357
understanding is obviously desirable. However, applications MUST
1358
understand the class of any status code, as indicated by the first
1359
digit, and treat any unrecognized response as being equivalent to the
1360
x00 status code of that class, with the exception that an
1361
unrecognized response MUST NOT be cached. For example, if an
1362
unrecognized status code of 431 is received by the client, it can
1363
safely assume that there was something wrong with its request and
1364
treat the response as if it had received a 400 status code. In such
1365
cases, user agents SHOULD present to the user the entity returned
1366
with the response, since that entity is likely to include human-
1367
readable information which will explain the unusual status.
1375
250 Low on Storage Space RECORD
1377
300 Multiple Choices all
1378
301 Moved Permanently all
1379
302 Moved Temporarily all
1402
Schulzrinne, et. al. Standards Track [Page 25]
1404
RFC 2326 Real Time Streaming Protocol April 1998
1408
401 Unauthorized all
1409
402 Payment Required all
1412
405 Method Not Allowed all
1413
406 Not Acceptable all
1414
407 Proxy Authentication Required all
1415
408 Request Timeout all
1417
411 Length Required all
1418
412 Precondition Failed DESCRIBE, SETUP
1419
413 Request Entity Too Large all
1420
414 Request-URI Too Long all
1421
415 Unsupported Media Type all
1422
451 Invalid parameter SETUP
1423
452 Illegal Conference Identifier SETUP
1424
453 Not Enough Bandwidth SETUP
1425
454 Session Not Found all
1426
455 Method Not Valid In This State all
1427
456 Header Field Not Valid all
1428
457 Invalid Range PLAY
1429
458 Parameter Is Read-Only SET_PARAMETER
1430
459 Aggregate Operation Not Allowed all
1431
460 Only Aggregate Operation Allowed all
1432
461 Unsupported Transport all
1433
462 Destination Unreachable all
1435
500 Internal Server Error all
1436
501 Not Implemented all
1438
503 Service Unavailable all
1439
504 Gateway Timeout all
1440
505 RTSP Version Not Supported all
1441
551 Option not support all
1444
Table 1: Status codes and their usage with RTSP methods
1446
7.1.2 Response Header Fields
1448
The response-header fields allow the request recipient to pass
1449
additional information about the response which cannot be placed in
1450
the Status-Line. These header fields give information about the
1451
server and about further access to the resource identified by the
1458
Schulzrinne, et. al. Standards Track [Page 26]
1460
RFC 2326 Real Time Streaming Protocol April 1998
1463
response-header = Location ; Section 12.25
1464
| Proxy-Authenticate ; Section 12.26
1465
| Public ; Section 12.28
1466
| Retry-After ; Section 12.31
1467
| Server ; Section 12.36
1468
| Vary ; Section 12.42
1469
| WWW-Authenticate ; Section 12.44
1471
Response-header field names can be extended reliably only in
1472
combination with a change in the protocol version. However, new or
1473
experimental header fields MAY be given the semantics of response-
1474
header fields if all parties in the communication recognize them to
1475
be response-header fields. Unrecognized header fields are treated as
1476
entity-header fields.
1480
Request and Response messages MAY transfer an entity if not otherwise
1481
restricted by the request method or response status code. An entity
1482
consists of entity-header fields and an entity-body, although some
1483
responses will only include the entity-headers.
1485
In this section, both sender and recipient refer to either the client
1486
or the server, depending on who sends and who receives the entity.
1488
8.1 Entity Header Fields
1490
Entity-header fields define optional metainformation about the
1491
entity-body or, if no body is present, about the resource identified
1494
entity-header = Allow ; Section 12.4
1495
| Content-Base ; Section 12.11
1496
| Content-Encoding ; Section 12.12
1497
| Content-Language ; Section 12.13
1498
| Content-Length ; Section 12.14
1499
| Content-Location ; Section 12.15
1500
| Content-Type ; Section 12.16
1501
| Expires ; Section 12.19
1502
| Last-Modified ; Section 12.24
1504
extension-header = message-header
1506
The extension-header mechanism allows additional entity-header fields
1507
to be defined without changing the protocol, but these fields cannot
1508
be assumed to be recognizable by the recipient. Unrecognized header
1509
fields SHOULD be ignored by the recipient and forwarded by proxies.
1514
Schulzrinne, et. al. Standards Track [Page 27]
1516
RFC 2326 Real Time Streaming Protocol April 1998
1525
RTSP requests can be transmitted in several different ways:
1527
* persistent transport connections used for several
1528
request-response transactions;
1529
* one connection per request/response transaction;
1530
* connectionless mode.
1532
The type of transport connection is defined by the RTSP URI (Section
1533
3.2). For the scheme "rtsp", a persistent connection is assumed,
1534
while the scheme "rtspu" calls for RTSP requests to be sent without
1535
setting up a connection.
1537
Unlike HTTP, RTSP allows the media server to send requests to the
1538
media client. However, this is only supported for persistent
1539
connections, as the media server otherwise has no reliable way of
1540
reaching the client. Also, this is the only way that requests from
1541
media server to client are likely to traverse firewalls.
1545
A client that supports persistent connections or connectionless mode
1546
MAY "pipeline" its requests (i.e., send multiple requests without
1547
waiting for each response). A server MUST send its responses to those
1548
requests in the same order that the requests were received.
1550
9.2 Reliability and Acknowledgements
1552
Requests are acknowledged by the receiver unless they are sent to a
1553
multicast group. If there is no acknowledgement, the sender may
1554
resend the same message after a timeout of one round-trip time (RTT).
1555
The round-trip time is estimated as in TCP (RFC 1123) [18], with an
1556
initial round-trip value of 500 ms. An implementation MAY cache the
1557
last RTT measurement as the initial value for future connections.
1559
If a reliable transport protocol is used to carry RTSP, requests MUST
1560
NOT be retransmitted; the RTSP application MUST instead rely on the
1561
underlying transport to provide reliability.
1563
If both the underlying reliable transport such as TCP and the RTSP
1564
application retransmit requests, it is possible that each packet
1565
loss results in two retransmissions. The receiver cannot typically
1566
take advantage of the application-layer retransmission since the
1570
Schulzrinne, et. al. Standards Track [Page 28]
1572
RFC 2326 Real Time Streaming Protocol April 1998
1575
transport stack will not deliver the application-layer
1576
retransmission before the first attempt has reached the receiver.
1577
If the packet loss is caused by congestion, multiple
1578
retransmissions at different layers will exacerbate the congestion.
1580
If RTSP is used over a small-RTT LAN, standard procedures for
1581
optimizing initial TCP round trip estimates, such as those used in
1582
T/TCP (RFC 1644) [22], can be beneficial.
1584
The Timestamp header (Section 12.38) is used to avoid the
1585
retransmission ambiguity problem [23, p. 301] and obviates the need
1586
for Karn's algorithm.
1588
Each request carries a sequence number in the CSeq header (Section
1589
12.17), which is incremented by one for each distinct request
1590
transmitted. If a request is repeated because of lack of
1591
acknowledgement, the request MUST carry the original sequence number
1592
(i.e., the sequence number is not incremented).
1594
Systems implementing RTSP MUST support carrying RTSP over TCP and MAY
1595
support UDP. The default port for the RTSP server is 554 for both UDP
1598
A number of RTSP packets destined for the same control end point may
1599
be packed into a single lower-layer PDU or encapsulated into a TCP
1600
stream. RTSP data MAY be interleaved with RTP and RTCP packets.
1601
Unlike HTTP, an RTSP message MUST contain a Content-Length header
1602
whenever that message contains a payload. Otherwise, an RTSP packet
1603
is terminated with an empty line immediately following the last
1606
10 Method Definitions
1608
The method token indicates the method to be performed on the resource
1609
identified by the Request-URI. The method is case-sensitive. New
1610
methods may be defined in the future. Method names may not start with
1611
a $ character (decimal 24) and must be a token. Methods are
1612
summarized in Table 2.
1626
Schulzrinne, et. al. Standards Track [Page 29]
1628
RFC 2326 Real Time Streaming Protocol April 1998
1631
method direction object requirement
1632
DESCRIBE C->S P,S recommended
1633
ANNOUNCE C->S, S->C P,S optional
1634
GET_PARAMETER C->S, S->C P,S optional
1635
OPTIONS C->S, S->C P,S required
1637
PAUSE C->S P,S recommended
1638
PLAY C->S P,S required
1639
RECORD C->S P,S optional
1640
REDIRECT S->C P,S optional
1641
SETUP C->S S required
1642
SET_PARAMETER C->S, S->C P,S optional
1643
TEARDOWN C->S P,S required
1645
Table 2: Overview of RTSP methods, their direction, and what
1646
objects (P: presentation, S: stream) they operate on
1648
Notes on Table 2: PAUSE is recommended, but not required in that a
1649
fully functional server can be built that does not support this
1650
method, for example, for live feeds. If a server does not support a
1651
particular method, it MUST return "501 Not Implemented" and a client
1652
SHOULD not try this method again for this server.
1656
The behavior is equivalent to that described in [H9.2]. An OPTIONS
1657
request may be issued at any time, e.g., if the client is about to
1658
try a nonstandard request. It does not influence server state.
1662
C->S: OPTIONS * RTSP/1.0
1664
Require: implicit-play
1665
Proxy-Require: gzipped-messages
1667
S->C: RTSP/1.0 200 OK
1669
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
1671
Note that these are necessarily fictional features (one would hope
1672
that we would not purposefully overlook a truly useful feature just
1673
so that we could have a strong example in this section).
1682
Schulzrinne, et. al. Standards Track [Page 30]
1684
RFC 2326 Real Time Streaming Protocol April 1998
1689
The DESCRIBE method retrieves the description of a presentation or
1690
media object identified by the request URL from a server. It may use
1691
the Accept header to specify the description formats that the client
1692
understands. The server responds with a description of the requested
1693
resource. The DESCRIBE reply-response pair constitutes the media
1694
initialization phase of RTSP.
1698
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
1700
Accept: application/sdp, application/rtsl, application/mheg
1702
S->C: RTSP/1.0 200 OK
1704
Date: 23 Jan 1997 15:35:06 GMT
1705
Content-Type: application/sdp
1709
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
1711
i=A Seminar on the session description protocol
1712
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
1713
e=mjh@isi.edu (Mark Handley)
1714
c=IN IP4 224.2.17.12/127
1715
t=2873397496 2873404696
1717
m=audio 3456 RTP/AVP 0
1718
m=video 2232 RTP/AVP 31
1719
m=whiteboard 32416 UDP WB
1722
The DESCRIBE response MUST contain all media initialization
1723
information for the resource(s) that it describes. If a media client
1724
obtains a presentation description from a source other than DESCRIBE
1725
and that description contains a complete set of media initialization
1726
parameters, the client SHOULD use those parameters and not then
1727
request a description for the same media via RTSP.
1729
Additionally, servers SHOULD NOT use the DESCRIBE response as a means
1730
of media indirection.
1732
Clear ground rules need to be established so that clients have an
1733
unambiguous means of knowing when to request media initialization
1734
information via DESCRIBE, and when not to. By forcing a DESCRIBE
1738
Schulzrinne, et. al. Standards Track [Page 31]
1740
RFC 2326 Real Time Streaming Protocol April 1998
1743
response to contain all media initialization for the set of streams
1744
that it describes, and discouraging use of DESCRIBE for media
1745
indirection, we avoid looping problems that might result from other
1748
Media initialization is a requirement for any RTSP-based system,
1749
but the RTSP specification does not dictate that this must be done
1750
via the DESCRIBE method. There are three ways that an RTSP client
1751
may receive initialization information:
1753
* via RTSP's DESCRIBE method;
1754
* via some other protocol (HTTP, email attachment, etc.);
1755
* via the command line or standard input (thus working as a browser
1756
helper application launched with an SDP file or other media
1757
initialization format).
1759
In the interest of practical interoperability, it is highly
1760
recommended that minimal servers support the DESCRIBE method, and
1761
highly recommended that minimal clients support the ability to act
1762
as a "helper application" that accepts a media initialization file
1763
from standard input, command line, and/or other means that are
1764
appropriate to the operating environment of the client.
1768
The ANNOUNCE method serves two purposes:
1770
When sent from client to server, ANNOUNCE posts the description of a
1771
presentation or media object identified by the request URL to a
1772
server. When sent from server to client, ANNOUNCE updates the session
1773
description in real-time.
1775
If a new media stream is added to a presentation (e.g., during a live
1776
presentation), the whole presentation description should be sent
1777
again, rather than just the additional components, so that components
1782
C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
1784
Date: 23 Jan 1997 15:35:06 GMT
1786
Content-Type: application/sdp
1790
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
1794
Schulzrinne, et. al. Standards Track [Page 32]
1796
RFC 2326 Real Time Streaming Protocol April 1998
1800
i=A Seminar on the session description protocol
1801
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
1802
e=mjh@isi.edu (Mark Handley)
1803
c=IN IP4 224.2.17.12/127
1804
t=2873397496 2873404696
1806
m=audio 3456 RTP/AVP 0
1807
m=video 2232 RTP/AVP 31
1809
S->C: RTSP/1.0 200 OK
1814
The SETUP request for a URI specifies the transport mechanism to be
1815
used for the streamed media. A client can issue a SETUP request for a
1816
stream that is already playing to change transport parameters, which
1817
a server MAY allow. If it does not allow this, it MUST respond with
1818
error "455 Method Not Valid In This State". For the benefit of any
1819
intervening firewalls, a client must indicate the transport
1820
parameters even if it has no influence over these parameters, for
1821
example, where the server advertises a fixed multicast address.
1823
Since SETUP includes all transport initialization information,
1824
firewalls and other intermediate network devices (which need this
1825
information) are spared the more arduous task of parsing the
1826
DESCRIBE response, which has been reserved for media
1829
The Transport header specifies the transport parameters acceptable to
1830
the client for data transmission; the response will contain the
1831
transport parameters selected by the server.
1833
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
1835
Transport: RTP/AVP;unicast;client_port=4588-4589
1837
S->C: RTSP/1.0 200 OK
1839
Date: 23 Jan 1997 15:35:06 GMT
1841
Transport: RTP/AVP;unicast;
1842
client_port=4588-4589;server_port=6256-6257
1844
The server generates session identifiers in response to SETUP
1845
requests. If a SETUP request to a server includes a session
1846
identifier, the server MUST bundle this setup request into the
1850
Schulzrinne, et. al. Standards Track [Page 33]
1852
RFC 2326 Real Time Streaming Protocol April 1998
1855
existing session or return error "459 Aggregate Operation Not
1856
Allowed" (see Section 11.3.10).
1860
The PLAY method tells the server to start sending data via the
1861
mechanism specified in SETUP. A client MUST NOT issue a PLAY request
1862
until any outstanding SETUP requests have been acknowledged as
1865
The PLAY request positions the normal play time to the beginning of
1866
the range specified and delivers stream data until the end of the
1867
range is reached. PLAY requests may be pipelined (queued); a server
1868
MUST queue PLAY requests to be executed in order. That is, a PLAY
1869
request arriving while a previous PLAY request is still active is
1870
delayed until the first has been completed.
1872
This allows precise editing.
1874
For example, regardless of how closely spaced the two PLAY requests
1875
in the example below arrive, the server will first play seconds 10
1876
through 15, then, immediately following, seconds 20 to 25, and
1877
finally seconds 30 through the end.
1879
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
1884
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
1889
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
1894
See the description of the PAUSE request for further examples.
1896
A PLAY request without a Range header is legal. It starts playing a
1897
stream from the beginning unless the stream has been paused. If a
1898
stream has been paused via PAUSE, stream delivery resumes at the
1899
pause point. If a stream is playing, such a PLAY request causes no
1900
further action and can be used by the client to test server liveness.
1906
Schulzrinne, et. al. Standards Track [Page 34]
1908
RFC 2326 Real Time Streaming Protocol April 1998
1911
The Range header may also contain a time parameter. This parameter
1912
specifies a time in UTC at which the playback should start. If the
1913
message is received after the specified time, playback is started
1914
immediately. The time parameter may be used to aid in synchronization
1915
of streams obtained from different sources.
1917
For a on-demand stream, the server replies with the actual range that
1918
will be played back. This may differ from the requested range if
1919
alignment of the requested range to valid frame boundaries is
1920
required for the media source. If no range is specified in the
1921
request, the current position is returned in the reply. The unit of
1922
the range in the reply is the same as that in the request.
1924
After playing the desired range, the presentation is automatically
1925
paused, as if a PAUSE request had been issued.
1927
The following example plays the whole presentation starting at SMPTE
1928
time code 0:10:20 until the end of the clip. The playback is to start
1929
at 15:36 on 23 Jan 1997.
1931
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
1934
Range: smpte=0:10:20-;time=19970123T153600Z
1936
S->C: RTSP/1.0 200 OK
1938
Date: 23 Jan 1997 15:35:06 GMT
1939
Range: smpte=0:10:22-;time=19970123T153600Z
1941
For playing back a recording of a live presentation, it may be
1942
desirable to use clock units:
1944
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
1947
Range: clock=19961108T142300Z-19961108T143520Z
1949
S->C: RTSP/1.0 200 OK
1951
Date: 23 Jan 1997 15:35:06 GMT
1953
A media server only supporting playback MUST support the npt format
1954
and MAY support the clock and smpte formats.
1962
Schulzrinne, et. al. Standards Track [Page 35]
1964
RFC 2326 Real Time Streaming Protocol April 1998
1969
The PAUSE request causes the stream delivery to be interrupted
1970
(halted) temporarily. If the request URL names a stream, only
1971
playback and recording of that stream is halted. For example, for
1972
audio, this is equivalent to muting. If the request URL names a
1973
presentation or group of streams, delivery of all currently active
1974
streams within the presentation or group is halted. After resuming
1975
playback or recording, synchronization of the tracks MUST be
1976
maintained. Any server resources are kept, though servers MAY close
1977
the session and free resources after being paused for the duration
1978
specified with the timeout parameter of the Session header in the
1983
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
1987
S->C: RTSP/1.0 200 OK
1989
Date: 23 Jan 1997 15:35:06 GMT
1991
The PAUSE request may contain a Range header specifying when the
1992
stream or presentation is to be halted. We refer to this point as the
1993
"pause point". The header must contain exactly one value rather than
1994
a time range. The normal play time for the stream is set to the pause
1995
point. The pause request becomes effective the first time the server
1996
is encountering the time point specified in any of the currently
1997
pending PLAY requests. If the Range header specifies a time outside
1998
any currently pending PLAY requests, the error "457 Invalid Range" is
1999
returned. If a media unit (such as an audio or video frame) starts
2000
presentation at exactly the pause point, it is not played or
2001
recorded. If the Range header is missing, stream delivery is
2002
interrupted immediately on receipt of the message and the pause point
2003
is set to the current normal play time.
2005
A PAUSE request discards all queued PLAY requests. However, the pause
2006
point in the media stream MUST be maintained. A subsequent PLAY
2007
request without Range header resumes from the pause point.
2009
For example, if the server has play requests for ranges 10 to 15 and
2010
20 to 29 pending and then receives a pause request for NPT 21, it
2011
would start playing the second range and stop at NPT 21. If the pause
2012
request is for NPT 12 and the server is playing at NPT 13 serving the
2013
first play request, the server stops immediately. If the pause
2014
request is for NPT 16, the server stops after completing the first
2018
Schulzrinne, et. al. Standards Track [Page 36]
2020
RFC 2326 Real Time Streaming Protocol April 1998
2023
play request and discards the second play request.
2025
As another example, if a server has received requests to play ranges
2026
10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
2027
request for NPT=14 would take effect while the server plays the first
2028
range, with the second PLAY request effectively being ignored,
2029
assuming the PAUSE request arrives before the server has started
2030
playing the second, overlapping range. Regardless of when the PAUSE
2031
request arrives, it sets the NPT to 14.
2033
If the server has already sent data beyond the time specified in the
2034
Range header, a PLAY would still resume at that point in time, as it
2035
is assumed that the client has discarded data after that point. This
2036
ensures continuous pause/play cycling without gaps.
2040
The TEARDOWN request stops the stream delivery for the given URI,
2041
freeing the resources associated with it. If the URI is the
2042
presentation URI for this presentation, any RTSP session identifier
2043
associated with the session is no longer valid. Unless all transport
2044
parameters are defined by the session description, a SETUP request
2045
has to be issued before the session can be played again.
2048
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
2051
S->C: RTSP/1.0 200 OK
2056
The GET_PARAMETER request retrieves the value of a parameter of a
2057
presentation or stream specified in the URI. The content of the reply
2058
and response is left to the implementation. GET_PARAMETER with no
2059
entity body may be used to test client or server liveness ("ping").
2063
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
2065
Content-Type: text/parameters
2074
Schulzrinne, et. al. Standards Track [Page 37]
2076
RFC 2326 Real Time Streaming Protocol April 1998
2079
C->S: RTSP/1.0 200 OK
2082
Content-Type: text/parameters
2084
packets_received: 10
2087
The "text/parameters" section is only an example type for
2088
parameter. This method is intentionally loosely defined with the
2089
intention that the reply content and response content will be
2090
defined after further experimentation.
2094
This method requests to set the value of a parameter for a
2095
presentation or stream specified by the URI.
2097
A request SHOULD only contain a single parameter to allow the client
2098
to determine why a particular request failed. If the request contains
2099
several parameters, the server MUST only act on the request if all of
2100
the parameters can be set successfully. A server MUST allow a
2101
parameter to be set repeatedly to the same value, but it MAY disallow
2102
changing parameter values.
2104
Note: transport parameters for the media stream MUST only be set with
2107
Restricting setting transport parameters to SETUP is for the
2108
benefit of firewalls.
2110
The parameters are split in a fine-grained fashion so that there
2111
can be more meaningful error indications. However, it may make
2112
sense to allow the setting of several parameters if an atomic
2113
setting is desirable. Imagine device control where the client does
2114
not want the camera to pan unless it can also tilt to the right
2115
angle at the same time.
2119
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
2122
Content-type: text/parameters
2126
S->C: RTSP/1.0 451 Invalid Parameter
2130
Schulzrinne, et. al. Standards Track [Page 38]
2132
RFC 2326 Real Time Streaming Protocol April 1998
2137
Content-type: text/parameters
2141
The "text/parameters" section is only an example type for
2142
parameter. This method is intentionally loosely defined with the
2143
intention that the reply content and response content will be
2144
defined after further experimentation.
2148
A redirect request informs the client that it must connect to another
2149
server location. It contains the mandatory header Location, which
2150
indicates that the client should issue requests for that URL. It may
2151
contain the parameter Range, which indicates when the redirection
2152
takes effect. If the client wants to continue to send or receive
2153
media for this URI, the client MUST issue a TEARDOWN request for the
2154
current session and a SETUP for the new session at the designated
2157
This example request redirects traffic for this URI to the new server
2158
at the given play time:
2160
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
2162
Location: rtsp://bigserver.com:8001
2163
Range: clock=19960213T143205Z-
2167
This method initiates recording a range of media data according to
2168
the presentation description. The timestamp reflects start and end
2169
time (UTC). If no time range is given, use the start or end time
2170
provided in the presentation description. If the session has already
2171
started, commence recording immediately.
2173
The server decides whether to store the recorded data under the
2174
request-URI or another URI. If the server does not use the request-
2175
URI, the response SHOULD be 201 (Created) and contain an entity which
2176
describes the status of the request and refers to the new resource,
2177
and a Location header.
2179
A media server supporting recording of live presentations MUST
2180
support the clock range format; the smpte format does not make sense.
2186
Schulzrinne, et. al. Standards Track [Page 39]
2188
RFC 2326 Real Time Streaming Protocol April 1998
2191
In this example, the media server was previously invited to the
2192
conference indicated.
2194
C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
2197
Conference: 128.16.64.19/32492374
2199
10.12 Embedded (Interleaved) Binary Data
2201
Certain firewall designs and other circumstances may force a server
2202
to interleave RTSP methods and stream data. This interleaving should
2203
generally be avoided unless necessary since it complicates client and
2204
server operation and imposes additional overhead. Interleaved binary
2205
data SHOULD only be used if RTSP is carried over TCP.
2207
Stream data such as RTP packets is encapsulated by an ASCII dollar
2208
sign (24 hexadecimal), followed by a one-byte channel identifier,
2209
followed by the length of the encapsulated binary data as a binary,
2210
two-byte integer in network byte order. The stream data follows
2211
immediately afterwards, without a CRLF, but including the upper-layer
2212
protocol headers. Each $ block contains exactly one upper-layer
2213
protocol data unit, e.g., one RTP packet.
2215
The channel identifier is defined in the Transport header with the
2216
interleaved parameter(Section 12.39).
2218
When the transport choice is RTP, RTCP messages are also interleaved
2219
by the server over the TCP connection. As a default, RTCP packets are
2220
sent on the first available channel higher than the RTP channel. The
2221
client MAY explicitly request RTCP packets on another channel. This
2222
is done by specifying two channels in the interleaved parameter of
2223
the Transport header(Section 12.39).
2225
RTCP is needed for synchronization when two or more streams are
2226
interleaved in such a fashion. Also, this provides a convenient way
2227
to tunnel RTP/RTCP packets through the TCP control connection when
2228
required by the network configuration and transfer them onto UDP
2231
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
2233
Transport: RTP/AVP/TCP;interleaved=0-1
2235
S->C: RTSP/1.0 200 OK
2237
Date: 05 Jun 1997 18:57:18 GMT
2238
Transport: RTP/AVP/TCP;interleaved=0-1
2242
Schulzrinne, et. al. Standards Track [Page 40]
2244
RFC 2326 Real Time Streaming Protocol April 1998
2249
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
2253
S->C: RTSP/1.0 200 OK
2256
Date: 05 Jun 1997 18:59:15 GMT
2257
RTP-Info: url=rtsp://foo.com/bar.file;
2258
seq=232433;rtptime=972948234
2260
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
2261
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
2262
S->C: $\001{2 byte length}{"length" bytes RTCP packet}
2264
11 Status Code Definitions
2266
Where applicable, HTTP status [H10] codes are reused. Status codes
2267
that have the same meaning are not repeated here. See Table 1 for a
2268
listing of which status codes may be returned by which requests.
2272
11.1.1 250 Low on Storage Space
2274
The server returns this warning after receiving a RECORD request that
2275
it may not be able to fulfill completely due to insufficient storage
2276
space. If possible, the server should use the Range header to
2277
indicate what time period it may still be able to record. Since other
2278
processes on the server may be consuming storage space
2279
simultaneously, a client should take this only as an estimate.
2281
11.2 Redirection 3xx
2285
Within RTSP, redirection may be used for load balancing or
2286
redirecting stream requests to a server topologically closer to the
2287
client. Mechanisms to determine topological proximity are beyond the
2288
scope of this specification.
2298
Schulzrinne, et. al. Standards Track [Page 41]
2300
RFC 2326 Real Time Streaming Protocol April 1998
2303
11.3 Client Error 4xx
2305
11.3.1 405 Method Not Allowed
2307
The method specified in the request is not allowed for the resource
2308
identified by the request URI. The response MUST include an Allow
2309
header containing a list of valid methods for the requested resource.
2310
This status code is also to be used if a request attempts to use a
2311
method not indicated during SETUP, e.g., if a RECORD request is
2312
issued even though the mode parameter in the Transport header only
2315
11.3.2 451 Parameter Not Understood
2317
The recipient of the request does not support one or more parameters
2318
contained in the request.
2320
11.3.3 452 Conference Not Found
2322
The conference indicated by a Conference header field is unknown to
2325
11.3.4 453 Not Enough Bandwidth
2327
The request was refused because there was insufficient bandwidth.
2328
This may, for example, be the result of a resource reservation
2331
11.3.5 454 Session Not Found
2333
The RTSP session identifier in the Session header is missing,
2334
invalid, or has timed out.
2336
11.3.6 455 Method Not Valid in This State
2338
The client or server cannot process this request in its current
2339
state. The response SHOULD contain an Allow header to make error
2342
11.3.7 456 Header Field Not Valid for Resource
2344
The server could not act on a required request header. For example,
2345
if PLAY contains the Range header field but the stream does not allow
2354
Schulzrinne, et. al. Standards Track [Page 42]
2356
RFC 2326 Real Time Streaming Protocol April 1998
2359
11.3.8 457 Invalid Range
2361
The Range value given is out of bounds, e.g., beyond the end of the
2364
11.3.9 458 Parameter Is Read-Only
2366
The parameter to be set by SET_PARAMETER can be read but not
2369
11.3.10 459 Aggregate Operation Not Allowed
2371
The requested method may not be applied on the URL in question since
2372
it is an aggregate (presentation) URL. The method may be applied on a
2375
11.3.11 460 Only Aggregate Operation Allowed
2377
The requested method may not be applied on the URL in question since
2378
it is not an aggregate (presentation) URL. The method may be applied
2379
on the presentation URL.
2381
11.3.12 461 Unsupported Transport
2383
The Transport field did not contain a supported transport
2386
11.3.13 462 Destination Unreachable
2388
The data transmission channel could not be established because the
2389
client address could not be reached. This error will most likely be
2390
the result of a client attempt to place an invalid Destination
2391
parameter in the Transport field.
2393
11.3.14 551 Option not supported
2395
An option given in the Require or the Proxy-Require fields was not
2396
supported. The Unsupported header should be returned stating the
2397
option for which there is no support.
2410
Schulzrinne, et. al. Standards Track [Page 43]
2412
RFC 2326 Real Time Streaming Protocol April 1998
2415
12 Header Field Definitions
2417
HTTP/1.1 [2] or other, non-standard header fields not listed here
2418
currently have no well-defined meaning and SHOULD be ignored by the
2421
Table 3 summarizes the header fields used by RTSP. Type "g"
2422
designates general request headers to be found in both requests and
2423
responses, type "R" designates request headers, type "r" designates
2424
response headers, and type "e" designates entity header fields.
2425
Fields marked with "req." in the column labeled "support" MUST be
2426
implemented by the recipient for a particular method, while fields
2427
marked "opt." are optional. Note that not all fields marked "req."
2428
will be sent in every request of this type. The "req." means only
2429
that client (for response headers) and server (for request headers)
2430
MUST implement the fields. The last column lists the method for which
2431
this header field is meaningful; the designation "entity" refers to
2432
all methods that return a message body. Within this specification,
2433
DESCRIBE and GET_PARAMETER fall into this class.
2466
Schulzrinne, et. al. Standards Track [Page 44]
2468
RFC 2326 Real Time Streaming Protocol April 1998
2471
Header type support methods
2472
Accept R opt. entity
2473
Accept-Encoding R opt. entity
2474
Accept-Language R opt. all
2476
Authorization R opt. all
2477
Bandwidth R opt. all
2478
Blocksize R opt. all but OPTIONS, TEARDOWN
2479
Cache-Control g opt. SETUP
2480
Conference R opt. SETUP
2481
Connection g req. all
2482
Content-Base e opt. entity
2483
Content-Encoding e req. SET_PARAMETER
2484
Content-Encoding e req. DESCRIBE, ANNOUNCE
2485
Content-Language e req. DESCRIBE, ANNOUNCE
2486
Content-Length e req. SET_PARAMETER, ANNOUNCE
2487
Content-Length e req. entity
2488
Content-Location e opt. entity
2489
Content-Type e req. SET_PARAMETER, ANNOUNCE
2490
Content-Type r req. entity
2493
Expires e opt. DESCRIBE, ANNOUNCE
2495
If-Modified-Since R opt. DESCRIBE, SETUP
2496
Last-Modified e opt. entity
2498
Proxy-Require R req. all
2500
Range R opt. PLAY, PAUSE, RECORD
2501
Range r opt. PLAY, PAUSE, RECORD
2504
Retry-After r opt. all
2505
RTP-Info r req. PLAY
2506
Scale Rr opt. PLAY, RECORD
2507
Session Rr req. all but SETUP, OPTIONS
2510
Transport Rr req. SETUP
2511
Unsupported r req. all
2512
User-Agent R opt. all
2514
WWW-Authenticate r opt. all
2522
Schulzrinne, et. al. Standards Track [Page 45]
2524
RFC 2326 Real Time Streaming Protocol April 1998
2527
Overview of RTSP header fields
2531
The Accept request-header field can be used to specify certain
2532
presentation description content types which are acceptable for the
2535
The "level" parameter for presentation descriptions is properly
2536
defined as part of the MIME type registration, not here.
2538
See [H14.1] for syntax.
2541
Accept: application/rtsl, application/sdp;level=2
2543
12.2 Accept-Encoding
2547
12.3 Accept-Language
2549
See [H14.4]. Note that the language specified applies to the
2550
presentation description and any reason phrases, not the media
2555
The Allow response header field lists the methods supported by the
2556
resource identified by the request-URI. The purpose of this field is
2557
to strictly inform the recipient of valid methods associated with the
2558
resource. An Allow header field must be present in a 405 (Method not
2562
Allow: SETUP, PLAY, RECORD, SET_PARAMETER
2570
The Bandwidth request header field describes the estimated bandwidth
2571
available to the client, expressed as a positive integer and measured
2572
in bits per second. The bandwidth available to the client may change
2573
during an RTSP session, e.g., due to modem retraining.
2578
Schulzrinne, et. al. Standards Track [Page 46]
2580
RFC 2326 Real Time Streaming Protocol April 1998
2583
Bandwidth = "Bandwidth" ":" 1*DIGIT
2590
This request header field is sent from the client to the media server
2591
asking the server for a particular media packet size. This packet
2592
size does not include lower-layer headers such as IP, UDP, or RTP.
2593
The server is free to use a blocksize which is lower than the one
2594
requested. The server MAY truncate this packet size to the closest
2595
multiple of the minimum, media-specific block size, or override it
2596
with the media-specific size if necessary. The block size MUST be a
2597
positive decimal number, measured in octets. The server only returns
2598
an error (416) if the value is syntactically invalid.
2602
The Cache-Control general header field is used to specify directives
2603
that MUST be obeyed by all caching mechanisms along the
2604
request/response chain.
2606
Cache directives must be passed through by a proxy or gateway
2607
application, regardless of their significance to that application,
2608
since the directives may be applicable to all recipients along the
2609
request/response chain. It is not possible to specify a cache-
2610
directive for a specific cache.
2612
Cache-Control should only be specified in a SETUP request and its
2613
response. Note: Cache-Control does not govern the caching of
2614
responses as for HTTP, but rather of the stream identified by the
2615
SETUP request. Responses to RTSP requests are not cacheable, except
2616
for responses to DESCRIBE.
2618
Cache-Control = "Cache-Control" ":" 1#cache-directive
2619
cache-directive = cache-request-directive
2620
| cache-response-directive
2621
cache-request-directive = "no-cache"
2626
cache-response-directive = "public"
2634
Schulzrinne, et. al. Standards Track [Page 47]
2636
RFC 2326 Real Time Streaming Protocol April 1998
2639
| "proxy-revalidate"
2640
| "max-age" "=" delta-seconds
2642
cache-extension = token [ "=" ( token | quoted-string ) ]
2645
Indicates that the media stream MUST NOT be cached anywhere.
2646
This allows an origin server to prevent caching even by caches
2647
that have been configured to return stale responses to client
2651
Indicates that the media stream is cacheable by any cache.
2654
Indicates that the media stream is intended for a single user
2655
and MUST NOT be cached by a shared cache. A private (non-
2656
shared) cache may cache the media stream.
2659
An intermediate cache (proxy) may find it useful to convert
2660
the media type of a certain stream. A proxy might, for
2661
example, convert between video formats to save cache space or
2662
to reduce the amount of traffic on a slow link. Serious
2663
operational problems may occur, however, when these
2664
transformations have been applied to streams intended for
2665
certain kinds of applications. For example, applications for
2666
medical imaging, scientific data analysis and those using
2667
end-to-end authentication all depend on receiving a stream
2668
that is bit-for-bit identical to the original entity-body.
2669
Therefore, if a response includes the no-transform directive,
2670
an intermediate cache or proxy MUST NOT change the encoding of
2671
the stream. Unlike HTTP, RTSP does not provide for partial
2672
transformation at this point, e.g., allowing translation into
2673
a different language.
2676
In some cases, such as times of extremely poor network
2677
connectivity, a client may want a cache to return only those
2678
media streams that it currently has stored, and not to receive
2679
these from the origin server. To do this, the client may
2680
include the only-if-cached directive in a request. If it
2681
receives this directive, a cache SHOULD either respond using a
2682
cached media stream that is consistent with the other
2683
constraints of the request, or respond with a 504 (Gateway
2684
Timeout) status. However, if a group of caches is being
2685
operated as a unified system with good internal connectivity,
2686
such a request MAY be forwarded within that group of caches.
2690
Schulzrinne, et. al. Standards Track [Page 48]
2692
RFC 2326 Real Time Streaming Protocol April 1998
2696
Indicates that the client is willing to accept a media stream
2697
that has exceeded its expiration time. If max-stale is
2698
assigned a value, then the client is willing to accept a
2699
response that has exceeded its expiration time by no more than
2700
the specified number of seconds. If no value is assigned to
2701
max-stale, then the client is willing to accept a stale
2702
response of any age.
2705
Indicates that the client is willing to accept a media stream
2706
whose freshness lifetime is no less than its current age plus
2707
the specified time in seconds. That is, the client wants a
2708
response that will still be fresh for at least the specified
2712
When the must-revalidate directive is present in a SETUP
2713
response received by a cache, that cache MUST NOT use the
2714
entry after it becomes stale to respond to a subsequent
2715
request without first revalidating it with the origin server.
2716
That is, the cache must do an end-to-end revalidation every
2717
time, if, based solely on the origin server's Expires, the
2718
cached response is stale.)
2722
This request header field establishes a logical connection between a
2723
pre-established conference and an RTSP stream. The conference-id must
2724
not be changed for the same RTSP session.
2726
Conference = "Conference" ":" conference-id Example:
2727
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
2729
A response code of 452 (452 Conference Not Found) is returned if the
2730
conference-id is not valid.
2740
12.12 Content-Encoding
2746
Schulzrinne, et. al. Standards Track [Page 49]
2748
RFC 2326 Real Time Streaming Protocol April 1998
2751
12.13 Content-Language
2755
12.14 Content-Length
2757
This field contains the length of the content of the method (i.e.
2758
after the double CRLF following the last header). Unlike HTTP, it
2759
MUST be included in all messages that carry content beyond the header
2760
portion of the message. If it is missing, a default value of zero is
2761
assumed. It is interpreted according to [H14.14].
2763
12.15 Content-Location
2769
See [H14.18]. Note that the content types suitable for RTSP are
2770
likely to be restricted in practice to presentation descriptions and
2771
parameter-value types.
2775
The CSeq field specifies the sequence number for an RTSP request-
2776
response pair. This field MUST be present in all requests and
2777
responses. For every RTSP request containing the given sequence
2778
number, there will be a corresponding response having the same
2779
number. Any retransmitted request must contain the same sequence
2780
number as the original (i.e. the sequence number is not incremented
2781
for retransmissions of the same request).
2789
The Expires entity-header field gives a date and time after which the
2790
description or media-stream should be considered stale. The
2791
interpretation depends on the method:
2794
The Expires header indicates a date and time after which the
2795
description should be considered stale.
2802
Schulzrinne, et. al. Standards Track [Page 50]
2804
RFC 2326 Real Time Streaming Protocol April 1998
2807
A stale cache entry may not normally be returned by a cache (either a
2808
proxy cache or an user agent cache) unless it is first validated with
2809
the origin server (or with an intermediate cache that has a fresh
2810
copy of the entity). See section 13 for further discussion of the
2813
The presence of an Expires field does not imply that the original
2814
resource will change or cease to exist at, before, or after that
2817
The format is an absolute date and time as defined by HTTP-date in
2818
[H3.3]; it MUST be in RFC1123-date format:
2820
Expires = "Expires" ":" HTTP-date
2822
An example of its use is
2824
Expires: Thu, 01 Dec 1994 16:00:00 GMT
2826
RTSP/1.0 clients and caches MUST treat other invalid date formats,
2827
especially including the value "0", as having occurred in the past
2828
(i.e., "already expired").
2830
To mark a response as "already expired," an origin server should use
2831
an Expires date that is equal to the Date header value. To mark a
2832
response as "never expires," an origin server should use an Expires
2833
date approximately one year from the time the response is sent.
2834
RTSP/1.0 servers should not send Expires dates more than one year in
2837
The presence of an Expires header field with a date value of some
2838
time in the future on a media stream that otherwise would by default
2839
be non-cacheable indicates that the media stream is cacheable, unless
2840
indicated otherwise by a Cache-Control header field (Section 12.8).
2848
This HTTP request header field is not needed for RTSP. It should be
2849
silently ignored if sent.
2858
Schulzrinne, et. al. Standards Track [Page 51]
2860
RFC 2326 Real Time Streaming Protocol April 1998
2863
This field is especially useful for ensuring the integrity of the
2864
presentation description, in both the case where it is fetched via
2865
means external to RTSP (such as HTTP), or in the case where the
2866
server implementation is guaranteeing the integrity of the
2867
description between the time of the DESCRIBE message and the SETUP
2870
The identifier is an opaque identifier, and thus is not specific to
2871
any particular session description language.
2873
12.23 If-Modified-Since
2875
The If-Modified-Since request-header field is used with the DESCRIBE
2876
and SETUP methods to make them conditional. If the requested variant
2877
has not been modified since the time specified in this field, a
2878
description will not be returned from the server (DESCRIBE) or a
2879
stream will not be set up (SETUP). Instead, a 304 (not modified)
2880
response will be returned without any message-body.
2882
If-Modified-Since = "If-Modified-Since" ":" HTTP-date
2884
An example of the field is:
2886
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
2890
The Last-Modified entity-header field indicates the date and time at
2891
which the origin server believes the presentation description or
2892
media stream was last modified. See [H14.29]. For the methods
2893
DESCRIBE or ANNOUNCE, the header field indicates the last
2894
modification date and time of the description, for SETUP that of the
2901
12.26 Proxy-Authenticate
2907
The Proxy-Require header is used to indicate proxy-sensitive features
2908
that MUST be supported by the proxy. Any Proxy-Require header
2909
features that are not supported by the proxy MUST be negatively
2910
acknowledged by the proxy to the client if not supported. Servers
2914
Schulzrinne, et. al. Standards Track [Page 52]
2916
RFC 2326 Real Time Streaming Protocol April 1998
2919
should treat this field identically to the Require field.
2921
See Section 12.32 for more details on the mechanics of this message
2922
and a usage example.
2930
This request and response header field specifies a range of time.
2931
The range can be specified in a number of units. This specification
2932
defines the smpte (Section 3.5), npt (Section 3.6), and clock
2933
(Section 3.7) range units. Within RTSP, byte ranges [H14.36.1] are
2934
not meaningful and MUST NOT be used. The header may also contain a
2935
time parameter in UTC, specifying the time at which the operation is
2936
to be made effective. Servers supporting the Range header MUST
2937
understand the NPT range format and SHOULD understand the SMPTE range
2938
format. The Range response header indicates what range of time is
2939
actually being played or recorded. If the Range header is given in a
2940
time format that is not understood, the recipient should return "501
2943
Ranges are half-open intervals, including the lower point, but
2944
excluding the upper point. In other words, a range of a-b starts
2945
exactly at time a, but stops just before b. Only the start time of a
2946
media unit such as a video or audio frame is relevant. As an example,
2947
assume that video frames are generated every 40 ms. A range of 10.0-
2948
10.1 would include a video frame starting at 10.0 or later time and
2949
would include a video frame starting at 10.08, even though it lasted
2950
beyond the interval. A range of 10.0-10.08, on the other hand, would
2951
exclude the frame at 10.08.
2953
Range = "Range" ":" 1\#ranges-specifier
2954
[ ";" "time" "=" utc-time ]
2955
ranges-specifier = npt-range | utc-range | smpte-range
2958
Range: clock=19960213T143205Z-;time=19970123T143720Z
2960
The notation is similar to that used for the HTTP/1.1 [2] byte-
2961
range header. It allows clients to select an excerpt from the media
2962
object, and to play from a given point to the end as well as from
2963
the current location to a given point. The start of playback can be
2964
scheduled for any time in the future, although a server may refuse
2965
to keep server resources for extended idle periods.
2970
Schulzrinne, et. al. Standards Track [Page 53]
2972
RFC 2326 Real Time Streaming Protocol April 1998
2977
See [H14.37]. The URL refers to that of the presentation description,
2978
typically retrieved via HTTP.
2986
The Require header is used by clients to query the server about
2987
options that it may or may not support. The server MUST respond to
2988
this header by using the Unsupported header to negatively acknowledge
2989
those options which are NOT supported.
2991
This is to make sure that the client-server interaction will
2992
proceed without delay when all options are understood by both
2993
sides, and only slow down if options are not understood (as in the
2994
case above). For a well-matched client-server pair, the interaction
2995
proceeds quickly, saving a round-trip often required by negotiation
2996
mechanisms. In addition, it also removes state ambiguity when the
2997
client requires features that the server does not understand.
2999
Require = "Require" ":" 1#option-tag
3002
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
3004
Require: funky-feature
3005
Funky-Parameter: funkystuff
3007
S->C: RTSP/1.0 551 Option not supported
3009
Unsupported: funky-feature
3011
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
3014
S->C: RTSP/1.0 200 OK
3017
In this example, "funky-feature" is the feature tag which indicates
3018
to the client that the fictional Funky-Parameter field is required.
3019
The relationship between "funky-feature" and Funky-Parameter is not
3020
communicated via the RTSP exchange, since that relationship is an
3021
immutable property of "funky-feature" and thus should not be
3022
transmitted with every exchange.
3026
Schulzrinne, et. al. Standards Track [Page 54]
3028
RFC 2326 Real Time Streaming Protocol April 1998
3031
Proxies and other intermediary devices SHOULD ignore features that
3032
are not understood in this field. If a particular extension requires
3033
that intermediate devices support it, the extension should be tagged
3034
in the Proxy-Require field instead (see Section 12.27).
3038
This field is used to set RTP-specific parameters in the PLAY
3042
Indicates the stream URL which for which the following RTP
3043
parameters correspond.
3046
Indicates the sequence number of the first packet of the
3047
stream. This allows clients to gracefully deal with packets
3048
when seeking. The client uses this value to differentiate
3049
packets that originated before the seek from packets that
3050
originated after the seek.
3053
Indicates the RTP timestamp corresponding to the time value in
3054
the Range response header. (Note: For aggregate control, a
3055
particular stream may not actually generate a packet for the
3056
Range time value returned or implied. Thus, there is no
3057
guarantee that the packet with the sequence number indicated
3058
by seq actually has the timestamp indicated by rtptime.) The
3059
client uses this value to calculate the mapping of RTP time to
3062
A mapping from RTP timestamps to NTP timestamps (wall clock) is
3063
available via RTCP. However, this information is not sufficient to
3064
generate a mapping from RTP timestamps to NPT. Furthermore, in
3065
order to ensure that this information is available at the necessary
3066
time (immediately at startup or after a seek), and that it is
3067
delivered reliably, this mapping is placed in the RTSP control
3070
In order to compensate for drift for long, uninterrupted
3071
presentations, RTSP clients should additionally map NPT to NTP,
3072
using initial RTCP sender reports to do the mapping, and later
3073
reports to check drift against the mapping.
3082
Schulzrinne, et. al. Standards Track [Page 55]
3084
RFC 2326 Real Time Streaming Protocol April 1998
3089
RTP-Info = "RTP-Info" ":" 1#stream-url 1*parameter
3090
stream-url = "url" "=" url
3091
parameter = ";" "seq" "=" 1*DIGIT
3092
| ";" "rtptime" "=" 1*DIGIT
3096
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
3097
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
3101
A scale value of 1 indicates normal play or record at the normal
3102
forward viewing rate. If not 1, the value corresponds to the rate
3103
with respect to normal viewing rate. For example, a ratio of 2
3104
indicates twice the normal viewing rate ("fast forward") and a ratio
3105
of 0.5 indicates half the normal viewing rate. In other words, a
3106
ratio of 2 has normal play time increase at twice the wallclock rate.
3107
For every second of elapsed (wallclock) time, 2 seconds of content
3108
will be delivered. A negative value indicates reverse direction.
3110
Unless requested otherwise by the Speed parameter, the data rate
3111
SHOULD not be changed. Implementation of scale changes depends on the
3112
server and media type. For video, a server may, for example, deliver
3113
only key frames or selected key frames. For audio, it may time-scale
3114
the audio while preserving pitch or, less desirably, deliver
3117
The server should try to approximate the viewing rate, but may
3118
restrict the range of scale values that it supports. The response
3119
MUST contain the actual scale value chosen by the server.
3121
If the request contains a Range parameter, the new scale value will
3122
take effect at that time.
3124
Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
3126
Example of playing in reverse at 3.5 times normal rate:
3138
Schulzrinne, et. al. Standards Track [Page 56]
3140
RFC 2326 Real Time Streaming Protocol April 1998
3145
This request header fields parameter requests the server to deliver
3146
data to the client at a particular speed, contingent on the server's
3147
ability and desire to serve the media stream at the given speed.
3148
Implementation by the server is OPTIONAL. The default is the bit rate
3151
The parameter value is expressed as a decimal ratio, e.g., a value of
3152
2.0 indicates that data is to be delivered twice as fast as normal. A
3153
speed of zero is invalid. If the request contains a Range parameter,
3154
the new speed value will take effect at that time.
3156
Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]
3161
Use of this field changes the bandwidth used for data delivery. It is
3162
meant for use in specific circumstances where preview of the
3163
presentation at a higher or lower rate is necessary. Implementors
3164
should keep in mind that bandwidth for the session may be negotiated
3165
beforehand (by means other than RTSP), and therefore re-negotiation
3166
may be necessary. When data is delivered over UDP, it is highly
3167
recommended that means such as RTCP be used to track packet loss
3176
This request and response header field identifies an RTSP session
3177
started by the media server in a SETUP response and concluded by
3178
TEARDOWN on the presentation URL. The session identifier is chosen by
3179
the media server (see Section 3.4). Once a client receives a Session
3180
identifier, it MUST return it for any request related to that
3181
session. A server does not have to set up a session identifier if it
3182
has other means of identifying a session, such as dynamically
3185
Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
3187
The timeout parameter is only allowed in a response header. The
3188
server uses it to indicate to the client how long the server is
3189
prepared to wait between RTSP commands before closing the session due
3190
to lack of activity (see Section A). The timeout is measured in
3194
Schulzrinne, et. al. Standards Track [Page 57]
3196
RFC 2326 Real Time Streaming Protocol April 1998
3199
seconds, with a default of 60 seconds (1 minute).
3201
Note that a session identifier identifies a RTSP session across
3202
transport sessions or connections. Control messages for more than one
3203
RTSP URL may be sent within a single RTSP session. Hence, it is
3204
possible that clients use the same session for controlling many
3205
streams constituting a presentation, as long as all the streams come
3206
from the same server. (See example in Section 14). However, multiple
3207
"user" sessions for the same URL from the same client MUST use
3208
different session identifiers.
3210
The session identifier is needed to distinguish several delivery
3211
requests for the same URL coming from the same client.
3213
The response 454 (Session Not Found) is returned if the session
3214
identifier is invalid.
3218
The timestamp general header describes when the client sent the
3219
request to the server. The value of the timestamp is of significance
3220
only to the client and may use any timescale. The server MUST echo
3221
the exact same value and MAY, if it has accurate information about
3222
this, add a floating point number indicating the number of seconds
3223
that has elapsed since it has received the request. The timestamp is
3224
used by the client to compute the round-trip time to the server so
3225
that it can adjust the timeout value for retransmissions.
3227
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
3228
delay = *(DIGIT) [ "." *(DIGIT) ]
3232
This request header indicates which transport protocol is to be used
3233
and configures its parameters such as destination address,
3234
compression, multicast time-to-live and destination port for a single
3235
stream. It sets those values not already determined by a presentation
3238
Transports are comma separated, listed in order of preference.
3239
Parameters may be added to each transport, separated by a semicolon.
3241
The Transport header MAY also be used to change certain transport
3242
parameters. A server MAY refuse to change parameters of an existing
3245
The server MAY return a Transport response header in the response to
3246
indicate the values actually chosen.
3250
Schulzrinne, et. al. Standards Track [Page 58]
3252
RFC 2326 Real Time Streaming Protocol April 1998
3255
A Transport request header field may contain a list of transport
3256
options acceptable to the client. In that case, the server MUST
3257
return a single option which was actually chosen.
3259
The syntax for the transport specifier is
3261
transport/profile/lower-transport.
3263
The default value for the "lower-transport" parameters is specific to
3264
the profile. For RTP/AVP, the default is UDP.
3266
Below are the configuration parameters associated with transport:
3270
unicast | multicast:
3271
mutually exclusive indication of whether unicast or multicast
3272
delivery will be attempted. Default value is multicast.
3273
Clients that are capable of handling both unicast and
3274
multicast transmission MUST indicate such capability by
3275
including two full transport-specs with separate parameters
3279
The address to which a stream will be sent. The client may
3280
specify the multicast address with the destination parameter.
3281
To avoid becoming the unwitting perpetrator of a remote-
3282
controlled denial-of-service attack, a server SHOULD
3283
authenticate the client and SHOULD log such attempts before
3284
allowing the client to direct a media stream to an address not
3285
chosen by the server. This is particularly important if RTSP
3286
commands are issued via UDP, but implementations cannot rely
3287
on TCP as reliable means of client identification by itself. A
3288
server SHOULD not allow a client to direct media streams to an
3289
address that differs from the address commands are coming
3293
If the source address for the stream is different than can be
3294
derived from the RTSP endpoint address (the server in playback
3295
or the client in recording), the source MAY be specified.
3297
This information may also be available through SDP. However, since
3298
this is more a feature of transport than media initialization, the
3299
authoritative source for this information should be in the SETUP
3306
Schulzrinne, et. al. Standards Track [Page 59]
3308
RFC 2326 Real Time Streaming Protocol April 1998
3312
The number of multicast layers to be used for this media
3313
stream. The layers are sent to consecutive addresses starting
3314
at the destination address.
3317
The mode parameter indicates the methods to be supported for
3318
this session. Valid values are PLAY and RECORD. If not
3319
provided, the default is PLAY.
3322
If the mode parameter includes RECORD, the append parameter
3323
indicates that the media data should append to the existing
3324
resource rather than overwrite it. If appending is requested
3325
and the server does not support this, it MUST refuse the
3326
request rather than overwrite the resource identified by the
3327
URI. The append parameter is ignored if the mode parameter
3328
does not contain RECORD.
3331
The interleaved parameter implies mixing the media stream with
3332
the control stream in whatever protocol is being used by the
3333
control stream, using the mechanism defined in Section 10.12.
3334
The argument provides the channel number to be used in the $
3335
statement. This parameter may be specified as a range, e.g.,
3336
interleaved=4-5 in cases where the transport choice for the
3337
media stream requires it.
3339
This allows RTP/RTCP to be handled similarly to the way that it is
3340
done with UDP, i.e., one channel for RTP and the other for RTCP.
3345
multicast time-to-live
3350
This parameter provides the RTP/RTCP port pair for a multicast
3351
session. It is specified as a range, e.g., port=3456-3457.
3354
This parameter provides the unicast RTP/RTCP port pair on
3355
which the client has chosen to receive media data and control
3356
information. It is specified as a range, e.g.,
3357
client_port=3456-3457.
3362
Schulzrinne, et. al. Standards Track [Page 60]
3364
RFC 2326 Real Time Streaming Protocol April 1998
3368
This parameter provides the unicast RTP/RTCP port pair on
3369
which the server has chosen to receive media data and control
3370
information. It is specified as a range, e.g.,
3371
server_port=3456-3457.
3374
The ssrc parameter indicates the RTP SSRC [24, Sec. 3] value
3375
that should be (request) or will be (response) used by the
3376
media server. This parameter is only valid for unicast
3377
transmission. It identifies the synchronization source to be
3378
associated with the media stream.
3380
Transport = "Transport" ":"
3382
transport-spec = transport-protocol/profile[/lower-transport]
3384
transport-protocol = "RTP"
3386
lower-transport = "TCP" | "UDP"
3387
parameter = ( "unicast" | "multicast" )
3388
| ";" "destination" [ "=" address ]
3389
| ";" "interleaved" "=" channel [ "-" channel ]
3392
| ";" "layers" "=" 1*DIGIT
3393
| ";" "port" "=" port [ "-" port ]
3394
| ";" "client_port" "=" port [ "-" port ]
3395
| ";" "server_port" "=" port [ "-" port ]
3396
| ";" "ssrc" "=" ssrc
3397
| ";" "mode" = <"> 1\#mode <">
3401
channel = 1*3(DIGIT)
3403
mode = <"> *Method <"> | Method
3407
Transport: RTP/AVP;multicast;ttl=127;mode="PLAY",
3408
RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"
3410
The Transport header is restricted to describing a single RTP
3411
stream. (RTSP can also control multiple streams as a single
3412
entity.) Making it part of RTSP rather than relying on a multitude
3413
of session description formats greatly simplifies designs of
3418
Schulzrinne, et. al. Standards Track [Page 61]
3420
RFC 2326 Real Time Streaming Protocol April 1998
3425
The Unsupported response header lists the features not supported by
3426
the server. In the case where the feature was specified via the
3427
Proxy-Require field (Section 12.32), if there is a proxy on the path
3428
between the client and the server, the proxy MUST insert a message
3429
reply with an error message "551 Option Not Supported".
3431
See Section 12.32 for a usage example.
3445
12.44 WWW-Authentica
3451
In HTTP, response-request pairs are cached. RTSP differs
3452
significantly in that respect. Responses are not cacheable, with the
3453
exception of the presentation description returned by DESCRIBE or
3454
included with ANNOUNCE. (Since the responses for anything but
3455
DESCRIBE and GET_PARAMETER do not return any data, caching is not
3456
really an issue for these requests.) However, it is desirable for the
3457
continuous media data, typically delivered out-of-band with respect
3458
to RTSP, to be cached, as well as the session description.
3460
On receiving a SETUP or PLAY request, a proxy ascertains whether it
3461
has an up-to-date copy of the continuous media content and its
3462
description. It can determine whether the copy is up-to-date by
3463
issuing a SETUP or DESCRIBE request, respectively, and comparing the
3464
Last-Modified header with that of the cached copy. If the copy is not
3465
up-to-date, it modifies the SETUP transport parameters as appropriate
3466
and forwards the request to the origin server. Subsequent control
3467
commands such as PLAY or PAUSE then pass the proxy unmodified. The
3468
proxy delivers the continuous media data to the client, while
3469
possibly making a local copy for later reuse. The exact behavior
3470
allowed to the cache is given by the cache-response directives
3474
Schulzrinne, et. al. Standards Track [Page 62]
3476
RFC 2326 Real Time Streaming Protocol April 1998
3479
described in Section 12.8. A cache MUST answer any DESCRIBE requests
3480
if it is currently serving the stream to the requestor, as it is
3481
possible that low-level details of the stream description may have
3482
changed on the origin-server.
3484
Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
3485
through" variety. Rather than retrieving the whole resource from the
3486
origin server, the cache simply copies the streaming data as it
3487
passes by on its way to the client. Thus, it does not introduce
3490
To the client, an RTSP proxy cache appears like a regular media
3491
server, to the media origin server like a client. Just as an HTTP
3492
cache has to store the content type, content language, and so on for
3493
the objects it caches, a media cache has to store the presentation
3494
description. Typically, a cache eliminates all transport-references
3495
(that is, multicast information) from the presentation description,
3496
since these are independent of the data delivery from the cache to
3497
the client. Information on the encodings remains the same. If the
3498
cache is able to translate the cached media data, it would create a
3499
new presentation description with all the encoding possibilities it
3504
The following examples refer to stream description formats that are
3505
not standards, such as RTSL. The following examples are not to be
3506
used as a reference for those formats.
3508
14.1 Media on Demand (Unicast)
3510
Client C requests a movie from media servers A ( audio.example.com)
3511
and V (video.example.com). The media description is stored on a web
3512
server W . The media description contains descriptions of the
3513
presentation and all its streams, including the codecs that are
3514
available, dynamic RTP payload types, the protocol stack, and content
3515
information such as language or copyright restrictions. It may also
3516
give an indication about the timeline of the movie.
3518
In this example, the client is only interested in the last part of
3521
C->W: GET /twister.sdp HTTP/1.1
3522
Host: www.example.com
3523
Accept: application/sdp
3525
W->C: HTTP/1.0 200 OK
3526
Content-Type: application/sdp
3530
Schulzrinne, et. al. Standards Track [Page 63]
3532
RFC 2326 Real Time Streaming Protocol April 1998
3536
o=- 2890844526 2890842807 IN IP4 192.16.24.202
3539
a=control:rtsp://audio.example.com/twister/audio.en
3540
m=video 0 RTP/AVP 31
3541
a=control:rtsp://video.example.com/twister/video
3543
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
3545
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
3547
A->C: RTSP/1.0 200 OK
3550
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
3551
server_port=5000-5001
3553
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
3555
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
3557
V->C: RTSP/1.0 200 OK
3560
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
3561
server_port=5002-5003
3563
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
3566
Range: smpte=0:10:00-
3568
V->C: RTSP/1.0 200 OK
3571
Range: smpte=0:10:00-0:20:00
3572
RTP-Info: url=rtsp://video.example.com/twister/video;
3573
seq=12312232;rtptime=78712811
3575
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
3578
Range: smpte=0:10:00-
3580
A->C: RTSP/1.0 200 OK
3586
Schulzrinne, et. al. Standards Track [Page 64]
3588
RFC 2326 Real Time Streaming Protocol April 1998
3591
Range: smpte=0:10:00-0:20:00
3592
RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
3593
seq=876655;rtptime=1032181
3595
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
3599
A->C: RTSP/1.0 200 OK
3602
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
3606
V->C: RTSP/1.0 200 OK
3609
Even though the audio and video track are on two different servers,
3610
and may start at slightly different times and may drift with respect
3611
to each other, the client can synchronize the two using standard RTP
3612
methods, in particular the time scale contained in the RTCP sender
3615
14.2 Streaming of a Container file
3617
For purposes of this example, a container file is a storage entity in
3618
which multiple continuous media types pertaining to the same end-user
3619
presentation are present. In effect, the container file represents an
3620
RTSP presentation, with each of its components being RTSP streams.
3621
Container files are a widely used means to store such presentations.
3622
While the components are transported as independent streams, it is
3623
desirable to maintain a common context for those streams at the
3626
This enables the server to keep a single storage handle open
3627
easily. It also allows treating all the streams equally in case of
3628
any prioritization of streams by the server.
3630
It is also possible that the presentation author may wish to prevent
3631
selective retrieval of the streams by the client in order to preserve
3632
the artistic effect of the combined media presentation. Similarly, in
3633
such a tightly bound presentation, it is desirable to be able to
3634
control all the streams via a single control message using an
3637
The following is an example of using a single RTSP session to control
3638
multiple streams. It also illustrates the use of aggregate URLs.
3642
Schulzrinne, et. al. Standards Track [Page 65]
3644
RFC 2326 Real Time Streaming Protocol April 1998
3647
Client C requests a presentation from media server M . The movie is
3648
stored in a container file. The client has obtained an RTSP URL to
3651
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
3654
M->C: RTSP/1.0 200 OK
3656
Content-Type: application/sdp
3660
o=- 2890844256 2890842807 IN IP4 172.16.2.93
3662
i=An Example of RTSP Session Usage
3663
a=control:rtsp://foo/twister
3666
a=control:rtsp://foo/twister/audio
3667
m=video 0 RTP/AVP 26
3668
a=control:rtsp://foo/twister/video
3670
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
3672
Transport: RTP/AVP;unicast;client_port=8000-8001
3674
M->C: RTSP/1.0 200 OK
3676
Transport: RTP/AVP;unicast;client_port=8000-8001;
3677
server_port=9000-9001
3680
C->M: SETUP rtsp://foo/twister/video RTSP/1.0
3682
Transport: RTP/AVP;unicast;client_port=8002-8003
3685
M->C: RTSP/1.0 200 OK
3687
Transport: RTP/AVP;unicast;client_port=8002-8003;
3688
server_port=9004-9005
3691
C->M: PLAY rtsp://foo/twister RTSP/1.0
3698
Schulzrinne, et. al. Standards Track [Page 66]
3700
RFC 2326 Real Time Streaming Protocol April 1998
3703
M->C: RTSP/1.0 200 OK
3706
RTP-Info: url=rtsp://foo/twister/video;
3707
seq=9810092;rtptime=3450012
3709
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
3713
M->C: RTSP/1.0 460 Only aggregate operation allowed
3716
C->M: PAUSE rtsp://foo/twister RTSP/1.0
3720
M->C: RTSP/1.0 200 OK
3724
C->M: SETUP rtsp://foo/twister RTSP/1.0
3726
Transport: RTP/AVP;unicast;client_port=10000
3728
M->C: RTSP/1.0 459 Aggregate operation not allowed
3732
In the first instance of failure, the client tries to pause one
3733
stream (in this case video) of the presentation. This is disallowed
3734
for that presentation by the server. In the second instance, the
3735
aggregate URL may not be used for SETUP and one control message is
3736
required per stream to set up transport parameters.
3738
This keeps the syntax of the Transport header simple and allows
3739
easy parsing of transport information by firewalls.
3741
14.3 Single Stream Container Files
3743
Some RTSP servers may treat all files as though they are "container
3744
files", yet other servers may not support such a concept. Because of
3745
this, clients SHOULD use the rules set forth in the session
3746
description for request URLs, rather than assuming that a consistent
3747
URL may always be used throughout. Here's an example of how a multi-
3748
stream server might expect a single-stream file to be served:
3750
Accept: application/x-rtsp-mh, application/sdp
3754
Schulzrinne, et. al. Standards Track [Page 67]
3756
RFC 2326 Real Time Streaming Protocol April 1998
3761
S->C RTSP/1.0 200 OK
3763
Content-base: rtsp://foo.com/test.wav/
3764
Content-type: application/sdp
3768
o=- 872653257 872653257 IN IP4 172.16.2.187
3773
a=control:streamid=0
3775
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
3776
Transport: RTP/AVP/UDP;unicast;
3777
client_port=6970-6971;mode=play
3780
S->C RTSP/1.0 200 OK
3781
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
3782
server_port=6970-6971;mode=play
3786
C->S PLAY rtsp://foo.com/test.wav RTSP/1.0
3790
S->C RTSP/1.0 200 OK
3793
RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
3794
seq=981888;rtptime=3781123
3796
Note the different URL in the SETUP command, and then the switch back
3797
to the aggregate URL in the PLAY command. This makes complete sense
3798
when there are multiple streams with aggregate control, but is less
3799
than intuitive in the special case where the number of streams is
3802
In this special case, it is recommended that servers be forgiving of
3803
implementations that send:
3805
C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
3810
Schulzrinne, et. al. Standards Track [Page 68]
3812
RFC 2326 Real Time Streaming Protocol April 1998
3815
In the worst case, servers should send back:
3817
S->C RTSP/1.0 460 Only aggregate operation allowed
3820
One would also hope that server implementations are also forgiving of
3823
C->S SETUP rtsp://foo.com/test.wav RTSP/1.0
3824
Transport: rtp/avp/udp;client_port=6970-6971;mode=play
3827
Since there is only a single stream in this file, it's not ambiguous
3830
14.4 Live Media Presentation Using Multicast
3832
The media server M chooses the multicast address and port. Here, we
3833
assume that the web server only contains a pointer to the full
3834
description, while the media server M maintains the full description.
3836
C->W: GET /concert.sdp HTTP/1.1
3837
Host: www.example.com
3839
W->C: HTTP/1.1 200 OK
3840
Content-Type: application/x-rtsl
3843
<track src="rtsp://live.example.com/concert/audio">
3846
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
3849
M->C: RTSP/1.0 200 OK
3851
Content-Type: application/sdp
3855
o=- 2890844526 2890842807 IN IP4 192.16.24.202
3857
m=audio 3456 RTP/AVP 0
3858
a=control:rtsp://live.example.com/concert/audio
3859
c=IN IP4 224.2.0.1/16
3861
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
3866
Schulzrinne, et. al. Standards Track [Page 69]
3868
RFC 2326 Real Time Streaming Protocol April 1998
3871
Transport: RTP/AVP;multicast
3873
M->C: RTSP/1.0 200 OK
3875
Transport: RTP/AVP;multicast;destination=224.2.0.1;
3876
port=3456-3457;ttl=16
3879
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
3883
M->C: RTSP/1.0 200 OK
3887
14.5 Playing media into an existing session
3889
A conference participant C wants to have the media server M play back
3890
a demo tape into an existing conference. C indicates to the media
3891
server that the network addresses and encryption keys are already
3892
given by the conference, so they should not be chosen by the server.
3893
The example omits the simple ACK responses.
3895
C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
3897
Accept: application/sdp
3899
M->C: RTSP/1.0 200 1 OK
3900
Content-type: application/sdp
3904
o=- 2890844526 2890842807 IN IP4 192.16.24.202
3910
C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0
3912
Transport: RTP/AVP;multicast;destination=225.219.201.15;
3913
port=7000-7001;ttl=127
3914
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
3916
M->C: RTSP/1.0 200 OK
3918
Transport: RTP/AVP;multicast;destination=225.219.201.15;
3922
Schulzrinne, et. al. Standards Track [Page 70]
3924
RFC 2326 Real Time Streaming Protocol April 1998
3927
port=7000-7001;ttl=127
3928
Session: 91389234234
3929
Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu%20Starr
3931
C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
3933
Session: 91389234234
3935
M->C: RTSP/1.0 200 OK
3940
The conference participant client C asks the media server M to record
3941
the audio and video portions of a meeting. The client uses the
3942
ANNOUNCE method to provide meta-information about the recorded
3943
session to the server.
3945
C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
3947
Content-Type: application/sdp
3951
o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
3952
s=IETF Meeting, Munich - 1
3953
i=The thirty-ninth IETF meeting will be held in Munich, Germany
3954
u=http://www.ietf.org/meetings/Munich.html
3955
e=IETF Channel 1 <ietf39-mbone@uni-koeln.de>
3956
p=IETF Channel 1 +49-172-2312 451
3957
c=IN IP4 224.0.1.11/127
3958
t=3080271600 3080703600
3961
m=audio 21010 RTP/AVP 5
3962
c=IN IP4 224.0.1.11/127
3964
m=video 61010 RTP/AVP 31
3965
c=IN IP4 224.0.1.12/127
3967
M->C: RTSP/1.0 200 OK
3970
C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
3972
Transport: RTP/AVP;multicast;destination=224.0.1.11;
3973
port=21010-21011;mode=record;ttl=127
3978
Schulzrinne, et. al. Standards Track [Page 71]
3980
RFC 2326 Real Time Streaming Protocol April 1998
3983
M->C: RTSP/1.0 200 OK
3986
Transport: RTP/AVP;multicast;destination=224.0.1.11;
3987
port=21010-21011;mode=record;ttl=127
3989
C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
3992
Transport: RTP/AVP;multicast;destination=224.0.1.12;
3993
port=61010-61011;mode=record;ttl=127
3995
M->C: RTSP/1.0 200 OK
3997
Transport: RTP/AVP;multicast;destination=224.0.1.12;
3998
port=61010-61011;mode=record;ttl=127
4000
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
4003
Range: clock=19961110T1925-19961110T2015
4005
M->C: RTSP/1.0 200 OK
4010
The RTSP syntax is described in an augmented Backus-Naur form (BNF)
4011
as used in RFC 2068 [2].
4015
OCTET = <any 8-bit sequence of data>
4016
CHAR = <any US-ASCII character (octets 0 - 127)>
4017
UPALPHA = <any US-ASCII uppercase letter "A".."Z">
4018
LOALPHA = <any US-ASCII lowercase letter "a".."z">
4019
ALPHA = UPALPHA | LOALPHA
4021
DIGIT = <any US-ASCII digit "0".."9">
4022
CTL = <any US-ASCII control character
4023
(octets 0 - 31) and DEL (127)>
4024
CR = <US-ASCII CR, carriage return (13)>
4025
LF = <US-ASCII LF, linefeed (10)>
4027
SP = <US-ASCII SP, space (32)>
4028
HT = <US-ASCII HT, horizontal-tab (9)>
4029
<"> = <US-ASCII double-quote mark (34)>
4034
Schulzrinne, et. al. Standards Track [Page 72]
4036
RFC 2326 Real Time Streaming Protocol April 1998
4039
LWS = [CRLF] 1*( SP | HT )
4040
TEXT = <any OCTET except CTLs>
4041
tspecials = "(" | ")" | "<" | ">" | "@"
4042
| "," | ";" | ":" | "\" | <">
4043
| "/" | "[" | "]" | "?" | "="
4044
| "{" | "}" | SP | HT
4046
token = 1*<any CHAR except CTLs or tspecials>
4047
quoted-string = ( <"> *(qdtext) <"> )
4048
qdtext = <any TEXT except <">>
4049
quoted-pair = "\" CHAR
4051
message-header = field-name ":" [ field-value ] CRLF
4053
field-value = *( field-content | LWS )
4054
field-content = <the OCTETs making up the field-value and
4055
consisting of either *TEXT or
4056
combinations of token, tspecials, and
4059
safe = "\$" | "-" | "_" | "." | "+"
4060
extra = "!" | "*" | "$'$" | "(" | ")" | ","
4062
hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
4063
"a" | "b" | "c" | "d" | "e" | "f"
4064
escape = "\%" hex hex
4065
reserved = ";" | "/" | "?" | ":" | "@" | "&" | "="
4067
unreserved = alpha | digit | safe | extra
4068
xchar = unreserved | reserved | escape
4070
16 Security Considerations
4072
Because of the similarity in syntax and usage between RTSP servers
4073
and HTTP servers, the security considerations outlined in [H15]
4074
apply. Specifically, please note the following:
4076
Authentication Mechanisms:
4077
RTSP and HTTP share common authentication schemes, and thus
4078
should follow the same prescriptions with regards to
4079
authentication. See [H15.1] for client authentication issues,
4080
and [H15.2] for issues regarding support for multiple
4081
authentication mechanisms.
4083
Abuse of Server Log Information:
4084
RTSP and HTTP servers will presumably have similar logging
4085
mechanisms, and thus should be equally guarded in protecting
4086
the contents of those logs, thus protecting the privacy of the
4090
Schulzrinne, et. al. Standards Track [Page 73]
4092
RFC 2326 Real Time Streaming Protocol April 1998
4095
users of the servers. See [H15.3] for HTTP server
4096
recommendations regarding server logs.
4098
Transfer of Sensitive Information:
4099
There is no reason to believe that information transferred via
4100
RTSP may be any less sensitive than that normally transmitted
4101
via HTTP. Therefore, all of the precautions regarding the
4102
protection of data privacy and user privacy apply to
4103
implementors of RTSP clients, servers, and proxies. See
4104
[H15.4] for further details.
4106
Attacks Based On File and Path Names:
4107
Though RTSP URLs are opaque handles that do not necessarily
4108
have file system semantics, it is anticipated that many
4109
implementations will translate portions of the request URLs
4110
directly to file system calls. In such cases, file systems
4111
SHOULD follow the precautions outlined in [H15.5], such as
4112
checking for ".." in path components.
4114
Personal Information:
4115
RTSP clients are often privy to the same information that HTTP
4116
clients are (user name, location, etc.) and thus should be
4117
equally. See [H15.6] for further recommendations.
4119
Privacy Issues Connected to Accept Headers:
4120
Since may of the same "Accept" headers exist in RTSP as in
4121
HTTP, the same caveats outlined in [H15.7] with regards to
4122
their use should be followed.
4125
Presumably, given the longer connection times typically
4126
associated to RTSP sessions relative to HTTP sessions, RTSP
4127
client DNS optimizations should be less prevalent.
4128
Nonetheless, the recommendations provided in [H15.8] are still
4129
relevant to any implementation which attempts to rely on a
4130
DNS-to-IP mapping to hold beyond a single use of the mapping.
4132
Location Headers and Spoofing:
4133
If a single server supports multiple organizations that do not
4134
trust one another, then it must check the values of Location
4135
and Content-Location headers in responses that are generated
4136
under control of said organizations to make sure that they do
4137
not attempt to invalidate resources over which they have no
4138
authority. ([H15.9])
4140
In addition to the recommendations in the current HTTP specification
4141
(RFC 2068 [2], as of this writing), future HTTP specifications may
4142
provide additional guidance on security issues.
4146
Schulzrinne, et. al. Standards Track [Page 74]
4148
RFC 2326 Real Time Streaming Protocol April 1998
4151
The following are added considerations for RTSP implementations.
4153
Concentrated denial-of-service attack:
4154
The protocol offers the opportunity for a remote-controlled
4155
denial-of-service attack. The attacker may initiate traffic
4156
flows to one or more IP addresses by specifying them as the
4157
destination in SETUP requests. While the attacker's IP address
4158
may be known in this case, this is not always useful in
4159
prevention of more attacks or ascertaining the attackers
4160
identity. Thus, an RTSP server SHOULD only allow client-
4161
specified destinations for RTSP-initiated traffic flows if the
4162
server has verified the client's identity, either against a
4163
database of known users using RTSP authentication mechanisms
4164
(preferably digest authentication or stronger), or other
4168
Since there is no relation between a transport layer
4169
connection and an RTSP session, it is possible for a malicious
4170
client to issue requests with random session identifiers which
4171
would affect unsuspecting clients. The server SHOULD use a
4172
large, random and non-sequential session identifier to
4173
minimize the possibility of this kind of attack.
4176
Servers SHOULD implement both basic and digest [8]
4177
authentication. In environments requiring tighter security for
4178
the control messages, the RTSP control stream may be
4182
RTSP only provides for stream control. Stream delivery issues
4183
are not covered in this section, nor in the rest of this memo.
4184
RTSP implementations will most likely rely on other protocols
4185
such as RTP, IP multicast, RSVP and IGMP, and should address
4186
security considerations brought up in those and other
4187
applicable specifications.
4189
Persistently suspicious behavior:
4190
RTSP servers SHOULD return error code 403 (Forbidden) upon
4191
receiving a single instance of behavior which is deemed a
4192
security risk. RTSP servers SHOULD also be aware of attempts
4193
to probe the server for weaknesses and entry points and MAY
4194
arbitrarily disconnect and ignore further requests clients
4195
which are deemed to be in violation of local security policy.
4202
Schulzrinne, et. al. Standards Track [Page 75]
4204
RFC 2326 Real Time Streaming Protocol April 1998
4207
Appendix A: RTSP Protocol State Machines
4209
The RTSP client and server state machines describe the behavior of
4210
the protocol from RTSP session initialization through RTSP session
4213
State is defined on a per object basis. An object is uniquely
4214
identified by the stream URL and the RTSP session identifier. Any
4215
request/reply using aggregate URLs denoting RTSP presentations
4216
composed of multiple streams will have an effect on the individual
4217
states of all the streams. For example, if the presentation /movie
4218
contains two streams, /movie/audio and /movie/video, then the
4221
PLAY rtsp://foo.com/movie RTSP/1.0
4225
will have an effect on the states of movie/audio and movie/video.
4227
This example does not imply a standard way to represent streams in
4228
URLs or a relation to the filesystem. See Section 3.2.
4230
The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER,
4231
SET_PARAMETER do not have any effect on client or server state and
4232
are therefore not listed in the state tables.
4234
A.1 Client State Machine
4236
The client can assume the following states:
4239
SETUP has been sent, waiting for reply.
4242
SETUP reply received or PAUSE reply received while in Playing
4249
RECORD reply received
4251
In general, the client changes state on receipt of replies to
4252
requests. Note that some requests are effective at a future time or
4253
position (such as a PAUSE), and state also changes accordingly. If no
4254
explicit SETUP is required for the object (for example, it is
4258
Schulzrinne, et. al. Standards Track [Page 76]
4260
RFC 2326 Real Time Streaming Protocol April 1998
4263
available via a multicast group), state begins at Ready. In this
4264
case, there are only two states, Ready and Playing. The client also
4265
changes state from Playing/Recording to Ready when the end of the
4266
requested range is reached.
4268
The "next state" column indicates the state assumed after receiving a
4269
success response (2xx). If a request yields a status code of 3xx, the
4270
state becomes Init, and a status code of 4xx yields no change in
4271
state. Messages not listed for each state MUST NOT be issued by the
4272
client in that state, with the exception of messages not affecting
4273
state, as listed above. Receiving a REDIRECT from the server is
4274
equivalent to receiving a 3xx redirect status from the server.
4277
state message sent next state after response
4287
SETUP Playing (changed transport)
4288
Recording PAUSE Ready
4291
SETUP Recording (changed transport)
4293
A.2 Server State Machine
4295
The server can assume the following states:
4298
The initial state, no valid SETUP has been received yet.
4301
Last SETUP received was successful, reply sent or after
4302
playing, last PAUSE received was successful, reply sent.
4305
Last PLAY received was successful, reply sent. Data is being
4309
The server is recording media data.
4314
Schulzrinne, et. al. Standards Track [Page 77]
4316
RFC 2326 Real Time Streaming Protocol April 1998
4319
In general, the server changes state on receiving requests. If the
4320
server is in state Playing or Recording and in unicast mode, it MAY
4321
revert to Init and tear down the RTSP session if it has not received
4322
"wellness" information, such as RTCP reports or RTSP commands, from
4323
the client for a defined interval, with a default of one minute. The
4324
server can declare another timeout value in the Session response
4325
header (Section 12.37). If the server is in state Ready, it MAY
4326
revert to Init if it does not receive an RTSP request for an interval
4327
of more than one minute. Note that some requests (such as PAUSE) may
4328
be effective at a future time or position, and server state changes
4329
at the appropriate time. The server reverts from state Playing or
4330
Recording to state Ready at the end of the range requested by the
4333
The REDIRECT message, when sent, is effective immediately unless it
4334
has a Range header specifying when the redirect is effective. In such
4335
a case, server state will also change at the appropriate time.
4337
If no explicit SETUP is required for the object, the state starts at
4338
Ready and there are only two states, Ready and Playing.
4340
The "next state" column indicates the state assumed after sending a
4341
success response (2xx). If a request results in a status code of 3xx,
4342
the state becomes Init. A status code of 4xx results in no change.
4344
state message received next state
4351
Playing PLAY Playing
4355
Recording RECORD Recording
4370
Schulzrinne, et. al. Standards Track [Page 78]
4372
RFC 2326 Real Time Streaming Protocol April 1998
4375
Appendix B: Interaction with RTP
4377
RTSP allows media clients to control selected, non-contiguous
4378
sections of media presentations, rendering those streams with an RTP
4379
media layer[24]. The media layer rendering the RTP stream should not
4380
be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
4381
timestamps MUST be continuous and monotonic across jumps of NPT.
4383
As an example, assume a clock frequency of 8000 Hz, a packetization
4384
interval of 100 ms and an initial sequence number and timestamp of
4385
zero. First we play NPT 10 through 15, then skip ahead and play NPT
4386
18 through 20. The first segment is presented as RTP packets with
4387
sequence numbers 0 through 49 and timestamp 0 through 39,200. The
4388
second segment consists of RTP packets with sequence number 50
4389
through 69, with timestamps 40,000 through 55,200.
4391
We cannot assume that the RTSP client can communicate with the RTP
4392
media agent, as the two may be independent processes. If the RTP
4393
timestamp shows the same gap as the NPT, the media agent will
4394
assume that there is a pause in the presentation. If the jump in
4395
NPT is large enough, the RTP timestamp may roll over and the media
4396
agent may believe later packets to be duplicates of packets just
4399
For certain datatypes, tight integration between the RTSP layer and
4400
the RTP layer will be necessary. This by no means precludes the
4401
above restriction. Combined RTSP/RTP media clients should use the
4402
RTP-Info field to determine whether incoming RTP packets were sent
4403
before or after a seek.
4405
For continuous audio, the server SHOULD set the RTP marker bit at the
4406
beginning of serving a new PLAY request. This allows the client to
4407
perform playout delay adaptation.
4409
For scaling (see Section 12.34), RTP timestamps should correspond to
4410
the playback timing. For example, when playing video recorded at 30
4411
frames/second at a scale of two and speed (Section 12.35) of one, the
4412
server would drop every second frame to maintain and deliver video
4413
packets with the normal timestamp spacing of 3,000 per frame, but NPT
4414
would increase by 1/15 second for each video frame.
4416
The client can maintain a correct display of NPT by noting the RTP
4417
timestamp value of the first packet arriving after repositioning. The
4418
sequence parameter of the RTP-Info (Section 12.33) header provides
4419
the first sequence number of the next segment.
4426
Schulzrinne, et. al. Standards Track [Page 79]
4428
RFC 2326 Real Time Streaming Protocol April 1998
4431
Appendix C: Use of SDP for RTSP Session Descriptions
4433
The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
4434
describe streams or presentations in RTSP. Such usage is limited to
4435
specifying means of access and encoding(s) for:
4438
A presentation composed of streams from one or more servers
4439
that are not available for aggregate control. Such a
4440
description is typically retrieved by HTTP or other non-RTSP
4441
means. However, they may be received with ANNOUNCE methods.
4443
non-aggregate control:
4444
A presentation composed of multiple streams from a single
4445
server that are available for aggregate control. Such a
4446
description is typically returned in reply to a DESCRIBE
4447
request on a URL, or received in an ANNOUNCE method.
4449
This appendix describes how an SDP file, retrieved, for example,
4450
through HTTP, determines the operation of an RTSP session. It also
4451
describes how a client should interpret SDP content returned in reply
4452
to a DESCRIBE request. SDP provides no mechanism by which a client
4453
can distinguish, without human guidance, between several media
4454
streams to be rendered simultaneously and a set of alternatives
4455
(e.g., two audio streams spoken in different languages).
4459
The terms "session-level", "media-level" and other key/attribute
4460
names and values used in this appendix are to be used as defined in
4465
The "a=control:" attribute is used to convey the control URL. This
4466
attribute is used both for the session and media descriptions. If
4467
used for individual media, it indicates the URL to be used for
4468
controlling that particular media stream. If found at the session
4469
level, the attribute indicates the URL for aggregate control.
4472
a=control:rtsp://example.com/foo
4474
This attribute may contain either relative and absolute URLs,
4475
following the rules and conventions set out in RFC 1808 [25].
4476
Implementations should look for a base URL in the following order:
4482
Schulzrinne, et. al. Standards Track [Page 80]
4484
RFC 2326 Real Time Streaming Protocol April 1998
4487
1. The RTSP Content-Base field
4488
2. The RTSP Content-Location field
4489
3. The RTSP request URL
4491
If this attribute contains only an asterisk (*), then the URL is
4492
treated as if it were an empty embedded URL, and thus inherits the
4497
The "m=" field is used to enumerate the streams. It is expected that
4498
all the specified streams will be rendered with appropriate
4499
synchronization. If the session is unicast, the port number serves as
4500
a recommendation from the server to the client; the client still has
4501
to include it in its SETUP request and may ignore this
4502
recommendation. If the server has no preference, it SHOULD set the
4503
port number value to zero.
4506
m=audio 0 RTP/AVP 31
4508
C.1.3 Payload type(s)
4510
The payload type(s) are specified in the "m=" field. In case the
4511
payload type is a static payload type from RFC 1890 [1], no other
4512
information is required. In case it is a dynamic payload type, the
4513
media attribute "rtpmap" is used to specify what the media is. The
4514
"encoding name" within the "rtpmap" attribute may be one of those
4515
specified in RFC 1890 (Sections 5 and 6), or an experimental encoding
4516
with a "X-" prefix as specified in SDP (RFC 2327 [6]). Codec-
4517
specific parameters are not specified in this field, but rather in
4518
the "fmtp" attribute described below. Implementors seeking to
4519
register new encodings should follow the procedure in RFC 1890 [1].
4520
If the media type is not suited to the RTP AV profile, then it is
4521
recommended that a new profile be created and the appropriate profile
4522
name be used in lieu of "RTP/AVP" in the "m=" field.
4524
C.1.4 Format-specific parameters
4526
Format-specific parameters are conveyed using the "fmtp" media
4527
attribute. The syntax of the "fmtp" attribute is specific to the
4528
encoding(s) that the attribute refers to. Note that the packetization
4529
interval is conveyed using the "ptime" attribute.
4538
Schulzrinne, et. al. Standards Track [Page 81]
4540
RFC 2326 Real Time Streaming Protocol April 1998
4543
C.1.5 Range of presentation
4545
The "a=range" attribute defines the total time range of the stored
4546
session. (The length of live sessions can be deduced from the "t" and
4547
"r" parameters.) Unless the presentation contains media streams of
4548
different durations, the range attribute is a session-level
4549
attribute. The unit is specified first, followed by the value range.
4550
The units and their values are as defined in Section 3.5, 3.6 and
4554
a=range:npt=0-34.4368
4555
a=range:clock=19971113T2115-19971113T2203
4557
C.1.6 Time of availability
4559
The "t=" field MUST contain suitable values for the start and stop
4560
times for both aggregate and non-aggregate stream control. With
4561
aggregate control, the server SHOULD indicate a stop time value for
4562
which it guarantees the description to be valid, and a start time
4563
that is equal to or before the time at which the DESCRIBE request was
4564
received. It MAY also indicate start and stop times of 0, meaning
4565
that the session is always available. With non-aggregate control, the
4566
values should reflect the actual period for which the session is
4567
available in keeping with SDP semantics, and not depend on other
4568
means (such as the life of the web page containing the description)
4571
C.1.7 Connection Information
4573
In SDP, the "c=" field contains the destination address for the media
4574
stream. However, for on-demand unicast streams and some multicast
4575
streams, the destination address is specified by the client via the
4576
SETUP request. Unless the media content has a fixed destination
4577
address, the "c=" field is to be set to a suitable null value. For
4578
addresses of type "IP4", this value is "0.0.0.0".
4582
The optional "a=etag" attribute identifies a version of the session
4583
description. It is opaque to the client. SETUP requests may include
4584
this identifier in the If-Match field (see section 12.22) to only
4585
allow session establishment if this attribute value still corresponds
4586
to that of the current description. The attribute value is opaque and
4587
may contain any character allowed within SDP attribute values.
4590
a=etag:158bb3e7c7fd62ce67f12b533f06b83a
4594
Schulzrinne, et. al. Standards Track [Page 82]
4596
RFC 2326 Real Time Streaming Protocol April 1998
4599
One could argue that the "o=" field provides identical
4600
functionality. However, it does so in a manner that would put
4601
constraints on servers that need to support multiple session
4602
description types other than SDP for the same piece of media
4605
C.2 Aggregate Control Not Available
4607
If a presentation does not support aggregate control and multiple
4608
media sections are specified, each section MUST have the control URL
4609
specified via the "a=control:" attribute.
4613
o=- 2890844256 2890842807 IN IP4 204.34.34.32
4614
s=I came from a web page
4617
m=video 8002 RTP/AVP 31
4618
a=control:rtsp://audio.com/movie.aud
4619
m=audio 8004 RTP/AVP 3
4620
a=control:rtsp://video.com/movie.vid
4622
Note that the position of the control URL in the description implies
4623
that the client establishes separate RTSP control sessions to the
4624
servers audio.com and video.com.
4626
It is recommended that an SDP file contains the complete media
4627
initialization information even if it is delivered to the media
4628
client through non-RTSP means. This is necessary as there is no
4629
mechanism to indicate that the client should request more detailed
4630
media stream information via DESCRIBE.
4632
C.3 Aggregate Control Available
4634
In this scenario, the server has multiple streams that can be
4635
controlled as a whole. In this case, there are both media-level
4636
"a=control:" attributes, which are used to specify the stream URLs,
4637
and a session-level "a=control:" attribute which is used as the
4638
request URL for aggregate control. If the media-level URL is
4639
relative, it is resolved to absolute URLs according to Section C.1.1
4642
If the presentation comprises only a single stream, the media-level
4643
"a=control:" attribute may be omitted altogether. However, if the
4644
presentation contains more than one stream, each media stream section
4645
MUST contain its own "a=control" attribute.
4650
Schulzrinne, et. al. Standards Track [Page 83]
4652
RFC 2326 Real Time Streaming Protocol April 1998
4657
o=- 2890844256 2890842807 IN IP4 204.34.34.32
4662
a=control:rtsp://example.com/movie/
4663
m=video 8002 RTP/AVP 31
4665
m=audio 8004 RTP/AVP 3
4668
In this example, the client is required to establish a single RTSP
4669
session to the server, and uses the URLs
4670
rtsp://example.com/movie/trackID=1 and
4671
rtsp://example.com/movie/trackID=2 to set up the video and audio
4672
streams, respectively. The URL rtsp://example.com/movie/ controls the
4706
Schulzrinne, et. al. Standards Track [Page 84]
4708
RFC 2326 Real Time Streaming Protocol April 1998
4711
Appendix D: Minimal RTSP implementation
4715
A client implementation MUST be able to do the following :
4717
* Generate the following requests: SETUP, TEARDOWN, and one of PLAY
4718
(i.e., a minimal playback client) or RECORD (i.e., a minimal
4719
recording client). If RECORD is implemented, ANNOUNCE must be
4720
implemented as well.
4721
* Include the following headers in requests: CSeq, Connection,
4722
Session, Transport. If ANNOUNCE is implemented, the capability to
4723
include headers Content-Language, Content-Encoding, Content-
4724
Length, and Content-Type should be as well.
4725
* Parse and understand the following headers in responses: CSeq,
4726
Connection, Session, Transport, Content-Language, Content-
4727
Encoding, Content-Length, Content-Type. If RECORD is implemented,
4728
the Location header must be understood as well. RTP-compliant
4729
implementations should also implement RTP-Info.
4730
* Understand the class of each error code received and notify the
4731
end-user, if one is present, of error codes in classes 4xx and
4732
5xx. The notification requirement may be relaxed if the end-user
4733
explicitly does not want it for one or all status codes.
4734
* Expect and respond to asynchronous requests from the server, such
4735
as ANNOUNCE. This does not necessarily mean that it should
4736
implement the ANNOUNCE method, merely that it MUST respond
4737
positively or negatively to any request received from the server.
4739
Though not required, the following are highly recommended at the time
4740
of publication for practical interoperability with initial
4741
implementations and/or to be a "good citizen".
4743
* Implement RTP/AVP/UDP as a valid transport.
4744
* Inclusion of the User-Agent header.
4745
* Understand SDP session descriptions as defined in Appendix C
4746
* Accept media initialization formats (such as SDP) from standard
4747
input, command line, or other means appropriate to the operating
4748
environment to act as a "helper application" for other
4749
applications (such as web browsers).
4751
There may be RTSP applications different from those initially
4752
envisioned by the contributors to the RTSP specification for which
4753
the requirements above do not make sense. Therefore, the
4754
recommendations above serve only as guidelines instead of strict
4762
Schulzrinne, et. al. Standards Track [Page 85]
4764
RFC 2326 Real Time Streaming Protocol April 1998
4767
D.1.1 Basic Playback
4769
To support on-demand playback of media streams, the client MUST
4770
additionally be able to do the following:
4771
* generate the PAUSE request;
4772
* implement the REDIRECT method, and the Location header.
4774
D.1.2 Authentication-enabled
4776
In order to access media presentations from RTSP servers that require
4777
authentication, the client MUST additionally be able to do the
4779
* recognize the 401 status code;
4780
* parse and include the WWW-Authenticate header;
4781
* implement Basic Authentication and Digest Authentication.
4785
A minimal server implementation MUST be able to do the following:
4787
* Implement the following methods: SETUP, TEARDOWN, OPTIONS and
4788
either PLAY (for a minimal playback server) or RECORD (for a
4789
minimal recording server). If RECORD is implemented, ANNOUNCE
4790
should be implemented as well.
4791
* Include the following headers in responses: Connection,
4792
Content-Length, Content-Type, Content-Language, Content-Encoding,
4793
Transport, Public. The capability to include the Location header
4794
should be implemented if the RECORD method is. RTP-compliant
4795
implementations should also implement the RTP-Info field.
4796
* Parse and respond appropriately to the following headers in
4797
requests: Connection, Session, Transport, Require.
4799
Though not required, the following are highly recommended at the time
4800
of publication for practical interoperability with initial
4801
implementations and/or to be a "good citizen".
4803
* Implement RTP/AVP/UDP as a valid transport.
4804
* Inclusion of the Server header.
4805
* Implement the DESCRIBE method.
4806
* Generate SDP session descriptions as defined in Appendix C
4808
There may be RTSP applications different from those initially
4809
envisioned by the contributors to the RTSP specification for which
4810
the requirements above do not make sense. Therefore, the
4811
recommendations above serve only as guidelines instead of strict
4818
Schulzrinne, et. al. Standards Track [Page 86]
4820
RFC 2326 Real Time Streaming Protocol April 1998
4823
D.2.1 Basic Playback
4825
To support on-demand playback of media streams, the server MUST
4826
additionally be able to do the following:
4828
* Recognize the Range header, and return an error if seeking is not
4830
* Implement the PAUSE method.
4832
In addition, in order to support commonly-accepted user interface
4833
features, the following are highly recommended for on-demand media
4836
* Include and parse the Range header, with NPT units.
4837
Implementation of SMPTE units is recommended.
4838
* Include the length of the media presentation in the media
4839
initialization information.
4840
* Include mappings from data-specific timestamps to NPT. When RTP
4841
is used, the rtptime portion of the RTP-Info field may be used to
4842
map RTP timestamps to NPT.
4844
Client implementations may use the presence of length information
4845
to determine if the clip is seekable, and visibly disable seeking
4846
features for clips for which the length information is unavailable.
4847
A common use of the presentation length is to implement a "slider
4848
bar" which serves as both a progress indicator and a timeline
4851
Mappings from RTP timestamps to NPT are necessary to ensure correct
4852
positioning of the slider bar.
4854
D.2.2 Authentication-enabled
4856
In order to correctly handle client authentication, the server MUST
4857
additionally be able to do the following:
4859
* Generate the 401 status code when authentication is required for
4861
* Parse and include the WWW-Authenticate header
4862
* Implement Basic Authentication and Digest Authentication
4874
Schulzrinne, et. al. Standards Track [Page 87]
4876
RFC 2326 Real Time Streaming Protocol April 1998
4879
Appendix E: Authors' Addresses
4882
Dept. of Computer Science
4884
1214 Amsterdam Avenue
4888
EMail: schulzrinne@cs.columbia.edu
4892
Netscape Communications Corp.
4893
501 E. Middlefield Road
4894
Mountain View, CA 94043
4897
EMail: anup@netscape.com
4902
1111 Third Avenue Suite 2900
4906
EMail: robla@real.com
4930
Schulzrinne, et. al. Standards Track [Page 88]
4932
RFC 2326 Real Time Streaming Protocol April 1998
4935
Appendix F: Acknowledgements
4937
This memo is based on the functionality of the original RTSP document
4938
submitted in October 96. It also borrows format and descriptions from
4941
This document has benefited greatly from the comments of all those
4942
participating in the MMUSIC-WG. In addition to those already
4943
mentioned, the following individuals have contributed to this
4946
Rahul Agarwal, Torsten Braun, Brent Browning, Bruce Butterfield,
4947
Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Dunsmuir,
4948
Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter
4949
Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka,
4950
Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan
4951
Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Papadopouli,
4952
Sujal Patel, Ema Patki, Alagu Periyannan, Igor Plotnikov, Pinaki
4953
Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and
4954
John Francis Stracke.
4986
Schulzrinne, et. al. Standards Track [Page 89]
4988
RFC 2326 Real Time Streaming Protocol April 1998
4993
1 Schulzrinne, H., "RTP profile for audio and video conferences
4994
with minimal control", RFC 1890, January 1996.
4996
2 Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
4997
Berners-Lee, "Hypertext transfer protocol - HTTP/1.1", RFC
5000
3 Yergeau, F., Nicol, G., Adams, G., and M. Duerst,
5001
"Internationalization of the hypertext markup language", RFC
5004
4 Bradner, S., "Key words for use in RFCs to indicate
5005
requirement levels", BCP 14, RFC 2119, March 1997.
5007
5 ISO/IEC, "Information technology - generic coding of moving
5008
pictures and associated audio information - part 6: extension
5009
for digital storage media and control," Draft International
5010
Standard ISO 13818-6, International Organization for
5011
Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
5014
6 Handley, M., and V. Jacobson, "SDP: Session Description
5015
Protocol", RFC 2327, April 1998.
5017
7 Franks, J., Hallam-Baker, P., and J. Hostetler, "An extension to
5018
HTTP: digest access authentication", RFC 2069, January 1997.
5020
8 Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
5023
9 Hinden, B. and C. Partridge, "Version 2 of the reliable data
5024
protocol (RDP)", RFC 1151, April 1990.
5026
10 Postel, J., "Transmission control protocol", STD 7, RFC 793,
5029
11 H. Schulzrinne, "A comprehensive multimedia control
5030
architecture for the Internet," in Proc. International
5031
Workshop on Network and Operating System Support for Digital
5032
Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997.
5034
12 International Telecommunication Union, "Visual telephone
5035
systems and equipment for local area networks which provide a
5036
non-guaranteed quality of service," Recommendation H.323,
5037
Telecommunication Standardization Sector of ITU, Geneva,
5038
Switzerland, May 1996.
5042
Schulzrinne, et. al. Standards Track [Page 90]
5044
RFC 2326 Real Time Streaming Protocol April 1998
5047
13 McMahon, P., "GSS-API authentication method for SOCKS version
5048
5", RFC 1961, June 1996.
5050
14 J. Miller, P. Resnick, and D. Singer, "Rating services and
5051
rating systems (and their machine readable descriptions),"
5052
Recommendation REC-PICS-services-961031, W3C (World Wide Web
5053
Consortium), Boston, Massachusetts, Oct. 1996.
5055
15 J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS
5056
label distribution label syntax and communication protocols,"
5057
Recommendation REC-PICS-labels-961031, W3C (World Wide Web
5058
Consortium), Boston, Massachusetts, Oct. 1996.
5060
16 Crocker, D. and P. Overell, "Augmented BNF for syntax
5061
specifications: ABNF", RFC 2234, November 1997.
5063
17 Braden, B., "Requirements for internet hosts - application and
5064
support", STD 3, RFC 1123, October 1989.
5066
18 Elz, R., "A compact representation of IPv6 addresses", RFC
5069
19 Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform
5070
resource locators (URL)", RFC 1738, December 1994.
5072
20 Yergeau, F., "UTF-8, a transformation format of ISO 10646",
5073
RFC 2279, January 1998.
5075
22 Braden, B., "T/TCP - TCP extensions for transactions
5076
functional specification", RFC 1644, July 1994.
5078
22 W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
5079
Reading, Massachusetts: Addison-Wesley, 1994.
5081
23 Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
5082
"RTP: a transport protocol for real-time applications", RFC
5085
24 Fielding, R., "Relative uniform resource locators", RFC 1808,
5098
Schulzrinne, et. al. Standards Track [Page 91]
5100
RFC 2326 Real Time Streaming Protocol April 1998
5103
Full Copyright Statement
5105
Copyright (C) The Internet Society (1998). All Rights Reserved.
5107
This document and translations of it may be copied and furnished to
5108
others, and derivative works that comment on or otherwise explain it
5109
or assist in its implementation may be prepared, copied, published
5110
and distributed, in whole or in part, without restriction of any
5111
kind, provided that the above copyright notice and this paragraph are
5112
included on all such copies and derivative works. However, this
5113
document itself may not be modified in any way, such as by removing
5114
the copyright notice or references to the Internet Society or other
5115
Internet organizations, except as needed for the purpose of
5116
developing Internet standards in which case the procedures for
5117
copyrights defined in the Internet Standards process must be
5118
followed, or as required to translate it into languages other than
5121
The limited permissions granted above are perpetual and will not be
5122
revoked by the Internet Society or its successors or assigns.
5124
This document and the information contained herein is provided on an
5125
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
5126
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
5127
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
5128
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
5129
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
5154
Schulzrinne, et. al. Standards Track [Page 92]
added
removed
doc/ietf/rfc2326.txt
