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Internet Engineering Task Force MMUSIC WG
Internet Draft H. Schulzrinne
draft-ietf-mmusic-rfc2326bis-10.txt Columbia U.
July 18, 2005 Anup Rao
Expires: January 18, 2006 Cisco
Robert Lanphier
Real Networks
Magnus Westerlund
Ericsson
Aravind Narasimhan
Overture Computing
Real Time Streaming Protocol (RTSP)
STATUS OF THIS MEMO
By submitting this Internet-Draft, each author represents that any
applicable patent or other IPR claims of which he or she is aware
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Abstract
This memorandum defines RTSP version 1.1 which is a revision of the
Proposed Standard RTSP version 1.0 which is defined in RFC 2326.
The Real Time Streaming Protocol, or RTSP, is an application-level
protocol for control over the delivery of data with real-time
properties. RTSP provides an extensible framework to enable
controlled, on-demand delivery of real-time data, such as audio and
video. Sources of data can include both live data feeds and stored
clips. This protocol is intended to control multiple data delivery
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sessions, provide a means for choosing delivery channels such as UDP,
multicast UDP and TCP, and provide a means for choosing delivery
mechanisms based upon RTP (RFC 3550).
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Table of Contents
1 Introduction ........................................ 9
1.1 RTSP Specification Update ........................... 9
1.2 Purpose ............................................. 9
1.3 Notational Conventions .............................. 11
1.4 Terminology ......................................... 12
1.5 Protocol Properties ................................. 15
1.6 Extending RTSP ...................................... 16
1.7 Overall Operation ................................... 17
1.8 RTSP States ......................................... 18
1.9 Relationship with Other Protocols ................... 19
2 RTSP Use Cases ...................................... 19
2.1 On-demand Playback of Stored Content ................ 20
2.2 Unicast distribution of Live Content ................ 21
2.3 On-demand Playback using Multicast .................. 21
2.4 Inviting a RTSP server into a conference ............ 22
2.5 Live Content using Multicast ........................ 23
3 Protocol Parameters ................................. 23
3.1 RTSP Version ........................................ 23
3.2 RTSP URI ............................................ 23
3.3 Session Identifiers ................................. 25
3.4 SMPTE Relative Timestamps ........................... 25
3.5 Normal Play Time .................................... 26
3.6 Absolute Time ....................................... 27
3.7 Feature-tags ........................................ 27
3.8 Entity Tags ......................................... 27
4 RTSP Message ........................................ 28
4.1 Message Types ....................................... 28
4.2 Message Headers ..................................... 28
4.3 Message Body ........................................ 28
4.4 Message Length ...................................... 29
5 General Header Fields ............................... 29
6 Request ............................................. 30
6.1 Request Line ........................................ 30
6.2 Request Header Fields ............................... 31
7 Response ............................................ 32
7.1 Status-Line ......................................... 32
7.1.1 Status Code and Reason Phrase ....................... 33
7.1.2 Response Header Fields .............................. 34
8 Entity .............................................. 34
8.1 Entity Header Fields ................................ 34
8.2 Entity Body ......................................... 34
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9 Connections ......................................... 35
9.1 Reliability and Acknowledgements .................... 35
9.2 Using Connections ................................... 38
9.3 Closing Connections ................................. 39
9.4 Timing Out Connections and RTSP Messages ............ 40
9.5 Use of IPv6 ......................................... 40
10 Capability Handling ................................. 41
11 Method Definitions .................................. 42
11.1 OPTIONS ............................................. 43
11.2 DESCRIBE ............................................ 44
11.3 SETUP ............................................... 46
11.3.1 Changing Transport Parameters ....................... 48
11.4 PLAY ................................................ 49
11.5 PAUSE ............................................... 53
11.6 TEARDOWN ............................................ 57
11.7 GET_PARAMETER ....................................... 58
11.8 SET_PARAMETER ....................................... 59
11.9 REDIRECT ............................................ 60
12 Embedded (Interleaved) Binary Data .................. 62
13 Status Code Definitions ............................. 64
13.1 Success 1xx ......................................... 64
13.1.1 100 Continue ........................................ 64
13.2 Success 2xx ......................................... 64
13.3 Redirection 3xx ..................................... 64
13.3.1 300 Multiple Choices ................................ 65
13.3.2 301 Moved Permanently ............................... 65
13.3.3 302 Found ........................................... 65
13.3.4 303 See Other ....................................... 65
13.3.5 304 Not Modified .................................... 65
13.3.6 305 Use Proxy ....................................... 66
13.4 Client Error 4xx .................................... 66
13.4.1 400 Bad Request ..................................... 66
13.4.2 405 Method Not Allowed .............................. 66
13.4.3 451 Parameter Not Understood ........................ 66
13.4.4 452 reserved ........................................ 67
13.4.5 453 Not Enough Bandwidth ............................ 67
13.4.6 454 Session Not Found ............................... 67
13.4.7 455 Method Not Valid in This State .................. 67
13.4.8 456 Header Field Not Valid for Resource ............. 67
13.4.9 457 Invalid Range ................................... 67
13.4.10 458 Parameter Is Read-Only .......................... 67
13.4.11 459 Aggregate Operation Not Allowed ................. 67
13.4.12 460 Only Aggregate Operation Allowed ................ 68
13.4.13 461 Unsupported Transport ........................... 68
13.4.14 462 Destination Unreachable ......................... 68
13.4.15 463 Destination Prohibited .......................... 68
13.4.16 470 Connection Authorization Required ............... 68
13.4.17 471 Connection Credentials not accepted ............. 68
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13.5 Server Error 5xx .................................... 68
13.5.1 551 Option not supported ............................ 68
14 Header Field Definitions ............................ 69
14.1 Accept .............................................. 71
14.2 Accept-Credentials .................................. 71
14.3 Accept-Encoding ..................................... 75
14.4 Accept-Language ..................................... 75
14.5 Accept-Ranges ....................................... 75
14.6 Allow ............................................... 76
14.7 Authorization ....................................... 76
14.8 Bandwidth ........................................... 76
14.9 Blocksize ........................................... 76
14.10 Cache-Control ....................................... 77
14.11 Connection .......................................... 79
14.12 Connection-Credentials .............................. 79
14.13 Content-Base ........................................ 80
14.14 Content-Encoding .................................... 80
14.15 Content-Language .................................... 80
14.16 Content-Length ...................................... 80
14.17 Content-Location .................................... 80
14.18 Content-Type ........................................ 80
14.19 CSeq ................................................ 81
14.20 Date ................................................ 81
14.21 ETag ................................................ 81
14.22 Expires ............................................. 82
14.23 From ................................................ 83
14.24 Host ................................................ 83
14.25 If-Match ............................................ 83
14.26 If-Modified-Since ................................... 83
14.27 If-None-Match ....................................... 83
14.28 Last-Modified ....................................... 84
14.29 Location ............................................ 84
14.30 Proxy-Authenticate .................................. 84
14.31 Proxy-Require ....................................... 84
14.32 Proxy-Supported ..................................... 84
14.33 Public .............................................. 85
14.34 Range ............................................... 86
14.35 Referer ............................................. 88
14.36 Retry-After ......................................... 88
14.37 Require ............................................. 88
14.38 RTP-Info ............................................ 89
14.39 Scale ............................................... 91
14.40 Speed ............................................... 91
14.41 Server .............................................. 92
14.42 Session ............................................. 92
14.43 Supported ........................................... 94
14.44 Timestamp ........................................... 94
14.45 Transport ........................................... 95
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14.46 Unsupported ......................................... 99
14.47 User-Agent .......................................... 100
14.48 Vary ................................................ 100
14.49 Via ................................................. 100
14.50 WWW-Authenticate .................................... 100
15 Proxies ............................................. 100
16 Caching ............................................. 101
17 Examples ............................................ 102
17.1 Media on Demand (Unicast) ........................... 102
17.2 Streaming of a Container file ....................... 105
17.3 Single Stream Container Files ....................... 108
17.4 Live Media Presentation Using Multicast ............. 110
17.5 Capability Negotiation .............................. 111
18 Security Framework .................................. 112
18.1 RTSP and HTTP Authentication ........................ 112
18.2 RTSP over TLS ....................................... 112
18.3 Security and Proxies ................................ 113
18.3.1 Accept-Credentials .................................. 114
18.3.2 User approved TLS procedure ......................... 115
19 Syntax .............................................. 117
19.1 Base Syntax ......................................... 117
19.2 RTSP Protocol Definition ............................ 119
19.2.1 Generic Protocol elements ........................... 119
19.2.2 Message Syntax ...................................... 120
19.2.3 Header Syntax ....................................... 124
19.3 SDP extension Syntax ................................ 129
20 Security Considerations ............................. 130
20.1 Remote denial of Service Attack ..................... 132
21 IANA Considerations ................................. 132
21.1 Feature-tags ........................................ 133
21.1.1 Description ......................................... 133
21.1.2 Registering New Feature-tags with IANA .............. 133
21.1.3 Registered entries .................................. 133
21.2 RTSP Methods ........................................ 134
21.2.1 Description ......................................... 134
21.2.2 Registering New Methods with IANA ................... 134
21.2.3 Registered Entries .................................. 134
21.3 RTSP Status Codes ................................... 134
21.3.1 Description ......................................... 134
21.3.2 Registering New Status Codes with IANA .............. 135
21.3.3 Registered Entries .................................. 135
21.4 RTSP Headers ........................................ 135
21.4.1 Description ......................................... 135
21.4.2 Registering New Headers with IANA ................... 135
21.4.3 Registered entries .................................. 136
21.5 Transport Header registries ......................... 136
21.5.1 Transport Protocols ................................. 136
21.5.2 Profile ............................................. 137
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21.5.3 Lower Transport ..................................... 137
21.5.4 Transport modes ..................................... 138
21.5.5 Transport Parameters ................................ 138
21.6 Cache Directive Extensions .......................... 139
21.7 Accept-Credentials policies ......................... 139
21.8 URI Schemes ......................................... 140
21.9 SDP attributes ...................................... 140
A RTSP Protocol State Machine ......................... 141
A.1 States .............................................. 141
A.2 State variables ..................................... 142
A.3 Abbreviations ....................................... 142
A.4 State Tables ........................................ 142
B Media Transport Alternatives ........................ 145
B.1 RTP ................................................. 146
B.1.1 AVP ................................................. 146
B.1.2 AVP/UDP ............................................. 146
B.1.3 AVP/TCP ............................................. 147
B.1.4 AVPF ................................................ 148
B.1.5 SAVP ................................................ 148
B.1.6 Handling NPT Jumps in the RTP Media Layer ........... 148
B.1.7 Handling RTP Timestamps after PAUSE ................. 151
B.1.8 RTSP / RTP Integration .............................. 153
B.1.9 Scaling with RTP .................................... 153
B.1.10 Maintaining NPT synchronization with RTP
timestamps .......................................... 153
B.1.11 Continuous Audio .................................... 153
B.1.12 Multiple Sources in an RTP Session .................. 153
B.1.13 Usage of SSRCs and the RTCP BYE Message During an
RTSP Session ........................................ 154
B.2 Future Additions .................................... 154
C Use of SDP for RTSP Session Descriptions ............ 155
C.1 Definitions ......................................... 155
C.1.1 Control URI ......................................... 155
C.1.2 Media Streams ....................................... 156
C.1.3 Payload Type(s) ..................................... 157
C.1.4 Format-Specific Parameters .......................... 157
C.1.5 Range of Presentation ............................... 157
C.1.6 Time of Availability ................................ 158
C.1.7 Connection Information .............................. 158
C.1.8 Entity Tag .......................................... 158
C.2 Aggregate Control Not Available ..................... 159
C.3 Aggregate Control Available ......................... 160
C.4 RTSP external SDP delivery .......................... 161
D Minimal RTSP implementation ......................... 161
D.1 Minimal Core Implementation ......................... 161
D.2 The Basic Playback Feature Support .................. 162
D.2.1 Client .............................................. 162
D.2.2 Server .............................................. 163
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D.2.3 Proxy ............................................... 163
D.3 Secure Transport .................................... 163
D.4 Old Implementation Text ............................. 163
D.5 Client .............................................. 164
D.5.1 Basic Playback ...................................... 164
D.5.2 Authentication-enabled .............................. 165
D.6 Server .............................................. 165
D.6.1 Basic Playback ...................................... 166
D.6.2 Authentication-enabled .............................. 166
E Requirements for Unreliable Transport of RTSP
messages ............................................ 167
F Backwards Compatibility Considerations .............. 168
F.1 Play Request in Play mode ........................... 168
F.2 Using Persistent Connections ........................ 168
G Open Issues ......................................... 169
H Changes ............................................. 169
H.1 Changes needing to be updated ....................... 175
I Author Addresses .................................... 175
J Contributors ........................................ 176
K Acknowledgements .................................... 176
L Normative References ................................ 177
M Informative References .............................. 179
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1 Introduction
1.1 RTSP Specification Update
This memorandum specifies RTSP 1.1 which is an update of RTSP 1.0, a
proposed standard defined in RFC 2326 [24]. The goal of this version
is to correct the many flaws that have been identified in RTSP 1.0
since its publication. The corrections are such that full backwards
compatibility was impossible. Thus a new version was decided the most
appropriate solution to get a more functional protocol. There are no
plans to revise RTSP 1.0. Appendix H catalogs the changes of this
version in relation to RTSP 1.0.
A few open issues still remain to be resolved, and are listed in
appendix G. These are intended to be close quickly.
A list of bugs against RFC 2326 is available at
"http://rtspspec.sourceforge.net". These bugs should be taken into
account when reading this memorandum. Input on the unresolved bugs
and other issues can be sent via e-mail to the MMUSIC WG's mailing
list mmusic@ietf.org and the authors.
RTSP 1.1 is reduced in functionality in regards to RTSP 1.0 and aims
at specifying the RTSP core, functionality and rules for extensions,
and basic interaction with the media delivery protocol RTP.
Any other functionality would be need to be published as extension
documents. The Working group has discussed a number of different
proposals to extensions and is currently working on:
o NAT and FW traversal mechanisms for RTSP are described in a
document called "How to make Real-Time Streaming Protocol
(RTSP) traverse Network Address Translators (NAT) and interact
with Firewalls." [25].
1.2 Purpose
The Real-Time Streaming Protocol (RTSP) establishes and controls one
or several time-synchronized streams of continuous media such as
audio and video. Put simply, RTSP acts as a "network remote control"
for multimedia servers.
There is no notion of an RTSP connection in the protocol. Instead, an
RTSP server maintains a session labeled by an identifier to associate
groups of media streams and their states. An RTSP session is not tied
to a transport-level connection such as a TCP connection. During a
session, a client may open and close many reliable transport
connections to the server to issue RTSP requests for that session.
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This memorandum describes the use of RTSP over a reliable connection
based transport level protocol such as TCP. RTSP may be implemented
over an unreliable connectionless transport protocol such as UDP.
While nothing in RTSP precludes this, additional definition of this
problem area needs to be handled as an extension to the core
specification.
The mechanisms of RTSP's operation over UDP were left out
of this spec. because they were poorly defined in RFC 2326
[24] and the tradeoff in size and complexity of this
memorandum for a small gain in a limited problem space was
not deemed justifiable.
The set of streams to be controlled in an RTSP session is defined by
a presentation description. This memorandum does not define a format
for the presentation description. However appendix C defines how SDP
[1] is used for this purpose. The streams controlled by RTSP may use
RTP [2] for their data transport, but the operation of RTSP does not
depend on the transport mechanism used to carry continuous media.
RTSP is intentionally similar in syntax and operation to HTTP/1.1 [3]
so that extension mechanisms to HTTP can in most cases also be added
to RTSP. However, RTSP differs in a number of important aspects from
HTTP:
o RTSP introduces a number of new methods and has a different
protocol identifier.
o RTSP has the notion of a session built into the protocol.
o An RTSP server needs to maintain state by default in almost
all cases, as opposed to the stateless nature of HTTP.
o Both an RTSP server and client can issue requests.
o Data is usually carried out-of-band by a different protocol.
Session descriptions returned in a DESCRIBE response (see
Section 11.2) and interleaving of RTP with RTSP over TCP are
exceptions to this rule (see Section 12).
o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO
8859-1, consistent with HTML internationalization efforts
[26].
o The Request-URI always contains the absolute URI. Because of
backward compatibility with a historical blunder, HTTP/1.1 [3]
carries only the absolute path in the request and puts the
host name in a separate header field.
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This makes "virtual hosting" easier, where a single
host with one IP address hosts several document trees.
The protocol supports the following operations:
Retrieval of media from media server: The client can either
request a presentation description via RTSP DESCRIBE, HTTP
or some other method. If the presentation is being
multicast, the presentation description contains the
multicast addresses and ports to be used for the continuous
media. If the presentation is to be sent only to the client
via unicast, the client provides the destination of
necessity.
Invitation of a media server to a conference: A media server can
be "invited" to join an existing conference to play back
media into the presentation. This mode is useful for
example distributed teaching applications. Several parties
in the conference may take turns "pushing the remote
control buttons".
RTSP requests may be handled by proxies, tunnels and caches as in
HTTP/1.1 [3].
1.3 Notational Conventions
Since many of the definitions and syntax are identical to HTTP/1.1,
this specification only points to the section where they are defined
rather than copying it. For brevity, [HX.Y] is to be taken to refer
to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [3]).
All the mechanisms specified in this document are described in both
prose and the Augmented Backus-Naur form (ABNF) described in detail
in RFC 2234 [4].
Indented and smaller-type paragraphs are used to provide informative
background and motivation. This is intended to give readers who were
not involved with the formulation of the specification an
understanding of why things are the way they are in RTSP.
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
document are to be interpreted as described in RFC 2119 [5].
The word, "unspecified" is used to indicate functionality or features
that are not defined in this specification. Such functionality cannot
be used in a standardized manner without further definition in an
extension specification to RTSP.
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1.4 Terminology
Some of the terminology has been adopted from HTTP/1.1 [3]. Terms not
listed here are defined as in HTTP/1.1.
Aggregate control: The concept of controlling multiple streams
using a single timeline, generally maintained by the
server. A client, for example, uses aggregate control when
it issues a single play or pause message to simultaneously
control both the audio and video in a movie.
Aggregate control URI: The URI used in an RTSP request to refer
to and control an aggregated session. It normally, but not
always, corresponds to the presentation URI specified in
the session description. See Section 11.3 for more
information.
Conference: a multiparty, multimedia presentation, where "multi"
implies greater than or equal to one.
Client: The client requests media service from the media server.
Connection: A transport layer virtual circuit established
between two programs for the purpose of communication.
Container file: A file which may contain multiple media streams
which often constitutes a presentation when played
together. The concept of a container file is not embedded
in the protocol. However, RTSP servers may offer aggregate
control on the media streams within these files.
Continuous media: Data where there is a timing relationship
between source and sink; that is, the sink needs to
reproduce the timing relationship that existed at the
source. The most common examples of continuous media are
audio and motion video. Continuous media can be real-time
(interactive or conversational), where there is a "tight"
timing relationship between source and sink, or streaming
(playback), where the relationship is less strict.
Entity: The information transferred as the payload of a request
or response. An entity consists of meta-information in the
form of entity-header fields and content in the form of an
entity-body, as described in Section 8.
Feature-tag: A tag representing a certain set of functionality,
i.e. a feature.
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Live: Normally used to describe a presentation or session with
media coming from an ongoing event. This generally results
in that the session has a unbound or only loosely defined
duration, and that no seek operations are possible.
Media initialization: Datatype/codec specific initialization.
This includes such things as clock rates, color tables,
etc. Any transport-independent information which is
required by a client for playback of a media stream occurs
in the media initialization phase of stream setup.
Media parameter: Parameter specific to a media type that may be
changed before or during stream playback.
Media server: The server providing playback services for one or
more media streams. Different media streams within a
presentation may originate from different media servers. A
media server may reside on the same host or on a different
host from which the presentation is invoked.
Media server indirection: Redirection of a media client to a
different media server.
(Media) stream: A single media instance, e.g., an audio stream
or a video stream as well as a single whiteboard or shared
application group. When using RTP, a stream consists of all
RTP and RTCP packets created by a source within an RTP
session.
Message: The basic unit of RTSP communication, consisting of a
structured sequence of octets matching the syntax defined
in Section 19 and transmitted over a connection or a
connectionless transport.
Non-Aggregated Control: Control of a single media stream. Only
possible in RTSP sessions with a single media.
Participant: Member of a conference. A participant may be a
machine, e.g., a playback server.
Presentation: A set of one or more streams presented to the
client as a complete media feed and described by a
presentation description as defined below. Presentations
with more than one media stream is often handled in RTSP
under aggregate control.
Presentation description: A presentation description contains
information about one or more media streams within a
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presentation, such as the set of encodings, network
addresses and information about the content. Other IETF
protocols such as SDP (RFC 2327 [1]) use the term "session"
for a presentation. The presentation description may take
several different formats, including but not limited to the
session description protocol format, SDP.
Response: An RTSP response. If an HTTP response is meant, that
is indicated explicitly.
Request: An RTSP request. If an HTTP request is meant, that is
indicated explicitly.
Request-URI: The URI used in a request to indicate the resource
on which the request is to be performed.
RTSP agent: Refers to either an RTSP client, an RTSP server, or
an RTSP Proxy. In this specification, there are many
capabilities that are common to these three entities such
as the capability to send requests or receive responses.
This term will be used when describing functionality that
is applicable to all three of these entities.
RTSP session: A stateful abstraction upon which the main control
methods of RTSP operate. An RTSP session is a server
entity; it is created, maintained and destroyed by the
server. It is established by an RTSP server upon the
completion of a successful SETUP request (when 200 OK
response is sent) and is labelled by a session identifier
at that time. The session exists until timed out by the
server or explicitly removed by a TEARDOWN request. An RTSP
session is a stateful entity; an RTSP server maintains an
explicit session state machine (see Appendix A) where most
state transitions are triggered by client requests. The
existence of a session implies the existence of state about
the session's media streams and their respective transport
mechanisms. A given session can have zero or more media
streams associated with it. An RTSP server uses the session
to aggregate control over multiple media streams.
Transport initialization: The negotiation of transport
information (e.g., port numbers, transport protocols)
between the client and the server.
URI: Universal Resource Identifier, see RFC 3986 [10]. In RTSP
the used URIs are as general rule in fact URL's as they
gives an location for the resource. As URLs are a subset of
URIs, they will be referred to as URIs to cover also the
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cases when an RTSP URI would not be an URL.
URL: Universal Resource Locator, is an URI which identifies the
resource through its primary access mechanism, rather than
identifying the resource by name or by some other
attribute(s) of that resource.
1.5 Protocol Properties
RTSP has the following properties:
Extendable: New methods and parameters can be easily added to
RTSP.
Easy to parse: RTSP can be parsed by standard HTTP or MIME
parsers.
Secure: RTSP re-uses web security mechanisms, either at the
transport level (TLS, RFC 2246 [6]) or within the protocol
itself. All HTTP authentication mechanisms such as basic
(RFC 2616 [3]) and digest authentication (RFC 2617 [7]) are
directly applicable.
Transport-independent: RTSP does not preclude the use of an
unreliable datagram protocol (UDP) (RFC 768 [8]) as it
would be possible to implement application-level
reliability. The use of a connectionless datagram protocol
such as UDP requires additional definition that may be
provided as extensions to the core RTSP specification. The
usage of the reliable stream protocol TCP (RFC 793 [9]) and
secured reliable stream protocol TLS over TCP [6] is what
is currently defined as transport protocol of RTSP
messages.
Multi-server capable: Each media stream within a presentation
can reside on a different server. The client automatically
establishes several concurrent control sessions with the
different media servers. Media synchronization is
performed at the transport level.
Separation of stream control and conference initiation: Stream
control is divorced from inviting a media server to a
conference. In particular, SIP [27] or H.323 [28] may be
used to invite a server to a conference.
Suitable for professional applications: RTSP supports frame-
level accuracy through SMPTE time stamps to allow remote
digital editing.
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Presentation description neutral: The protocol does not impose a
particular presentation description or metafile format and
can convey the type of format to be used. However, the
presentation description is required to contain at least
one RTSP URI.
Proxy and firewall friendly: The protocol should be readily
handled by both application and transport-layer (SOCKS
[29]) firewalls. A firewall may need to understand the
SETUP method to open a "hole" for the media stream.
HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so
that the existing infrastructure can be reused. This
infrastructure includes PICS (Platform for Internet Content
Selection [30,31]) for associating labels with content.
However, RTSP does not just add methods to HTTP since the
controlling continuous media requires server state in most
cases.
Appropriate server control: If a client can start a stream, it
needs to be able to stop a stream. Servers should not start
streaming to clients in such a way that clients cannot stop
the stream.
Transport negotiation: The client can negotiate the transport
method prior to actually needing to process a continuous
media stream.
1.6 Extending RTSP
Since not all media servers have the same functionality, media
servers by necessity will support different sets of requests. For
example:
o A server may not be capable of seeking (absolute positioning)
if it is to support live events only.
o Some servers may not support setting stream parameters and
thus not support GET_PARAMETER and SET_PARAMETER.
o Some server may support an RTSP extension.
A server SHOULD implement all header fields described in Section 14.
It is up to the creators of presentation descriptions not to ask the
impossible of a server. This situation is similar in HTTP/1.1 [3],
where the methods described in [H19.5] are not likely to be supported
across all servers.
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RTSP can be extended in three ways, listed here in order of the
magnitude of changes supported:
o Existing methods can be extended with new parameters, e.g.
headers, as long as these parameters can be safely ignored by
the recipient. If the client needs negative acknowledgement
when a method extension is not supported, a tag corresponding
to the extension may be added in the Require: field (see
Section 14.37).
o New methods can be added. If the recipient of the message does
not understand the request, it responds with error code 501
(Not Implemented) and the sender should not attempt to use
this method again. A client may also use the OPTIONS method to
inquire about methods supported by the server. The server MUST
list the methods it supports using the Public response header.
o A new version of the protocol can be defined, allowing almost
all aspects (except the position of the protocol version
number) to change.
The basic capability discovery mechanism can be used to both discover
support for a certain feature and to ensure that a feature is
available when performing a request. For detailed explanation of this
see section 10.
1.7 Overall Operation
Each presentation and media stream is identified by an RTSP URI. The
overall presentation and the properties of the media the presentation
is made up of are defined by a presentation description file, the
format of which is outside the scope of this specification. The
presentation description file may be obtained by the client using
HTTP or other means such as email and may not necessarily be stored
on the media server.
For the purposes of this specification, a presentation description is
assumed to describe one or more presentations, each of which
maintains a common time axis. For simplicity of exposition and
without loss of generality, it is assumed that the presentation
description contains exactly one such presentation. A presentation
may contain several media streams.
The presentation description file contains a description of the media
streams making up the presentation, including their encodings,
language, and other parameters that enable the client to choose the
most appropriate combination of media. In this presentation
description, each media stream that is individually controllable by
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RTSP is identified by an RTSP URI, which points to the media server
handling that particular media stream and names the stream stored on
that server. Several media streams can be located on different
servers; for example, audio and video streams can be split across
servers for load sharing. The description also enumerates which
transport methods the server is capable of.
Besides the media parameters, the network destination address and
port need to be determined. Several modes of operation can be
distinguished:
Unicast: The media is transmitted to the source of the RTSP
request, with the port number chosen by the client.
Alternatively, the media is transmitted on the same
reliable stream as RTSP.
Multicast, server chooses address: The media server picks the
multicast address and port. This is the typical case for a
live or near-media-on-demand transmission.
Multicast, client chooses address: If the server is to
participate in an existing multicast conference, the
multicast address, port and encryption key are given by the
conference description, established by means outside the
scope of this specification, for example by a SIP created
conference.
1.8 RTSP States
RTSP controls a stream which may be sent via a separate protocol,
independent of the control channel. For example, RTSP control may be
transported on a TCP connection while the media data is conveyed via
UDP. Thus, data delivery continues even if no RTSP requests are
received by the media server. Also, during its lifetime, a single
media stream may be controlled by RTSP requests issued sequentially
on different TCP connections. Therefore, the server needs to maintain
"session state" to be able to correlate RTSP requests with a stream.
The state transitions are described in Appendix A.
Many methods in RTSP do not contribute to state. However, the
following play a central role in defining the allocation and usage of
stream resources on the server: SETUP, PLAY, PAUSE, REDIRECT, and
TEARDOWN.
SETUP: Causes the server to allocate resources for a stream and
create an RTSP session.
PLAY: Starts data transmission on a stream allocated via SETUP.
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PAUSE: Temporarily halts a stream without freeing server
resources.
REDIRECT: Indicates that the session should be moved to new
server / location
TEARDOWN: Frees resources associated with the stream. The RTSP
session ceases to exist on the server.
RTSP methods that contribute to state use the Session header field
(Section 14.42) to identify the RTSP session whose state is being
manipulated. The server generates session identifiers in response to
SETUP requests (Section 11.3).
1.9 Relationship with Other Protocols
RTSP has some overlap in functionality with HTTP. It also may
interact with HTTP in that the initial contact with streaming content
is often to be made through a web page. The current protocol
specification aims to allow different hand-off points between a web
server and the media server implementing RTSP. For example, the
presentation description can be retrieved using HTTP or RTSP, which
reduces round trips in web-browser-based scenarios, yet also allows
for stand alone RTSP servers and clients which do not rely on HTTP at
all. However, RTSP differs fundamentally from HTTP in that most data
delivery takes place out-of-band in a different protocol. HTTP is an
asymmetric protocol where the client issues requests and the server
responds. In RTSP, both the media client and media server can issue
requests. RTSP requests are also stateful; they may set parameters
and continue to control a media stream long after the request has
been acknowledged.
Re-using HTTP functionality has advantages in at least two
areas, namely security and proxies. The requirements are
very similar, so having the ability to adopt HTTP work on
caches, proxies and authentication is valuable.
RTSP assumes the existence of a presentation description format that
can express both static and temporal properties of a presentation
containing several media streams. Session Description Protocol (SDP)
[1] is generally the format of choice; however, RTSP is not bound to
it. For data delivery, most real-time media will use RTP as a
transport protocol. While RTSP works well with RTP, it is not tied to
RTP.
2 RTSP Use Cases
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This section describes the most important and considered use cases
for RTSP. They are listed in descending order of importance in
regards to ensuring that all necessary functionality is present. This
specification does only fully support usage of the two first. Also in
these first two cases, there are special cases or exceptions that are
not supported without extensions, e.g. the redirection of media to
another address than the controlling entity.
2.1 On-demand Playback of Stored Content
An RTSP capable server stores content suitable for being streamed to
a client. A client desiring playback of any of the stored content
then uses RTSP to set up and configure the media transport required
for the desired content. Then RTSP is used to initiate, halt and
manipulate the actual transmission (playout) of the content. There
are also requirement on being able to use RTSP to carry necessary
description and synchronization information for the content. The
above high level description can be broken down into a number of
functionalities that RTSP needs to be capable of.
Presentation Description: The possibility to carry
initialization information about the presentation
(content), for example, which media codec(s) that are
needed for the content. Other information that are
important; how many media stream that the presentation
contains; what transport protocols used for the media
streams; and identifiers for these media streams. This
information is required before setup of the content is
possible. The information is also needed by the client to
determine if it is capable at all to support the content.
This information is not required to be sent using RTSP,
instead other external protocols can be utilized to
transport presentation descriptions. Two good examples are
the use of HTTP [3] or email to fetch or receive
presentation descriptions like SDP [1]. .XP Setup:
Performing setup of some or all of the media streams in a
presentation. The setup itself consist of determining which
protocols for media transport to use; the necessary
parameters for the protocol, like addresses and ports. .XP
Control of Transmission: After the necessary media streams
has been established the client can request the server to
start transmitting the content. There is need to allow the
client to at arbitrary times start or stop the transmission
of the content. There are also exist need to be able to
start the transmission at an any point in the timeline of
the presentation. .XP Synchronization: For media transport
protocols like RTP [16] it might be beneficial to carry
synchronization information within RTSP. Either due to the
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lack of inter media synchronization within the protocol
itself, or the potential delay before the synchronization
is established (which is the case for RTP when using RTCP).
.XP Termination There is also need to be able to terminate
the established contexts.
For this use cases there is a number of assumption about how it
works. These are listed below:
On-Demand content: The content available is stored at the server
and can be accessed at any time during a time period when
it is intended to be available. .XP Independent sessions: A
server is capable of serving a number of clients
simultaneously, including from the same piece of content at
different points in that presentations time-line. .XP
Unicast Transport: Content for each individual client is
transmitted to them using unicast traffic.
It is also possible to redirect the media traffic to another
destination than where the entity controlling traffic uses.
However allowing this without appropriate mechanisms for
checking that the destination approves of this is allows for
distributed denial of service attacks (DDoS).
2.2 Unicast distribution of Live Content
This use cases is not that different from the above on-demand content
case (see section 2.1. The difference is really the restriction the
content itself establish. Live content is continuously distributed as
it becomes available from a source, i.e. the main difference to on-
demand is that one starts distributing content before the end of it
has become available to the server. In many cases the consumer of
live content is only interested in consuming what is actually happens
"now", i.e. very similar to broadcast TV. However in this case it is
assumed that there exist no broadcast or multicast channel to the
users, and instead the server functions as a distribution node,
sending the same content to multiple receivers, using unicast traffic
between server and client. This unicast traffic and the transport
parameters are individually negotiated for each receiving client.
Another aspect of live content is that it has often very limited time
of availability, as it is only is available for the duration of the
event the content covers. A example of such a live content could for
example be a music concert, which lasts 2 hour and starts at a
predetermined time. Thus there is need to announce when and for how
long the live content is available.
2.3 On-demand Playback using Multicast
It is possible to use RTSP to request that media is delivered to a
multicast group. The entity setting up the session (the controller)
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will then control when and what media that is delivered to the group.
Also this use case has some potential for denial of service attacks,
in this case flooding any multicast group. Therefore there is need
for a mechanism indicating that the group actually accepts the
traffic from the RTSP server. An open issue in this use case is how
one ensures that all receivers listening to the multicast or
broadcast receives the session presentation configuring the
receivers.
2.4 Inviting a RTSP server into a conference
If one has an established conference or group session, it is possible
to have a RTSP server distribute media to the whole group. The
transmission to the group is simplest controlled by a single
participant or leader of the conference. Shared control might be
possible, but would require further investigation and possibly
extensions. There are some protocol mechanisms missing for this
scenario. For reasonable complexity in the media transmission stage,
this use case assumes that there exist either multicast or a
conference focus that redistribute media to all participants. In some
more detail, this use case is intended to be able to handle the
following scenario: A conference leader or participant (from here
called the controller) has some pre-stored content on a RTSP server
that he likes to share with the group. The controller sets up an RTSP
session at the streaming server for the content the controller likes
to share. The session description for the content is retrieved by the
controller. The media destination for the media content is sent to
the shared multicast group or conference focus. When desired by the
controller, he/she can start and stop the transmission of the media
to the conference group. There are several issues with this use case
that is not solved by this core specification for RTSP:
o Denial of service threat, to avoid a RTSP server from being a
unknowing participant of a denial of service attack the server
needs to be able to verify the destinations acceptance for the
media. Such a mechanism does not yet exist that can be used to
verify the approval to received media, instead only policies
can be used, which can be made to work in controlled
environments. .IP o 2 The problem of distributing the
presentation description to all participants in the group. To
enable a media receiver to decode the content correctly the
media configuration information will need to be distributed
reliable to all participants. This will most likely require
support from an external protocol. .IP o 2 Passing the
control. If it is desired to be able to pass the control of
the RTSP session between the participants some support will be
required by an external protocol for the necessary exchange of
state information and possibly floor control of who is
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controlling the RTSP session.
So if there interest in this use case further work on the necessary
extensions has to be performed.
2.5 Live Content using Multicast
This use case does in its simplest form do not require any use of
RTSP at all. This is what multicast conferences being announced with
SAP and SDP are intended to handle. However in use cases where more
advance features like access control to the multicast session is
desired, RTSP could be used for session establishment. A client
desiring to join a live multicasted media session with cryptographic
(encryption) access control could use RTSP in the following way. The
source of the session, announces the session and gives all interested
to join, a RTSP URI. The client connects to the server and requests
the presentation description allowing for configuration the
reception. In this step it is possible to use secured transport for
the client, and also desired levels of authentication, for example
for charging purposes or simply access control. An RTSP link also
allows for load balancing between multiple servers. However if this
was the only thing that occurred it could probably be solved as
simply using HTTP. However for session where the sender likes to keep
track of each individual receiver during the session, and possibly
use this side channel for pushing out key-updates or other side
information that is desirable to be done on a per receiver basis, and
the receivers are not know prior to the session start, the state
establishment that RTSP provides can be beneficial. In this case a
client would establish a RTSP session to the multicast group. The
RTSP server will not transmit any media, instead it will simply point
to the multicast group. However the client and server will be able to
keep the session alive for as long as the receiver participates in
the session. Thus enabling, for example server to client pushes of
updates. This use cases will most likely not be able to actually
implement without some extensions in relation to the server to client
push mechanism. Here a method like ANNOUNCE (see RFC 2326 [24] might
be suitable, however it will require a RTSP extension to revive the
method.
3 Protocol Parameters
3.1 RTSP Version
HTTP Specification Section [H3.1] applies, with HTTP replaced by
RTSP. This specification defines version 1.1 of RTSP.
3.2 RTSP URI
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The "rtsp", "rtsps" schemes are used to refer to network resources
via the RTSP protocol. This section defines the scheme-specific
syntax and semantics for RTSP URIs. The RTSP URI is case sensitive.
An URI scheme "rtspu" was defined in RFC 2326 for transport of RTSP
messages over unreliable transport (UDP) and is currently deprecated
and reserved, and MUST NOT be used . See Appendix E for further
information.
Informative RTSP URI syntax:
rtsp[u|s]://host[:port]/abspath[?query]#fragment
See section 19.2.1 for the formal definition of the RTSP URI syntax.
The fragment identifier is used as defined in section 4.1 of [10],
i.e. the fragment is to be stripped from the URI by the requestor and
not included in the request. The user agent also needs to interpret
the value of the fragment based on the media type the request relates
to, i.e. the media type indicated in Content-Type header in the
response to DESCRIBE.
The syntax of any URI query string is unspecified and responder
(usually the server) specific. As it is from the requestor an opaque
string, it needs to be handled as such.
The URI scheme rtsp requires that commands are issued via a reliable
protocol (within the Internet, TCP), while the scheme rtsps
identifies a reliable transport using secure transport (TLS [6]).
If the no port number is provided in the URI, port number 554 SHALL
be used. The semantics are that the identified resource can be
controlled by RTSP at the server listening for TCP (scheme "rtsp")
connections on that port of host, and the Request-URI for the
resource is rtsp_URI. For the scheme rtsps the TCP and UDP port 322
is registered and SHALL be assumed.
The use of IP addresses in URIs SHOULD be avoided whenever possible
(see RFC 1924 [11]). This specification updates the RTSP URI scheme
to allow for literal IPv6 addresses using the host specification in
RFC 2732 [12].
Note: Using qualified domain names in any URI is one
requirement for making it possible for RTSP 1.0 (RFC 2326)
implementations of RTSP to use IPv6.
A presentation or a stream is identified by a textual media
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identifier, using the character set and escape conventions [H3.2] of
URIs (RFC 3986 [10]). URIs may refer to a stream or an aggregate of
streams, i.e., a presentation. Accordingly, requests described in
Section 11 can apply to either the whole presentation or an
individual stream within the presentation. Note that some request
methods can only be applied to streams, not presentations and vice
versa.
For example, the RTSP URI:
rtsp://media.example.com:554/twister/audiotrack
may identify the audio stream within the presentation "twister",
which can be controlled via RTSP requests issued over a TCP
connection to port 554 of host media.example.com
Also, the RTSP URI:
rtsp://media.example.com:554/twister
identifies the presentation "twister", which may be composed of audio
and video streams.
This does not imply a standard way to reference streams in
URIs. The presentation description defines the hierarchical
relationships in the presentation and the URIs for the
individual streams. A presentation description may name a
stream "a.mov" and the whole presentation "b.mov".
The path components of the RTSP URI are opaque to the client and do
not imply any particular file system structure for the server.
This decoupling also allows presentation descriptions to be
used with non-RTSP media control protocols simply by
replacing the scheme in the URI.
3.3 Session Identifiers
Session identifiers are strings of any arbitrary length. A session
identifier MUST be chosen randomly and MUST be at least eight
characters long to make guessing it more difficult. (See Section 20.)
3.4 SMPTE Relative Timestamps
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A SMPTE relative timestamp expresses time relative to the start of
the clip. Relative timestamps are expressed as SMPTE time codes for
frame-level access accuracy. The time code has the format
hours:minutes:seconds:frames.subframes,
with the origin at the start of the clip. The default smpte format is
"SMPTE 30 drop" format, with frame rate is 29.97 frames per second.
Other SMPTE codes MAY be supported (such as "SMPTE 25") through the
use of alternative use of "smpte time". For the "frames" field in the
time value can assume the values 0 through 29. The difference between
30 and 29.97 frames per second is handled by dropping the first two
frame indices (values 00 and 01) of every minute, except every tenth
minute. If the frame value is zero, it may be omitted. Subframes are
measured in one-hundredth of a frame.
Examples:
smpte=10:12:33:20-
smpte=10:07:33-
smpte=10:07:00-10:07:33:05.01
smpte-25=10:07:00-10:07:33:05.01
3.5 Normal Play Time
Normal play time (NPT) indicates the stream absolute position
relative to the beginning of the presentation, not to be confused
with the Network Time Protocol (NTP) [32]. The timestamp consists of
a decimal fraction. The part left of the decimal may be expressed in
either seconds or hours, minutes, and seconds. The part right of the
decimal point measures fractions of a second.
The beginning of a presentation corresponds to 0.0 seconds. Negative
values are not defined. The special constant now is defined as the
current instant of a live type event. It MAY only be used for live
type events, and SHALL NOT be used for on-demand content.
NPT is defined as in DSM-CC [33]: "Intuitively, NPT is the clock the
viewer associates with a program. It is often digitally displayed on
a VCR. NPT advances normally when in normal play mode (scale = 1),
advances at a faster rate when in fast scan forward (high positive
scale ratio), decrements when in scan reverse (high negative scale
ratio) and is fixed in pause mode. NPT is (logically) equivalent to
SMPTE time codes."
Examples:
npt=123.45-125
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npt=12:05:35.3-
npt=now-
The syntax conforms to ISO 8601 [34]. The npt-sec notation
is optimized for automatic generation, the ntp-hhmmss
notation for consumption by human readers. The "now"
constant allows clients to request to receive the live feed
rather than the stored or time-delayed version. This is
needed since neither absolute time nor zero time are
appropriate for this case.
3.6 Absolute Time
Absolute time is expressed as ISO 8601 [34] timestamps, using UTC
(GMT). Fractions of a second may be indicated.
Example for November 8, 1996 at 14h37 and 20 and a quarter seconds
UTC:
19961108T143720.25Z
3.7 Feature-tags
Feature-tags are unique identifiers used to designate features in
RTSP. These tags are used in Require (Section 14.37), Proxy-Require
(Section 14.31), Proxy-Supported (Section 14.32), Unsupported
(Section 14.46), and Supported (Section 14.43) header fields.
Feature tag needs to indicate which combination of clients, servers,
or proxies they applies too.
The creator of a new RTSP feature-tag should either prefix the
feature-tag with a reverse domain name (e.g.,
"com.example.mynewfeature" is an apt name for a feature whose
inventor can be reached at "example.com"), or register the new
feature-tag with the Internet Assigned Numbers Authority (IANA), see
IANA Section 21.
The usage of feature tags are further described in section 10 that
deals with capability handling.
3.8 Entity Tags
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Entity tags are opaque strings that are used to compare two entities
from the same resource, for example in caches or to optimize setup
after a redirect. Further explanation is present in [H3.11]. For
explanation on how to compare Entity tags see [H13.3]. Entity tags
can be carried in the ETag header (see section 14.21) or in SDP (see
section C.1.8).
Entity tags are used in RTSP to make some methods conditional. The
methods are made conditional through the inclusion of headers, see
14.25 and 14.27.
4 RTSP Message
RTSP is a text-based protocol and uses the ISO 10646 character set in
UTF-8 encoding (RFC 2279 [13]). Lines SHALL be terminated by CRLF.
Text-based protocols make it easier to add optional
parameters in a self-describing manner. Since the number of
parameters and the frequency of commands is low, processing
efficiency is not a concern. Text-based protocols, if done
carefully, also allow easy implementation of research
prototypes in scripting languages such as Tcl, Visual Basic
and Perl.
The 10646 character set avoids tricky character set switching, but is
invisible to the application as long as US-ASCII is being used. This
is also the encoding used for RTCP. ISO 8859-1 translates directly
into Unicode with a high-order octet of zero. ISO 8859-1 characters
with the most-significant bit set are represented as 1100001x
10xxxxxx. (See RFC 2279 [13])
Requests contain methods, the object the method is operating upon and
parameters to further describe the method. Methods are idempotent,
unless otherwise noted. Methods are also designed to require little
or no state maintenance at the media server.
4.1 Message Types
See [H4.1].
4.2 Message Headers
See [H4.2].
4.3 Message Body
See [H4.3]
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4.4 Message Length
When a message body is included with a message, the length of that
body is determined by one of the following (in order of precedence):
1. Any response message which MUST NOT include a message body
(such as the 1xx, 204, and 304 responses) is always
terminated by the first empty line after the header fields,
regardless of the entity-header fields present in the
message. (Note: An empty line consists of only CRLF.)
2. If a Content-Length header field (section 14.16) is
present, its value in bytes represents the length of the
message-body. If this header field is not present, a value
of zero is assumed.
Unlike an HTTP message, an RTSP message MUST contain a Content-Length
header field whenever it contains a message body. Note that RTSP does
not (at present) support the HTTP/1.1 "chunked" transfer coding(see
[H3.6.1]).
Given the moderate length of presentation descriptions
returned, the server should always be able to determine its
length, even if it is generated dynamically, making the
chunked transfer encoding unnecessary.
5 General Header Fields
See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade,
and Warning headers are not defined. RTSP further defines the CSeq,
and Timestamp. The general headers are listed in table 1:
Header Name Comment
_________________________________
Cache-Control See section 14.10
Connection See section 14.11
CSeq See section 14.19
Date See section 14.20
Supported See section 14.43
Timestamp See section 14.44
Via See section 14.49
Table 1: The General headers used in RTSP.
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6 Request
A request messages uses the format outlined below, regardless of the
direction of a request, client to server or server to client:
o Request line, containing the method to be applied to the
resource, the identifier of the resource, and the protocol
version in use;
o zero or more Header lines, that can be of the following types:
general (Section 5), request (Section 6.2), or entity (Section
8.1);
o One empty line (CR/LF) to indicate the end of the header
section;
o Optionally a message body (entity), consisting of one or more
lines. the length of the message body in number of bytes is
indicated by the Content-Length entity header.
6.1 Request Line
The request line provides the key information about the request:
What method, on what resources and using which RTSP version. The
methods that are defined by this specification are listed in Table 2.
Method Defined In Section
_________________________________
DESCRIBE Section 11.2
GET_PARAMETER Section 11.7
OPTIONS Section 11.1
PAUSE Section 11.5
PLAY Section 11.4
REDIRECT Section 11.9
SETUP Section 11.3
SET_PARAMETER Section 11.8
TEARDOWN Section 11.6
Table 2: The RTSP Methods
The syntax of the RTSP request line is the following:
<Method> SP <Request-URI> SP <RTSP-Version> CRLF
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Note: This syntax cannot be freely changed in future versions of
RTSP. This line needs to remain parsable by older RTSP
implementations since it indicates the RTSP version of the message.
In contrast to HTTP/1.1 [3], RTSP requests identify the resource
through an absolute RTSP URI (scheme, host, and port)(see section
3.2) rather than just the absolute path.
HTTP/1.1 requires servers to understand the absolute URI,
but clients are supposed to use the Host request header.
This is purely needed for backward-compatibility with
HTTP/1.0 servers, a consideration that does not apply to
RTSP.
An asterisk "*" can be used in the Request-URI to indicate that the
request does not apply to a particular resource, but to the server or
proxy itself, and is only allowed when the request method does not
necessarily apply to a resource.
For example:
OPTIONS * RTSP/1.1
An OPTIONS in this form will determine the capabilities of the server
or the proxy that first receives the request. If the capability of
the specific server needs to be determined, without regard to the
capability of an intervening proxy, the server should be addressed
explicitly with an absolute URI that contains the server's address.
For example:
OPTIONS rtsp://example.com RTSP/1.1
6.2 Request Header Fields
The RTSP headers in Table 3 can be included in a request, as request
headers, to modify the specifics of the request. These headers may
also be used in the response to a request, as response headers, to
modify the specifics of a response (Section 7.1.2).
Detailed headers definition are provided in Section 14.
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Header Defined in Section
_____________________________________
Accept Section 14.1
Accept-Encoding Section 14.3
Accept-Language Section 14.4
Authorization Section 14.7
Bandwidth Section 14.8
Blocksize Section 14.9
From Section 14.23
If-Match Section 14.25
If-Modified-Since Section 14.26
If-None-Match Section 14.27
Proxy-Require Section 14.31
Range Section 14.34
Referer Section 14.35
Require Section 14.37
Scale Section 14.39
Session Section 14.42
Speed Section 14.40
Supported Section 14.43
Transport Section 14.45
User-Agent Section 14.47
Table 3: The RTSP request headers
7 Response
[H6] applies except that HTTP-Version is replaced by RTSP-Version.
Also, RTSP defines additional status codes and does not define some
HTTP codes. The valid response codes and the methods they can be used
with are defined in Table 4.
After receiving and interpreting a request message, the recipient
responds with an RTSP response message.
7.1 Status-Line
The first line of a Response message is the Status-Line, consisting
of the protocol version followed by a numeric status code, and the
textual phrase associated with the status code, with each element
separated by SP characters. No CR or LF is allowed except in the
final CRLF sequence.
<RTSP-Version> SP <Status-Code> SP <Reason-Phrase> CRLF
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7.1.1 Status Code and Reason Phrase
The Status-Code element is a 3-digit integer result code of the
attempt to understand and satisfy the request. These codes are fully
defined in Section 13. The Reason-Phrase is intended to give a short
textual description of the Status-Code. The Status-Code is intended
for use by automata and the Reason-Phrase is intended for the human
user. The client is not required to examine or display the Reason-
Phrase.
The first digit of the Status-Code defines the class of response. The
last two digits do not have any categorization role. There are 5
values for the first digit:
o 1xx: Informational - Request received, continuing process
o 2xx: Success - The action was successfully received,
understood, and accepted
o 3rr: Redirection - Further action needs to be taken in order
to complete the request
o 4xx: Client Error - The request contains bad syntax or cannot
be fulfilled
o 5xx: Server Error - The server failed to fulfill an apparently
valid request
The individual values of the numeric status codes defined for
RTSP/1.1, and an example set of corresponding Reason-Phrases, are
presented in table 4. The reason phrases listed here are only
recommended; they may be replaced by local equivalents without
affecting the protocol. Note that RTSP adopts most HTTP/1.1 [3]
status codes and adds RTSP-specific status codes starting at x50 to
avoid conflicts with newly defined HTTP status codes.
RTSP status codes are extensible. RTSP applications are not required
to understand the meaning of all registered status codes, though such
understanding is obviously desirable. However, applications MUST
understand the class of any status code, as indicated by the first
digit, and treat any unrecognized response as being equivalent to the
x00 status code of that class, with the exception that an
unrecognized response MUST NOT be cached. For example, if an
unrecognized status code of 431 is received by the client, it can
safely assume that there was something wrong with its request and
treat the response as if it had received a 400 status code. In such
cases, user agents SHOULD present to the user the entity returned
with the response, since that entity is likely to include human-
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readable information which will explain the unusual status.
7.1.2 Response Header Fields
The response-header fields allow the request recipient to pass
additional information about the response which cannot be placed in
the Status-Line. These header fields give information about the
server and about further access to the resource identified by the
Request-URI. All headers currently being classified as response
headers are listed in table 5.
Response-header field names can be extended reliably only in
combination with a change in the protocol version. However, new or
experimental header fields MAY be given the semantics of response-
header fields if all parties in the communication recognize them to
be response-header fields. Unrecognized header fields are treated as
entity-header fields.
8 Entity
Request and Response messages MAY transfer an entity if not otherwise
restricted by the request method or response status code. An entity
consists of entity-header fields and an entity-body, although some
responses will only include the entity-headers.
The SET_PARAMETER, and GET_PARAMETER request and response, and
DESCRIBE response MAY have an entity. All 4xx and 5xx responses MAY
also have an entity.
In this section, both sender and recipient refer to either the client
or the server, depending on who sends and who receives the entity.
8.1 Entity Header Fields
Entity-header fields define optional meta-information about the
entity-body or, if no body is present, about the resource identified
by the request. The entity header fields are listed in table 8.1.
The extension-header mechanism allows additional entity-header fields
to be defined without changing the protocol, but these fields cannot
be assumed to be recognizable by the recipient. Unrecognized header
fields SHOULD be ignored by the recipient and forwarded by proxies.
8.2 Entity Body
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See [H7.2] with the addition that an RTSP message with an entity body
MUST include the Content-Type and Content-Length headers.
9 Connections
RTSP requests can be transmitted over two different connection
scenarios listed below:
o persistent - transport connections used for several
request/response transactions;
o transient - transport connections used for a single
request/response transaction.
RFC 2326 attempted to specify an optional mechanism for transmitting
RTSP messages in connectionless mode over a transport protocol such
as UDP. However, it was not specified in sufficient enough detail to
allow for interoperable implementations. In an attempt to reduce
complexity and scope, and due to lack of interest, RTSP 1.1 does not
attempt to define a mechanism for supporting RTSP over UDP or other
connectionless transport protocols. A side-effect is that RTSP
requests SHALL NOT be sent to multicast groups since no connection
can be established with a specific receiver in multicast
environments.
Certain RTSP headers, such as the CSeq header (Section 14.19), which
may appear to be relevant to only connectionless transport scenarios
are still retained and must be implemented according to the
specification. In the case of CSeq it is quite useful in proxy
situations for keeping track of the different request when
aggregating several client requests to a single TCP connection.
9.1 Reliability and Acknowledgements
Since RTSP is transmitted primarily over connection oriented,
reliable transport protocols, all RTSP requests MUST be acknowledged
by the receiver. RTSP requests that are not immediately acknowledged
MUST NOT be retransmitted at the application level. Instead, the
application must rely on the underlying transport to provide
reliability.
If both the underlying reliable transport such as TCP and
the RTSP application retransmit requests, each packet loss
or message loss may result in two retransmissions. The
receiver typically cannot take advantage of the
application-layer retransmission since the transport stack
will not deliver the application-layer retransmission
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Code Reason Method
__________________________________________________________
100 Continue all
__________________________________________________________
200 OK all
201 Reserved n/a
250 Reserved n/a
__________________________________________________________
300 Multiple Choices all
301 Moved Permanently all
302 Found all
303 See Other all
305 Use Proxy all
__________________________________________________________
400 Bad Request all
401 Unauthorized all
402 Payment Required all
403 Forbidden all
404 Not Found all
405 Method Not Allowed all
406 Not Acceptable all
407 Proxy Authentication Required all
408 Request Timeout all
410 Gone all
411 Length Required all
412 Precondition Failed DESCRIBE, SETUP
413 Request Entity Too Large all
414 Request-URI Too Long all
415 Unsupported Media Type all
451 Parameter Not Understood SET_PARAMETER
452 reserved n/a
453 Not Enough Bandwidth SETUP
454 Session Not Found all
455 Method Not Valid In This State all
456 Header Field Not Valid all
457 Invalid Range PLAY, PAUSE
458 Parameter Is Read-Only SET_PARAMETER
459 Aggregate Operation Not Allowed all
460 Only Aggregate Operation Allowed all
461 Unsupported Transport all
462 Destination Unreachable all
463 Destination Prohibited SETUP
470 Connection Authorization Required all
471 Connection Credentials not accepted all
__________________________________________________________
500 Internal Server Error all
501 Not Implemented all
502 Bad Gateway all
503 Service Unavailable all
504 Gateway Timeout all
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Table 4: Status codes and their usage with RTSP methods
Header Defined in Section
__________________________________________
Accept-Ranges Section 14.5
Connection-Credentials Section 14.12
ETag Section 14.21
Location Section 14.29
Proxy-Authenticate Section 14.30
Public Section 14.33
Range Section 14.34
Retry-After Section 14.36
RTP-Info Section 14.38
Scale Section 14.39
Session Section 14.42
Server Section 14.41
Speed Section 14.40
Transport Section 14.45
Unsupported Section 14.46
Vary Section 14.48
WWW-Authenticate Section 14.50
Table 5: The RTSP response headers
Header Defined in Section
____________________________________
Allow Section 14.6
Content-Base Section 14.13
Content-Encoding Section 14.14
Content-Language Section 14.15
Content-Length Section 14.16
Content-Location Section 14.17
Content-Type Section 14.18
Expires Section 14.22
Last-Modified Section 14.28
Table 6: The RTSP entity headers
before the first attempt has reached the receiver. If the
packet loss is caused by congestion, multiple
retransmissions at different layers will exacerbate the
congestion.
Lack of acknowledgement of an RTSP request should be handled within
the constraints of the connection timeout considerations described
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below (Section 9.4).
9.2 Using Connections
A TCP transport can be used for both persistent connections (for
several message exchanges) and transient connections (for a single
message exchange). Implementations of this specification MUST support
RTSP over TCP. The scheme of the RTSP URI (Section 3.2) indicates the
default port that the server will listen on.
A server MUST handle both persistent and transient connections.
Transient connections facilitate mechanisms for fault
tolerance. They also allow for application layer mobility.
A server and client pair that support transient connections
can survive the loss of a TCP connection, e.g. due to a NAT
timeout. When the client has discovered that the TCP
connection has been lost, it can set up a new one when
there is need to communicate again.
A persistent connection MAY be used for all transactions between the
server and client, including messages to multiple RTSP sessions.
However a persistent connection MAY also be closed after a few
message exchanges. For example, a client may use a persistent
connection for the initial SETUP and PLAY message exchanges in a
session and then close the connection. Later, when the client wishes
to send a new request, such as a PAUSE for the session, a new
connection would be opened. This connection may either be transient
or persistent.
A client SHOULD NOT have more than one connection to the server at
any given point. If a client or proxy handles multiple RTSP sessions
on the same server, it SHOULD use only one connection for managing
those sessions.
This saves connection resources on the server. It also
reduces complexity by and enabling the server to maintain
less state about its sessions and connections.
Unlike HTTP, RTSP allows a server to send requests to a client.
However, this can be supported only if a client establishes a
persistent connection with the server. In cases where a persistent
connection does not exist between a server and its client, due to the
lack of a signalling channel, the server may be forced to drop an
RTSP session without notifying the client. An example of such a case
is when the server desires to send a REDIRECT request for an RTSP
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session to the client but is not able to do so because it cannot
reach the client.
Without a persistent connection between the client and the
server, the media server has no reliable way of reaching
the client. Also, this is the only way that requests from a
server to its client are likely to traverse firewalls.
In light of the above, it is RECOMMENDED that clients use persistent
connections whenever possible. A client that supports persistent
connections MAY "pipeline" its requests (i.e., send multiple requests
without waiting for each response). A server MUST send its responses
to those requests in the order that the requests were received.
9.3 Closing Connections
The client MAY close a connection at any point when no outstanding
request/response transactions exist for any RTSP session being
managed through the connection. The server, however, SHOULD NOT close
a connection until all RTSP sessions being managed through the
connection have been timed out (Section 14.42). A server SHOULD NOT
close a connection immediately after responding to a session-level
TEARDOWN request for the last RTSP session being controlled through
the connection. Instead, it should wait for a reasonable amount of
time for the client to: receive the TEARDOWN response, take
appropriate action, and initiate the connection closing. The server
SHOULD wait at least 10 seconds after sending the TEARDOWN response
before closing the connection.
This is to ensure that the client has time to issue a SETUP
for a new session on the existing connection after having
torn the last one down. 10 seconds should give the client
ample opportunity get its message to the server.
A server SHOULD NOT close the connection directly as a result of
responding to a request with an error code.
Certain error responses such as "460 Only Aggregate
Operation Allowed" (Section 13.4.12) are used for
negotiating capabilities of a server with respect to
content or other factors. In such cases, it is inefficient
for the server to close a connection on an error response.
Also, such behavior would prevent implementation of
advanced/special types of requests or result in extra
overhead for the client when testing for new features. On
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the flip side, keeping connections open after sending an
error response poses a Denial of Service security risk
(Section 20).
If a server initiates a connection close while the client is
attempting to send a new request, the client will have to close its
current connection, establish a new connection and send its request
over the new connection.
An RTSP message should not be terminated through a connection close.
Such a message will be considered to be incomplete by the receiver
and discarded. An RTSP message is properly terminated as defined in
Section 4.
9.4 Timing Out Connections and RTSP Messages
Receivers of a request (responder) SHOULD respond to requests in a
timely manner even when a reliable transport such as TCP is used.
Similarly, the sender of a request (requestor) SHOULD wait for a
sufficient time for a response before concluding that the responder
will not be acting upon its request.
A responder SHOULD respond to all requests within 5 seconds. If the
responder recognizes that processing of a request will take longer
than 5 seconds, it SHOULD send a 100 response as soon as possible. It
SHOULD continue sending a 100 response every 5 seconds thereafter
until it is ready to send the final response to the requestor. After
sending a 100 response, the receiver MUST send a final response
indicating the success or failure of the request.
A requestor SHOULD wait at least 10 seconds for a response before
concluding that the responder will not be responding to its request.
After receiving a 100 response, the requestor SHOULD continue waiting
for further responses. If more than 10 seconds elapses without
receiving any response, the requestor MAY assume that the responder
is unresponsive and abort the connection.
A requestor SHOULD wait longer than 10 seconds for a response if it
is experiencing significant transport delays on its connection to the
responder. The requestor is capable of determining the RTT of the
request/response cycle using the Timestamp header (section 14.44) in
any RTSP request.
9.5 Use of IPv6
Explicit IPv6 support was not present in RTSP 1.0 (RFC 2326). RTSP
1.1 has been updated for explicit IPv6 support. Implementations of
RTSP 1.1 MUST understand literal IPv6 addresses in URIs and headers.
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10 Capability Handling
This section describes the capability handling mechanism available in
RTSP which allows RTSP to be extended. Extensions to this version of
the protocol are basically done in two ways. First, new headers can
be added. Secondly, new methods can be added. The capability handling
mechanism is designed to handle both cases.
When a method is added, the involved parties can use the OPTIONS
method to discover wether it is supported. This is done by issuing a
OPTIONS request to the other party. Depending on the URI it will
either apply in regards to a certain media resource, the whole server
in general, or simply the next hop. The OPTIONS response will contain
a Public header which declares all methods supported for the
indicated resource.
It is not necessary to use OPTIONS to discover support of a method,
the client could simply try the method. If the receiver of the
request does not support the method it will respond with an error
code indicating the the method is either not implemented (501) or
does not apply for the resource (405). The choice between the two
discovery methods depends on the requirements of the service.
Feature-Tags are defined to handle functionality additions that are
not new methods. Each feature-tag represents a certain block of
functionality. The amount of functionality that a feature-tag
represents can vary significantly. A feature-tag can for example
represent the functionality a single RTSP header provides. Another
feature-tag can represent much more functionality, such as the
"play.basic" feature tag which represents the minimal playback
implementation.
Feature-tags are used to determine wether the client, server or proxy
supports the functionality that is necessary to achieve the desired
service. To determine support of a feature-tag, several different
headers can be used, each explained below:
Supported: The supported header is used to determine the
complete set of functionality that both client and server
have. The intended usage is to determine before one needs
to use a functionality that it is supported. It can be used
in any method, however OPTIONS is the most suitable one as
it at the same time determines all methods that are
implemented. When sending a request the requestor declares
all its capabilities by including all supported feature-
tags. This results in that the receiver learns the
requestors feature support. The receiver then includes its
set of features in the response.
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Proxy-Supported: The Proxy-Supported header is used similar to
the Supported header, but instead of giving the supported
functionality of the client or server it provides both the
requestor and the responder a view of what functionality
the proxy chain between the two supports. Proxies are
required to add this header whenever the Supported header
is present, but proxies may independently of the requestor
add it.
Require: The Require header can be included in any request where
the end-point, i.e. the client or server, is required to
understand the feature to correctly perform the request.
This can, for example, be a SETUP request where the server
is required to understand a certain parameter to be able to
set up the media delivery correctly. Ignoring this
parameter would not have the desired effect and is not
acceptable. Therefore the end-point receiving a request
containing a Require MUST negatively acknowledge any
feature that it does not understand and not perform the
request. The response in cases where features are not
supported are 551 (Option Not Supported). Also the
features that are not supported are given in the
Unsupported header in the response.
Proxy-Require: This method has the same purpose and workings as
Require except that it only applies to proxies and not the
end-point. Features that needs to be supported by both
proxies and end-point needs to be included in both the
Require and Proxy-Require header.
Unsupported: This header is used in a 551 error response, to
indicate which feature(s) that was not supported. Such a
response is only the result of the usage of the Require
and/or Proxy-Require header where one or more feature where
not supported. This information allows the requestor to
make the best of situations as it knows which features are
not supported.
11 Method Definitions
The method indicates what is to be performed on the resource
identified by the Request-URI. The method name is case-sensitive.
New methods may be defined in the future. Method names SHALL NOT
start with a $ character (decimal 24) and MUST be a token as defined
by the ABNF [4] in the syntax chapter 19. The methods are summarized
in Table 7.
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method direction object Server req. Client req.
___________________________________________________________________
DESCRIBE C -> S P,S recommended recommended
GET_PARAMETER C -> S, S -> C P,S optional optional
OPTIONS C -> S, S -> C P,S R=Req, Sd=Opt Sd=Req, R=Opt
PAUSE C -> S P,S required required
PLAY C -> S P,S required required
REDIRECT S -> C P,S optional required
SETUP C -> S S required required
SET_PARAMETER C -> S, S -> C P,S required optional
TEARDOWN C -> S P,S required required
Table 7: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on. Legend: R=Respond,
Sd=Send, Opt: Optional, Req: Required, Rec: Recommended
Note on Table 7: GET_PARAMETER is recommended, but not
required. For example, a fully functional server can be
built to deliver media without any parameters.
SET_PARAMETER is required however due to its usage for
keep-alive. PAUSE is now required due to that it is the
only way of getting out of the state machines play state
without terminating the whole session.
If an RTSP agent does not support a particular method, it MUST return
501 (Not Implemented) and the requesting RTSP agent, in turn, SHOULD
NOT try this method again for the given agent / resource combination.
11.1 OPTIONS
The semantics of the RTSP OPTIONS method is equivalent to that of the
HTTP OPTIONS method described in [H9.2]. In RTSP however, OPTIONS is
bi-directional, in that a client can request it to a server and vice
versa. A client MUST implement the capability to send an OPTIONS
request and a server or a proxy MUST implement the capability to
respond to an OPTIONS request. The client, server or proxy MAY also
implement the converse of their required capability.
An OPTIONS request may be issued at any time. Such a request does not
modify the session state. However, it may prolong the session
lifespan (see below). The URI in an OPTIONS request determines the
scope of the request and the corresponding response. If the Request-
URI refers to a specific media resource on a given host, the scope is
limited to the set of methods supported for that media resource by
the indicated RTSP agent. A Request-URI with only the host address
limits the scope to the specified RTSP agent's general capabilities
without regard to any specific media. If the Request-URI is an
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asterisk ("*"), the scope is limited to the general capabilities of
the next hop (i.e. the RTSP agent in direct communication with the
request sender).
Regardless of scope of the request, the Public header MUST always be
included in the OPTIONS response listing the methods that are
supported by the responding RTSP agent. In addition, if the scope of
the request is limited to a media resource, the Allow header MUST be
included in the response to enumerate the set of methods that are
allowed for that resource unless the set of methods completely
matches the set in the Public header. If the given resource is not
available, the RTSP agent SHOULD return an appropriate response code
such as 3rr or 4xx. The Supported header MAY be included in the
request to query the set of features that are supported by the
responding RTSP agent.
The OPTIONS method can be used to keep an RTSP session alive.
However, it is not the preferred means of session keep-alive
signalling, see section 14.42. An OPTIONS request intended for
keeping alive an RTSP session MUST include the Session header with
the associated session ID. Such a request SHOULD also use the media
or the aggregated control URI as the Request-URI.
Example:
C->S: OPTIONS * RTSP/1.1
CSeq: 1
User-Agent: PhonyClient/1.2
Require:
Proxy-Require: gzipped-messages
Supported: play.basic
S->C: RTSP/1.1 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Supported: play.basic, implicit-play, gzipped-messages
Server: PhonyServer/1.1
Note that some of the feature-tags in Require and Proxy-Require are
necessarily fictional features (one would hope that we would not
purposefully overlook a truly useful feature just so that we could
have a strong example in this section).
11.2 DESCRIBE
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The DESCRIBE method is used to retrieve the description of a
presentation or media object from a server. The Request-URI of the
DESCRIBE request identifies the media resource of interest. The
client MAY include the Accept header in the request to list the
description formats that it understands. The server SHALL respond
with a description of the requested resource and return the
description in the entity of the response. The DESCRIBE reply-
response pair constitutes the media initialization phase of RTSP.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.1
CSeq: 312
User-Agent: PhonyClient 1.2
Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.1 200 OK
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.1
Content-Type: application/sdp
Content-Length: 367
v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.example.com/lectures/sdp.ps
e=seminar@example.com (Seminar Management)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
m=application 32416 UDP WB
a=orient:portrait
The DESCRIBE response SHOULD contain all media initialization
information for the resource(s) that it describes. Servers SHOULD NOT
use the DESCRIBE response as a means of media indirection by having
the description point at another server, instead usage of 3rr
responses are recommended.
By forcing a DESCRIBE response to contain all media
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initialization for the set of streams that it describes,
and discouraging the use of DESCRIBE for media indirection,
any looping problems can be avoided that might have
resulted from other approaches.
Media initialization is a requirement for any RTSP-based system, but
the RTSP specification does not dictate that this is required to be
done via the DESCRIBE method. There are three ways that an RTSP
client may receive initialization information:
o via an RTSP DESCRIBE request
o via some other protocol (HTTP, email attachment, etc.)
o via some form of a user interface
If a client obtains a valid description from an alternate source, the
client MAY use this description for initialization purposes without
issuing a DESCRIBE request for the same media.
It is RECOMMENDED that minimal servers support the DESCRIBE method,
and highly recommended that minimal clients support the ability to
act as "helper applications" that accept a media initialization file
from a user interface, and/or other means that are appropriate to the
operating environment of the clients.
11.3 SETUP
The SETUP request for an URI specifies the transport mechanism to be
used for the streamed media. The SETUP method may be used in three
different cases; Create an RTSP session, add a media to a session,
and change the transport parameters of already set up media stream.
When in PLAY state, using SETUP to create or add media to a session
when in PLAY state is unspecified. Otherwise SETUP can be used in all
three states; INIT, and READY, for both purposes and in PLAY to
change the transport parameters.
The Transport header, see section 14.45, specifies the transport
parameters acceptable to the client for data transmission; the
response will contain the transport parameters selected by the
server. This allows the client to enumerate in priority order the
transport mechanisms and parameters acceptable to it, while the
server can select the most appropriate. It is expected that the
session description format used will enable the client to select a
limited number possible configurations that are offered to the server
to choose from. All transport parameters SHOULD be included in the
Transport header, the use of other headers for this purpose is
discouraged due to middle boxes such as firewalls, or NATs.
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For the benefit of any intervening firewalls, a client SHOULD
indicate the transport parameters even if it has no influence over
these parameters, for example, where the server advertises a fixed
multicast address.
Since SETUP includes all transport initialization
information, firewalls and other intermediate network
devices (which need this information) are spared the more
arduous task of parsing the DESCRIBE response, which has
been reserved for media initialization.
In a SETUP response the server SHOULD include the Accept-Ranges
header (see section 14.5 to indicate which time formats that are
acceptable to use for this media resource.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.1
CSeq: 302
Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589",
RTP/AVP/TCP;unicast;interleaved=0-1
S->C: RTSP/1.1 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.1
Session: 47112344;timeout=60
Transport: RTP/AVP;unicast;dest_addr=":4588"/":4589";
src_addr="192.0.2.241:6256"/"192.0.2.241:6257";
ssrc=2A3F93ED
Accept-Ranges: NPT
In the above example the client wants to create an RTSP session
containing the media resource "rtsp://example.com/foo/bar/baz.rm".
The transport parameters acceptable to the client is either
RTP/AVP/UDP (UDP per default) to be received on client port 4588 and
4589 or RTP/AVP interleaved on the RTSP control channel. The server
selects the RTP/AVP/UDP transport and adds the ports it will send and
received RTP and RTCP from, and the RTP SSRC that will be used by the
server.
The server MUST generate a session identifier in response to a
successful SETUP request, unless a SETUP request to a server includes
a session identifier, in which case the server MUST bundle this setup
request into the existing session (aggregated session) or return
error 459 (Aggregate Operation Not Allowed) (see Section 13.4.11).
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An Aggregate control URI MUST be used to control an aggregated
session. This URI MUST be different from the stream control URIs of
the individual media streams included in the aggregate. The Aggregate
control URI is to be specified by the session description if the
server supports aggregated control and aggregated control is desired
for the session. However even if aggregated control is offered the
client MAY chose to not set up the session in aggregated control. If
an Aggregate control URI is not specified in the session description,
it is normally an indication that non-aggregated control should be
used. The SETUP of media streams in an aggregate which has not been
given an aggregated control URI is unspecified.
While the session ID sometimes has enough information for
aggregate control of a session, the Aggregate control URI
is still important for some methods such as SET_PARAMETER
where the control URI enables the resource in question to
be easily identified. The Aggregate control URI is also
useful for proxies, enabling them to route the request to
the appropriate server, and for logging, where it is useful
to note the actual resource that a request was operating
on.
A session will exist until it is either removed by a TEARDOWN request
or is timed-out by the server. The server MAY remove a session that
has not demonstrated liveness signs from the client(s) within a
certain timeout period. The default timeout value is 60 seconds; the
server MAY set this to a different value and indicate so in the
timeout field of the Session header in the SETUP response. For
further discussion see section 14.42. Signs of liveness for an RTSP
session are:
o Any RTSP request from a client(s) which includes a Session
header with that session's ID.
o If RTP is used as a transport for the underlying media
streams, an RTCP sender or receiver report from the client(s)
for any of the media streams in that RTSP session. RTCP Sender
Reports may for example be received in sessions where the
server is invited into a conference session and is as valid
for keep-alive.
If a SETUP request on a session fails for any reason, the session
state, as well as transport and other parameters for associated
streams SHALL remain unchanged from their values as if the SETUP
request had never been received by the server.
11.3.1 Changing Transport Parameters
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A client MAY issue a SETUP request for a stream that is already set
up or playing in the session to change transport parameters, which a
server MAY allow. If it does not allow changing of parameters, it
MUST respond with error 455 (Method Not Valid In This State). Reasons
to support changing transport parameters, is to allow for application
layer mobility and flexibility to utilize the best available
transport as it becomes available. If a client receives a 455 when
trying to change transport parameters while the server is in play
state, it MAY try to put the server in ready state using PAUSE.
Before trying issuing the SETUP request again. If also that fails the
changing of transport parameters will require that the client
performs a TEARDOWN of the affected media and then setting it up
again. In aggregated session avoiding tearing down all the media at
the same time will avoid the creation of a new session.
All transport parameters MAY be changed. However the primary usage
expected is to either change transport protocol completely, like
switching from Interleaved TCP mode to RTP or vise versa or change
delivery address.
In a SETUP response for a request to change the transport parameters
while in Play state, the server SHOULD include the Range to indicate
from what point the new transport parameters are used. Further, if
RTP is used for delivery, the server SHOULD also include the RTP-Info
header to indicate from what timestamp and RTP sequence number the
change has taken place. If both RTP-Info and Range is included in the
response the "rtp_time" parameter and range MUST be for the
corresponding time, i.e. be used in the same way as for PLAY to
ensure the correct synchronization information is available.
If the transport parameters change while in PLAY state results in a
change of synchronization related information, for example changing
RTP SSRC, the server MUST provide in the SETUP response the necessary
synchronization information. However the server is RECOMMENDED to
avoid changing the synchronization information if possible.
11.4 PLAY
The PLAY method tells the server to start sending data via the
mechanism specified in SETUP. A client MUST NOT issue a PLAY request
until any outstanding SETUP requests have been acknowledged as
successful. PLAY requests are valid when the session is in READY or
PLAY states. A PLAY request MUST include a Session header to indicate
which session the request applies to.
In an aggregated session the PLAY request MUST contain an aggregated
control URI. A server SHALL responde with error 460 (Only Aggregate
Operation Allowed) if the client PLAY Request-URI is for one of the
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media. The media in an aggregate SHALL be played in sync. If a client
want individual control of the media it needs to use separate RTSP
sessions for each media.
The PLAY request SHALL position the normal play time to the beginning
of the range specified by the Range header and delivers stream data
until the end of the range if given, else to the end of the media is
reached. To allow for precise composition multiple ranges MAY be
specified in one PLAY Request. The range values are valid if all
given ranges are part of any media within the aggregate. If a given
range value points outside of the media, the response SHALL be the
457 (Invalid Range) error code.
The below example will first play seconds 10 through 15, then,
immediately following, seconds 20 to 25, and finally seconds 30
through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.1
CSeq: 835
Session: 12345678
Range: npt=10-15, npt=20-25, npt=30-
See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It SHALL start
playing a stream from the beginning (npt=0-) unless the stream has
been paused or is currently playing. If a stream has been paused via
PAUSE, stream delivery resumes at the pause point. If a stream is
currently playing, the new PLAY begins at the current stream
position. The stream SHALL play until the end of the media.
The Range header MUST NOT contain a time parameter. The usage of time
in PLAY method has been deprecated. If a request with time parameter
is received the server SHOULD respond with a 457 (Invalid Range) to
indicate that the time parameter is not supported.
Server MUST include a "Range" header in any PLAY response. The
response MUST use the same format as the request's range header
contained. If no Range header was in the request, the NPT time format
SHOULD be used unless the client showed support for an other format
more appropriate. Also for a session with live media streams the
Range header MUST indicate a valid time. It is RECOMMENDED that
normal play time is used, either the "now" indicator, for example
"npt=now-", or the time since session start as an open interval, e.g.
"npt=96.23-". An absolute time value (clock) for the corresponding
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time MAY be given, i.e. "clock=20030213T143205Z-". The UTC clock
format SHOULD only be used if client has shown support for it.
A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats.
For an on-demand stream, the server MUST reply with the actual range
that will be played back, i.e. for which duration any media (having
content at this time) is delivered. This may differ from the
requested range if alignment of the requested range to valid frame
boundaries is required for the media source. Note that some media
streams in an aggregate may need to be delivered from even earlier
points. Also, some media format have a very long duration per
individual data unit, therefore it might be necessary for the client
to parse the data unit, and select where to start.
Example: Single audio stream (MIDI)
C->S: PLAY rtsp://example.com/audio RTSP/1.1
CSeq: 836
Session: 12345678
Range: npt=7.05-
S->C: RTSP/1.1 200 OK
CSeq: 836
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: npt=3.52-
RTP-Info:url="rtsp://example.com/audio"
ssrc=0D12F123:seq=14783;rtptime=2345962545
S->C: RTP Packet TS=2345962545 => NPT=3.52
Duration: 4.15 seconds
In this example the client receives the first media packet that
stretches all the way up and past the requested playtime. Thus, it is
the client's decision if to render to the user the time between 3.52
and 7.05, or to skip it. In most cases it is probably most suitable
to not render that time period.
For live media sources it might be impossible to specify from which
point in time all media streams carrying active content can actually
be delivered. Therefore a server MAY specify a start time (or now-)
in the range header, for which not all media will be available from.
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If no range is specified in the request, the start position SHALL
still be returned in the reply. If the medias that are part of an
aggregate has different lengths, the PLAY request SHALL be performed
as long as the given range is valid for any media, for example the
longest media. Media will be sent whenever it is available for the
given play-out point.
A PLAY response MAY include a header(s) carrying synchronization
information. As the information necessary is dependent on the media
transport format, further rules specifying the header and its usage
is needed. For RTP the RTP-Info header is specified, see section
14.38.
After playing the desired range, the presentation does NOT transition
to the READY state, media delivery simply stops. A PAUSE request MUST
be issued before the stream enters the READY state. A PLAY request
while the stream is still in the PLAYING state is legal, and can be
issued without an intervening PAUSE request. Such a request SHALL
replace the current PLAY action with the new one requested, i.e.
being handle the same as the request was received in ready state. In
the case the first time range in Range header has a open start time
(-endtime), the server SHALL continue to play from where it currently
was.
A client desiring to play the media from the beginning MUST send a
PLAY request with a Range header pointing at the beginning, e.g.
npt=0-. If a PLAY request is received without a Range header when
media delivery has stopped at the end, the server SHOULD respond with
a 457 "Invalid Range" error response. In that response the current
pause point in a Range header SHALL be included.
The following example plays the whole presentation starting at SMPTE
time code 0:10:20 until the end of the clip. Note: The RTP-Info
headers has been broken into several lines to fit the page.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.1
CSeq: 833
Session: 12345678
Range: smpte=0:10:20-
S->C: RTSP/1.1 200 OK
CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: smpte=0:10:22-0:15:45
RTP-Info:url="rtsp://example.com/twister.en"
ssrc=0D12F123:seq=14783;rtptime=2345962545
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For playing back a recording of a live presentation, it may be
desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.1
CSeq: 835
Session: 12345678
Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.1 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT
Server:PhonyServer 1.0
Range: clock=19961108T142300Z-19961108T143520Z
RTP-Info:url="rtsp://example.com/meeting.en"
ssrc=0D12F123:seq=53745;rtptime=484589019
All range specifiers in this specification allow for ranges with
unspecified begin times (e.g. "npt=-30"). When used in a PLAY
request, the server treats this as a request to start/resume playback
from the current pause point, ending at the end time specified in the
Range header. If the pause point is located later than the given end
value, a 457 (Invalid Range) response SHALL be given.
The possibility to replace a current PLAY request with a new one
replaces two RTSP 1.0 functions:
o The queued play functionality described in RFC 2326 [24] is
removed and multiple ranges can be used to achieve a similar
functionality.
o The use of PLAY for keep-alive signaling, i.e. PLAY request
without a range header in PLAY state, has also been
deprecated. Instead a client can use, SET_PARAMETER
(recommended) or OPTIONS (allowed) for keep alive.
11.5 PAUSE
The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. A PAUSE request MUST be done with the
aggregated control URI for aggregated sessions, resulting in all
media being halted, or the media URI for non-aggregated sessions.
Any attempt to do muting of a single media with an PAUSE request in
an aggregated session SHALL be responded with error 460 (Only
Aggregate Operation Allowed). After resuming playback,
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synchronization of the tracks MUST be maintained. Any server
resources are kept, though servers MAY close the session and free
resources after being paused for the duration specified with the
timeout parameter of the Session header in the SETUP message.
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 834
Session: 12345678
S->C: RTSP/1.1 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
Range: npt=45.76-
The PAUSE request MAY contain a Range header specifying when the
stream or presentation is to be halted. This point is referred to as
the "pause point". The time parameter in the Range MUST NOT be used.
The Range header MUST contain a single value, expressed as the
beginning value an open range. For example, the following clip will
be played from 10 seconds through 21 seconds of the clip's normal
play time, under the assumption that the PAUSE request reaches the
server within 11 seconds of the PLAY request. Note that some lines
has been broken in an non-correct way to fit the page:
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 834
Session: 12345678
Range: npt=10-30
S->C: RTSP/1.1 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: npt=10-30
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=4FAD8726:seq=57654;rtptime=2792482193
Session: 12345678
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 835
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Session: 12345678
Range: npt=21-
S->C: RTSP/1.1 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:09 GMT
Server: PhonyServer 1.0
Range: npt=21-
Session: 12345678
The pause request becomes effective the first time the server is
encountering the time point specified in any of the multiple ranges.
If the Range header specifies a time outside any range from the PLAY
request, the error 457 (Invalid Range) SHALL be returned. If a media
unit (such as an audio or video frame) starts presentation at exactly
the pause point, it is not played. If the Range header is missing,
stream delivery is interrupted immediately on receipt of the message
and the pause point is set to the current normal play time. However,
the pause point in the media stream MUST be maintained. A subsequent
PLAY request without Range header SHALL resume from the pause point
and play until media end.
If the server has already sent data beyond the time specified in the
PAUSE request's Range header, a PLAY without range SHALL resume at
the point in time specified by the PAUSE request's Range header, as
it is assumed that the client has discarded data after that point.
This ensures continuous pause/play cycling without gaps.
The pause point after any PAUSE request SHALL be returned to the
client by adding a Range header with what remains unplayed of the
PLAY request's ranges, i.e. including all the remaining ranges part
of multiple range specification. If one desires to resume playing a
ranged request, one simply includes the Range header from the PAUSE
response.
For example, if the server have a play request for ranges 10 to 15
and 20 to 29 pending and then receives a pause request for NPT 21, it
would start playing the second range and stop at NPT 21. If the pause
request is for NPT 12 and the server is playing at NPT 13 serving the
first play request, the server stops immediately. If the pause
request is for NPT 16, the server returns a 457 error message. To
prevent that the second range is played and the server stops after
completing the first range, a PAUSE request for NPT 20 needs to be
issued.
As another example, if a server has received requests to play ranges
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10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
request for NPT=14 would take effect while the server plays the first
range, with the second range effectively being ignored, assuming the
PAUSE request arrives before the server has started playing the
second, overlapping range. Regardless of when the PAUSE request
arrives, it sets the pause point to 14. The below example messages is
for the above case when the PAUSE request arrives before the first
occurrence of NPT=14.
C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 834
Session: 12345678
Range: npt=10-15, npt=13-20
S->C: RTSP/1.1 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
Server: PhonyServer 1.0
Range: npt=10-15, npt=13-20
RTP-Info:url="rtsp://example.com/fizzle/audiotrack"
ssrc=0D12F123:seq=5712;rtptime=934207921,
url="rtsp://example.com/fizzle/videotrack"
ssrc=789DAF12:seq=57654;rtptime=2792482193
Session: 12345678
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 835
Session: 12345678
Range: npt=14-
S->C: RTSP/1.1 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:09 GMT
Server: PhonyServer 1.0
Range: npt=14-15, npt=13-20
Session: 12345678
If a client issues a PAUSE request and the server acknowledges and
enters the READY state, the proper server response, if the player
issues another PAUSE, is still 200 OK. The 200 OK response MUST
include the Range header with the current pause point, even if the
PAUSE request is asking for some other pause point. See examples
below:
Examples:
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C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 834
Session: 12345678
S->C: RTSP/1.1 200 OK
CSeq: 834
Session: 12345678
Date: 23 Jan 1997 15:35:06 GMT
Range: npt=45.76-98.36
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 835
Session: 12345678
Range: 86-
S->C: RTSP/1.1 200 OK
CSeq: 835
Session: 12345678
Date: 23 Jan 1997 15:35:07 GMT
Range: npt=45.76-98.36
11.6 TEARDOWN
The TEARDOWN client to server request stops the stream delivery for
the given URI, freeing the resources associated with it. A TEARDOWN
request MAY be performed on either an aggregated or a media control
URI. However some restrictions apply depending on the current state.
The TEARDOWN request SHALL contain a Session header indicating what
session the request applies to.
A TEARDOWN using the aggregated control URI or the media URI in a
session under non-aggregated control MAY be done in any state (Ready,
and Play). A successful request SHALL result in that media delivery
is immediately halted and the session state is destroyed. This SHALL
be indicated through the lack of a Session header in the response.
A TEARDOWN using a media URI in an aggregated session MAY only be
done in Ready state. Such a request only removes the indicated media
stream and associated resources from the session. This may result in
that a session returns to non-aggregated control, due to that it only
contains a single media after the requests completion. A session that
will exist after the processing of the TEARDOWN request SHALL in the
response to that TEARDOWN request contain a Session header. Thus the
presence of the Session indicates to the receiver of the response if
the session is still existing or has been removed.
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Example:
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 892
Session: 12345678
S->C: RTSP/1.1 200 OK
CSeq: 892
Server: PhonyServer 1.0
11.7 GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter or
parameters for a presentation or stream specified in the URI. If the
Session header is present in a request, the value of a parameter MUST
be retrieved in the specified session context. The content of the
reply and response is left to the implementation.
The method MAY also be used without a body (entity). If the this
request is successful, i.e. a 200 OK response is received, then the
keep-alive timer has been updated. Any non-required header present in
such a request may or may not been processed. To allow a client to
determine if any such header has been processed, it is necessary to
use a feature tag and the Require header. Due to this reason it is
RECOMMENDED that any parameters to be retrieved are sent in the body,
rather than using any header.
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 431
Content-Type: text/parameters
Session: 12345678
Content-Length: 26
packets_received
jitter
C->S: RTSP/1.1 200 OK
CSeq: 431
Content-Length: 38
Content-Type: text/parameters
packets_received: 10
jitter: 0.3838
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The "text/parameters" section is only an example type for a
body carrying parameters.
11.8 SET_PARAMETER
This method requests to set the value of a parameter or a set of
parameters for a presentation or stream specified by the URI. The
method MAY also be used without a body (entity). It is the
RECOMMENDED method to use in request sent for the sole purpose of
updating the keep-alive timer. If this request is successful, i.e. a
200 OK response is received, then the keep-alive timer has been
updated. Any non-required header present in such a request may or may
not been processed. To allow a client to determine if any such header
has been processed, it is necessary to use a feature tag and the
Require header. Due to this reason it is RECOMMENDED that any
parameters are sent in the body, rather than using any header.
A request is RECOMMENDED to only contain a single parameter to allow
the client to determine why a particular request failed. If the
request contains several parameters, the server MUST only act on the
request if all of the parameters can be set successfully. A server
MUST allow a parameter to be set repeatedly to the same value, but it
MAY disallow changing parameter values. If the receiver of the
request does not understand or cannot locate a parameter, error 451
(Parameter Not Understood) SHALL be used. In the case a parameter is
not allowed to change, the error code is 458 (Parameter Is Read-
Only). The response body SHOULD contain only the parameters that have
errors. Otherwise no body SHALL be returned.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
Restricting setting transport parameters to SETUP is for
the benefit of firewalls.
The parameters are split in a fine-grained fashion so that
there can be more meaningful error indications. However, it
may make sense to allow the setting of several parameters
if an atomic setting is desirable. Imagine device control
where the client does not want the camera to pan unless it
can also tilt to the right angle at the same time.
Example:
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 421
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Content-length: 20
Content-type: text/parameters
barparam: barstuff
S->C: RTSP/1.1 451 Parameter Not Understood
CSeq: 421
Content-length: 10
Content-type: text/parameters
barparam
The "text/parameters" section is only an example type for
parameter. This method is intentionally loosely defined
with the intention that the reply content and response
content will be defined after further experimentation.
11.9 REDIRECT
The REDIRECT method is issued by a server to inform a client that it
required to connect to another server location to access the resource
indicated by the Request-URI. The presence of the Session header in a
REDIRECT request indicates the scope of the request, and determines
the specific semantics of the request.
A REDIRECT request with a Session header has end-to-end (i.e. server
to client) scope and applies only to the given session. Any
intervening proxies SHOULD NOT disconnect the control channel while
there are other remaining end-to-end sessions. The OPTIONAL Location
header, if included in such a request, SHALL contain a complete
absolute URI pointing to the resource to which the client SHOULD
reconnect. Specifically, the Location SHALL NOT contain just the
host and port. A client may receive a REDIRECT request with a Session
header, if and only if, an end-to-end session has been established.
A client may receive a REDIRECT request without a Session header at
any time when it has communication or a connection established with a
server. The scope of such a request is limited to the next-hop (i.e.
the RTSP agent in direct communication with the server) and applies,
as well, to the control connection between the next-hop RTSP agent
and the server. A REDIRECT request without a Session header
indicates that all sessions and pending requests being managed via
the control connection MUST be redirected. The OPTIONAL Location
header, if included in such a request, SHOULD contain an absolute URI
with only the host address and the OPTIONAL port number of the server
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to which the RTSP agent SHOULD reconnect. Any intervening proxies
SHOULD do all of the following in the order listed:
1. respond to the REDIRECT request
2. disconnect the control channel from the requesting server
3. connect to the server at the given host address
4. pass the REDIRECT request to each applicable client
(typically those clients with an active session or an
unanswered request)
Note: The proxy is responsible for accepting REDIRECT responses from
its clients; these responses MUST NOT be passed on to either the
original server or the redirected server.
The lack of a Location header in any REDIRECT request is indicative
of the server no longer being able to fulfill the current request and
having no alternatives for the client to continue with its normal
operation. It is akin to a server initiated TEARDOWN that applies
both to sessions as well as the general connection associated with
that client.
When the Range header is not included in a REDIRECT request, the
client SHOULD perform the redirection immediately and return a
response to the server. The server can consider the session as
terminated and can free any associated state after it receives the
successful (2xx) response. The server MAY close the signalling
connection upon receiving the response and the client SHOULD close
the signalling connection after sending the 2xx response. The
exception to this is when the client has several sessions on the
server being managed by the given signalling connection. In this
case, the client SHOULD close the connection when it has received and
responded to REDIRECT requests for all the sessions managed by the
signalling connection.
If the OPTIONAL Range header is included in a REDIRECT request, it
indicates when the redirection takes effect. The range value MUST be
an open ended single value, e.g. npt=59-, indicating the play out
time when redirection SHALL occur. Alternatively, a range with a
time= parameter indicates the wall clock time by when the redirection
MUST take place. When the time= parameter is present in the range,
any range value MUST be ignored even though it MUST be syntactically
correct. When the indicated redirect point is reached, a client MUST
issue a TEARDOWN request and SHOULD close the signalling connection
after receiving a 2xx response. The normal connection considerations
apply for the server.
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The differentiation of REDIRECT requests with and without
range headers is to allow for clear and explicit state
handling. As the state in the server needs to be kept until
the point of redirection, the handling becomes more clear
if the client is required to TEARDOWN the session at the
redirect point.
After a REDIRECT request has been processed, a client that wants to
continue to send or receive media for the resource identified by the
Request-URI will have to establish a new session with the designated
host. If the URI given in the Location header is a valid resource
URI, a client SHOULD issue a DESCRIBE request for the URI.
Note: The media resource indicated by the Location header
can be identical, slightly different or totally different.
This is the reason why a new DESCRIBE request SHOULD be
issued.
If the Location header contains only a host address, the client MAY
assume that the media on the new server is identical to the media on
the old server, i.e. all media configuration information from the old
session is still valid except for the host address.
This example request redirects traffic for this session to the new
server at the given absolute time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.1
CSeq: 732
Location: rtsp://s2.example.com:8001
Range: npt=0- ;time=19960213T143205Z
Session: uZ3ci0K+Ld-M
12 Embedded (Interleaved) Binary Data
In order to fulfill certain requirements on the network side, e.g.
in conjunction with network address translators that block RTP
traffic over UDP, it may be necessary to interleave RTSP messages and
media stream data. This interleaving should generally be avoided
unless necessary since it complicates client and server operation and
imposes additional overhead. Also head of line blocking may cause
problems. Interleaved binary data SHOULD only be used if RTSP is
carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar
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sign (24 decimal), followed by a one-byte channel identifier,
followed by the length of the encapsulated binary data as a binary,
two-byte integer in network byte order. The stream data follows
immediately afterwards, without a CRLF, but including the upper-layer
protocol headers. Each $ block SHALL contain exactly one upper-layer
protocol data unit, e.g., one RTP packet.
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| "$" = 24 | Channel ID | Length in bytes |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
: Length number of bytes of binary data :
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
The channel identifier is defined in the Transport header with the
interleaved parameter(Section 14.45).
When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. The usage of RTCP messages is
indicated by including a range containing a second channel in the
interleaved parameter of the Transport header, see section 14.45. If
RTCP is used, packets SHALL be sent on the first available channel
higher than the RTP channel. The channels are bi-directional and
therefore RTCP traffic are sent on the second channel in both
directions.
RTCP is needed for synchronization when two or more streams
are interleaved in such a fashion. Also, this provides a
convenient way to tunnel RTP/RTCP packets through the TCP
control connection when required by the network
configuration and transfer them onto UDP when possible.
C->S: SETUP rtsp://example.com/bar.file RTSP/1.1
CSeq: 2
Transport: RTP/AVP/TCP;unicast;interleaved=0-1
S->C: RTSP/1.1 200 OK
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CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;unicast;interleaved=5-6
Session: 12345678
C->S: PLAY rtsp://example.com/bar.file RTSP/1.1
CSeq: 3
Session: 12345678
S->C: RTSP/1.1 200 OK
CSeq: 3
Session: 12345678
Date: 05 Jun 1997 18:59:15 GMT
RTP-Info: url="rtsp://example.com/bar.file"
ssrc=0D12F123:seq=232433;rtptime=972948234
S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
S->C: $005{2 byte length}{"length" bytes data, w/RTP header}
S->C: $006{2 byte length}{"length" bytes RTCP packet}
13 Status Code Definitions
Where applicable, HTTP status [H10] codes are reused. Status codes
that have the same meaning are not repeated here. See Table 4 for a
listing of which status codes may be returned by which requests. All
error messages, 4xx and 5xx MAY return a body containing further
information about the error.
13.1 Success 1xx
13.1.1 100 Continue
See, [H10.1.1].
13.2 Success 2xx
13.3 Redirection 3xx
The notation "3rr" indicates response codes from 300 to 399 inclusive
which are meant for redirection. The response code 304 is excluded
from this set, as it is not used for redirection.
See [H10.3] for definition of status code 300 to 305. However
comments are given for some to how they apply to RTSP.
Within RTSP, redirection may be used for load balancing or
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redirecting stream requests to a server topologically closer to the
client. Mechanisms to determine topological proximity are beyond the
scope of this specification.
A 3rr code MAY be used to respond to any request. It is RECOMMENDED
that they are used if necessary before a session is established, i.e.
in response to DESCRIBE or SETUP. However in cases where a server is
not able to send a REDIRECT request to the client, the server MAY
need to resort to using 3rr responses to inform a client with a
established session about the need for redirecting the session. If an
3rr response is received for an request in relation to a established
session, the client SHOULD send a TEARDOWN request for the session,
and MAY reestablish the session using the resource indicated by the
Location.
If the the Location header is used in a response it SHALL contain an
absolute URI pointing out the media resource the client is redirected
to, the URI SHALL NOT only contain the host name.
13.3.1 300 Multiple Choices
See [H10.3.1] [TBW]
13.3.2 301 Moved Permanently
The request resource are moved permanently and resides now at the URI
given by the location header. The user client SHOULD redirect
automatically to the given URI. This response MUST NOT contain a
message-body. The Location header MUST be included in the response.
13.3.3 302 Found
The requested resource reside temporarily at the URI given by the
Location header. The Location header MUST be included in the
response. Is intended to be used for many types of temporary
redirects, e.g. load balancing. It is RECOMMENDED that one set the
reason phrase to something more meaningful than "Found" in these
cases. The user client SHOULD redirect automatically to the given
URI. This response MUST NOT contain a message-body.
13.3.4 303 See Other
This status code SHALL NOT be used in RTSP. However as it was allowed
to use in RTSP 1.1 (RFC 2326).
13.3.5 304 Not Modified
If the client has performed a conditional DESCRIBE or SETUP (see
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14.26) and the requested resource has not been modified, the server
SHOULD send a 304 response. This response MUST NOT contain a
message-body.
The response MUST include the following header fields:
o Date
o ETag and/or Content-Location, if the header would have been
sent in a 200 response to the same request.
o Expires, Cache-Control, and/or Vary, if the field-value might
differ from that sent in any previous response for the same
variant.
This response is independent for the DESCRIBE and SETUP requests.
That is, a 304 response to DESCRIBE does NOT imply that the resource
content is unchanged (only the session description) and a 304
response to SETUP does NOT imply that the resource description is
unchanged. The ETag and If-Match headers may be used to link the
DESCRIBE and SETUP in this manner.
13.3.6 305 Use Proxy
See [H10.3.6].
13.4 Client Error 4xx
13.4.1 400 Bad Request
The request could not be understood by the server due to malformed
syntax. The client SHOULD NOT repeat the request without
modifications [H10.4.1]. If the request does not have a CSeq header,
the server MUST NOT include a CSeq in the response.
13.4.2 405 Method Not Allowed
The method specified in the request is not allowed for the resource
identified by the Request-URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource.
This status code is also to be used if a request attempts to use a
method not indicated during SETUP, e.g., if a RECORD request is
issued even though the mode parameter in the Transport header only
specified PLAY.
13.4.3 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
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contained in the request. When returning this error message the
sender SHOULD return a entity body containing the offending
parameter(s).
13.4.4 452 reserved
This error code was removed from RFC 2326 [24] and is obsolete.
13.4.5 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.
13.4.6 454 Session Not Found
The RTSP session identifier in the Session header is missing,
invalid, or has timed out.
13.4.7 455 Method Not Valid in This State
The client or server cannot process this request in its current
state. The response SHOULD contain an Allow header to make error
recovery easier.
13.4.8 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example,
if PLAY contains the Range header field but the stream does not allow
seeking. This error message may also be used for specifying when the
time format in Range is impossible for the resource. In that case the
Accept-Ranges header SHOULD be returned to inform the client of which
format(s) that are allowed.
13.4.9 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
13.4.10 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not
modified. When returning this error message the sender SHOULD return
a entity body containing the offending parameter(s).
13.4.11 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URI in question since
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it is an aggregate (presentation) URI. The method may be applied on a
media URI.
13.4.12 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URI in question since
it is not an aggregate control (presentation) URI. The method may be
applied on the aggregate control URI.
13.4.13 461 Unsupported Transport
The Transport field did not contain a supported transport
specification.
13.4.14 462 Destination Unreachable
The data transmission channel could not be established because the
client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid dest_addr
parameter in the Transport field.
13.4.15 463 Destination Prohibited
The data transmission channel was not established because the server
prohibited access to the client address. This error is most likely
the result of a client attempt to redirect media traffic to another
destination with a dest_addr parameter in the Transport header.
13.4.16 470 Connection Authorization Required
The secured connection attempt need user or client authorization
before proceeding. The next hops certificate is included in this
response in the Accept-Credentials header.
13.4.17 471 Connection Credentials not accepted
When performing a secure connection over multiple connections, a
intermediary has refused to connect to the next hop and carry out the
request due to unacceptable credentials for the used policy.
13.5 Server Error 5xx
13.5.1 551 Option not supported
A feature-tag given in the Require or the Proxy-Require fields was
not supported. The Unsupported header SHOULD be returned stating the
feature for which there is no support.
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14 Header Field Definitions
method direction object acronym Body
_________________________________________________
DESCRIBE C -> S P,S DES r
GET_PARAMETER C -> S, S -> C P,S GPR R,r
OPTIONS C -> S P,S OPT
S -> C
PAUSE C -> S P,S PSE
PLAY C -> S P,S PLY
REDIRECT S -> C P,S RDR
SETUP C -> S S STP
SET_PARAMETER C -> S, S -> C P,S SPR R,r
TEARDOWN C -> S P,S TRD
Table 8: Overview of RTSP methods, their direction, and what objects
(P: presentation, S: stream) they operate on. Body notes if a method
is allowed to carry body and in which direction, R = Request,
r=response. Note: It is allowed for all error messages 4xx and 5xx to
have a body
The general syntax for header fields is covered in Section 4.2 This
section lists the full set of header fields along with notes on
meaning, and usage. The syntax definition for header fields are
present in section 19.2.3. Throughout this section, we use [HX.Y] to
refer to Section X.Y of the current HTTP/1.1 specification RFC 2616
[3]. Examples of each header field are given.
Information about header fields in relation to methods and proxy
processing is summarized in Tables 9, 10, 11, and 12.
The "where" column describes the request and response types in which
the header field can be used. Values in this column are:
R: header field may only appear in requests;
r: header field may only appear in responses;
2xx, 4xx, etc.: A numerical value or range indicates response
codes with which the header field can be used;
c: header field is copied from the request to the response.
An empty entry in the "where" column indicates that the header field
may be present in all requests and responses.
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The "proxy" column describes the operations a proxy may perform on a
header field. An empty proxy column indicates that the proxy SHALL
NOT do any changes to that header, all allowed operations are
explicitly stated:
a: A proxy can add or concatenate the header field if not
present.
m: A proxy can modify an existing header field value.
d: A proxy can delete a header field value.
r: A proxy needs to be able to read the header field, and thus
this header field cannot be encrypted.
The rest of the columns relate to the presence of a header field in a
method. The method names when abbreviated, are according to table 8:
c: Conditional; requirements on the header field depend on the
context of the message.
m: The header field is mandatory.
m*: The header field SHOULD be sent, but clients/servers need to
be prepared to receive messages without that header field.
o: The header field is optional.
*: The header field is SHALL be present if the message body is
not empty. See sections 14.16, 14.18 and 4.3 for details.
-: The header field is not applicable.
"Optional" means that a Client/Server MAY include the header field in
a request or response. The Client/Server behavior when receiving such
headers varies, for some it may ignore the header field, in other
case it is request to process the header. This is regulated by the
method and header descriptions. Example of such headers that require
processing are the Require and Proxy-Require header fields discussed
in 14.37 and 14.31. A "mandatory" header field MUST be present in a
request, and MUST be understood by the Client/Server receiving the
request. A mandatory response header field MUST be present in the
response, and the header field MUST be understood by the
Client/Server processing the response. "Not applicable" means that
the header field MUST NOT be present in a request. If one is placed
in a request by mistake, it MUST be ignored by the Client/Server
receiving the request. Similarly, a header field labeled "not
applicable" for a response means that the Client/Server MUST NOT
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place the header field in the response, and the Client/Server MUST
ignore the header field in the response.
A Client/Server SHOULD ignore extension header parameters that are
not understood.
The From and Location header fields contain an URI. If the URI
contains a comma, or semicolon, the URI MUST be enclosed in double
quotas ("). Any URI parameters are contained within these quotas. If
the URI is not enclosed in double quotas, any semicolon- delimited
parameters are header-parameters, not URI parameters.
14.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
The "level" parameter for presentation descriptions is
properly defined as part of the MIME type registration, not
here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl q=1.0, application/sdp
14.2 Accept-Credentials
The Accept-Credentials header is a request header used to indicate to
any trusted intermediary how to handle further secured connections to
proxies or servers. See section 18 for the usage of this header. It
SHALL only be included in client to server requests.
In a request the header SHALL contain the method (User, Proxy, or
Any) for approving credentials selected by the requestor. The method
SHALL NOT be changed by any proxy. If the method is "User" the header
contains zero or more of credentials that the client accept. Each
credential SHALL consist of one URI identifying the proxy or server,
and the SHA-1 [14] hash computed over that entity's DER encoded
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Header Where Proxy DES OPT SETUP PLAY PAUSE TRD
_________________________________________________________________
Accept R o - - - - -
Accept-Credentials R r o o o o o o
Accept-Encoding R r o - - - - -
Accept-Language R r o - - - - -
Accept-Ranges r r - - o - - -
Accept-Ranges 456 r - - - o o -
Allow r am - c - - - -
Allow 405 am m m m m m m
Authorization R o o o o o o
Bandwidth R o o o o - -
Blocksize R o - o o - -
Cache-Control r - - o - - -
Connection o o o o o o
Connection-Credentials 470,407 ar o o o o o o
Content-Base r o - - - - -
Content-Base 4xx o o o o o o
Content-Encoding R r - - - - - -
Content-Encoding r r o - - - - -
Content-Encoding 4xx r o o o o o o
Content-Language R r - - - - - -
Content-Language r r o - - - - -
Content-Language 4xx r o o o o o o
Content-Length r r * - - - - -
Content-Length 4xx r * * * * * *
Content-Location r o - - - - -
Content-Location 4xx o o o o o o
Content-Type r * - - - - -
Content-Type 4xx * * * * * *
CSeq Rc rm m m m m m m
Date am o o o o o o
ETag r r o - o - - -
Expires r r o - - - - -
From R r o o o o o o
Host - - - - - -
If-Match R r - - o - - -
If-Modified-Since R r o - o - - -
If-None-Match R r o - - - - -
Last-Modified r r o - - - - -
Location 3rr o o o o o o
Table 9: Overview of RTSP header fields (A-L) related to methods
DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.
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Header Where Proxy DES OPT SETUP PLAY PAUSE TRD
_____________________________________________________________
Proxy-Authenticate 407 amr m m m m m m
Proxy-Require R ar o o o o o o
Proxy-Require r r c c c c c c
Proxy-Supported R amr oc oc oc oc oc oc
Proxy-Supported r c c c c c c
Public r admr - m* - - - -
Public 501 admr m* m* m* m* m* m*
Range R - - - o o -
Range r - - c m* m* -
Referer R o o o o o o
Require R o o o o o o
Retry-After 3rr,503 o o o - - -
RTP-Info r - - o c - -
Scale - - - o - -
Session R r - o o m m m
Session r r - c m m m o
Server R r - o - - - -
Server r r o o o o o o
Speed - - - o - -
Supported R amr o o o o o o
Supported r amr c c c c c c
Timestamp R admr o o o o o o
Timestamp c admr m m m m m m
Transport amr - - m - - -
Unsupported r c c c c c c
User-Agent R m* m* m* m* m* m*
Vary r c c c c c c
Via R amr o o o o o o
Via c dr m m m m m m
WWW-Authenticate 401 m m m m m m
_____________________________________________________________
Header Where Proxy DES OPT SETUP PLAY PAUSE TRD
Table 10: Overview of RTSP header fields (P-W) related to methods
DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN.
certificate [15] in Base64 [35].
Example:
Accept-Credentials:User,
"rtsps://proxy2.example.com/";exaIl9VMbQMOFGClx5rXnPJKVNI=,
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Header Where Proxy GPR SPR RDR
__________________________________________________
Accept-Credentials R r o o o
Allow 405 amr m m m
Authorization R o o o
Bandwidth R - o -
Blocksize R - o -
Connection o o o
Connection-Credentials 470,407 ar o o o
Content-Base R o o -
Content-Base r o o -
Content-Base 4xx o o o
Content-Encoding R r o o -
Content-Encoding r r o o -
Content-Encoding 4xx r o o o
Content-Language R r o o -
Content-Language r r o o -
Content-Language 4xx r o o o
Content-Length R r * * -
Content-Length r r * * -
Content-Length 4xx r * * *
Content-Location R o o -
Content-Location r o o -
Content-Location 4xx o o o
Content-Type R * * -
Content-Type r * * -
Content-Type 4xx * * *
CSeq Rc mr m m m
Date am o o o
From R r o o o
Host - - -
Last-Modified R r - - -
Last-Modified r r o - -
Location 3rr o o o
Location R - - m
Proxy-Authenticate 407 amr m m m
Proxy-Require R ar o o o
Proxy-Require r r c c c
Proxy-Supported R amr oc oc oc
Proxy-Supported r c c c
Public 501 admr m* m* m*
__________________________________________________
Header Where Proxy GPR SPR RDR
Table 11: Overview of RTSP header fields (A-P) related to methods
GET_PARAMETER, SET_PARAMETER, and REDIRECT.
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Header Where Proxy GPR SPR RDR
____________________________________________
Range R - - o
Referer R o o o
Require R r o o o
Retry-After 3rr,503 o o -
Scale - - -
Session R r o o o
Session r r c c o
Server R r o o o
Server r r o o -
Supported R adrm o o o
Supported r adrm c c c
Timestamp R adrm o o o
Timestamp c adrm m m m
Unsupported r arm c c c
User-Agent R r m* m* -
User-Agent r r - - m*
Vary r c c -
Via R amr o o o
Via c dr m m m
WWW-Authenticate 401 m m m
____________________________________________
Header Where Proxy GPR SPR RDR
Table 12: Overview of RTSP header fields (R-W) related to methods
GET_PARAMETER, SET_PARAMETER, and REDIRECT.
14.3 Accept-Encoding
See [H14.3]
14.4 Accept-Language
See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media
content.
14.5 Accept-Ranges
The Accept-Ranges response-header field allows the server to indicate
its acceptance of range requests and possible formats for a resource:
Accept-Ranges: NPT, SMPTE
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This header has the same syntax as [H14.5] and the syntax is defined
in 19.2.3. However, new range-units are defined. Inclusion of any of
the time formats indicates acceptance by the server for PLAY and
PAUSE requests with this format. The headers value is valid for the
resource specified by the URI in the request, this response
corresponds to. A server SHOULD use this header in SETUP responses to
indicate to the client which range time formats the media supports.
The header SHOULD also be included in "456" responses which is a
result of use of unsupported range formats.
14.6 Allow
The Allow entity-header field lists the methods supported by the
resource identified by the Request-URI. The purpose of this field is
to strictly inform the recipient of valid methods associated with the
resource. An Allow header field MUST be present in a 405 (Method Not
Allowed) response. See [H14.7] for syntax definition. The Allow
header MUST also be present in all OPTIONS responses where the
content of the header will not include exactly the same methods as
listed in the Public header.
Example of use:
Allow: SETUP, PLAY, SET_PARAMETER
14.7 Authorization
See [H14.8]
14.8 Bandwidth
The Bandwidth request-header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured
in bits per second. The bandwidth available to the client may change
during an RTSP session, e.g., due to mobility.
Example:
Bandwidth: 4000
14.9 Blocksize
The Blocksize request-header field is sent from the client to the
media server asking the server for a particular media packet size.
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This packet size does not include lower-layer headers such as IP,
UDP, or RTP. The server is free to use a blocksize which is lower
than the one requested. The server MAY truncate this packet size to
the closest multiple of the minimum, media-specific block size, or
override it with the media-specific size if necessary. The block size
MUST be a positive decimal number, measured in octets. The server
only returns an error (4xx) if the value is syntactically invalid.
14.10 Cache-Control
The Cache-Control general-header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the
request/response chain.
Cache directives MUST be passed through by a proxy or gateway
application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a cache-
directive for a specific cache.
Cache-Control should only be specified in a SETUP request and its
response. Note: Cache-Control does not govern the caching of
responses as for HTTP, instead it applies to the media stream
identified by the SETUP request. The caching of RTSP requests are
generally not cacheable, for further information see section 16.
Below is the description of the cache directives that can be included
in the Cache-Control header.
no-cache: Indicates that the media stream MUST NOT be cached
anywhere. This allows an origin server to prevent caching
even by caches that have been configured to return stale
responses to client requests.
public: Indicates that the media stream is cacheable by any
cache.
private: Indicates that the media stream is intended for a
single user and MUST NOT be cached by a shared cache. A
private (non-shared) cache may cache the media stream.
no-transform: An intermediate cache (proxy) may find it useful
to convert the media type of a certain stream. A proxy
might, for example, convert between video formats to save
cache space or to reduce the amount of traffic on a slow
link. Serious operational problems may occur, however,
when these transformations have been applied to streams
intended for certain kinds of applications. For example,
applications for medical imaging, scientific data analysis
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and those using end-to-end authentication all depend on
receiving a stream that is bit-for-bit identical to the
original media stream. Therefore, if a response includes
the no-transform directive, an intermediate cache or proxy
MUST NOT change the encoding of the stream. Unlike HTTP,
RTSP does not provide for partial transformation at this
point, e.g., allowing translation into a different
language.
only-if-cached: In some cases, such as times of extremely poor
network connectivity, a client may want a cache to return
only those media streams that it currently has stored, and
not to receive these from the origin server. To do this,
the client may include the only-if-cached directive in a
request. If it receives this directive, a cache SHOULD
either respond using a cached media stream that is
consistent with the other constraints of the request, or
respond with a 504 (Gateway Timeout) status. However, if a
group of caches is being operated as a unified system with
good internal connectivity, such a request MAY be forwarded
within that group of caches.
max-stale: Indicates that the client is willing to accept a
media stream that has exceeded its expiration time. If
max-stale is assigned a value, then the client is willing
to accept a response that has exceeded its expiration time
by no more than the specified number of seconds. If no
value is assigned to max-stale, then the client is willing
to accept a stale response of any age.
min-fresh: Indicates that the client is willing to accept a
media stream whose freshness lifetime is no less than its
current age plus the specified time in seconds. That is,
the client wants a response that will still be fresh for at
least the specified number of seconds.
must-revalidate: When the must-revalidate directive is present
in a SETUP response received by a cache, that cache MUST
NOT use the entry after it becomes stale to respond to a
subsequent request without first revalidating it with the
origin server. That is, the cache is required to do an
end-to-end revalidation every time, if, based solely on the
origin server's Expires, the cached response is stale.)
proxy-revalidate: The proxy-revalidate directive has the same
meaning as the must-revalidate directive, except that it
does not apply to non-shared user agent caches. It can be
used on a response to an authenticated request to permit
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the user's cache to store and later return the response
without needing to revalidate it (since it has already been
authenticated once by that user), while still requiring
proxies that service many users to revalidate each time (in
order to make sure that each user has been authenticated).
Note that such authenticated responses also need the public
cache control directive in order to allow them to be cached
at all.
max-age: When an intermediate cache is forced, by means of a
max-age=0 directive, to revalidate its own cache entry, and
the client has supplied its own validator in the request,
the supplied validator might differ from the validator
currently stored with the cache entry. In this case, the
cache MAY use either validator in making its own request
without affecting semantic transparency.
However, the choice of validator might affect performance.
The best approach is for the intermediate cache to use its
own validator when making its request. If the server
replies with 304 (Not Modified), then the cache can return
its now validated copy to the client with a 200 (OK)
response. If the server replies with a new entity and cache
validator, however, the intermediate cache can compare the
returned validator with the one provided in the client's
request, using the strong comparison function. If the
client's validator is equal to the origin server's, then
the intermediate cache simply returns 304 (Not Modified).
Otherwise, it returns the new entity with a 200 (OK)
response.
14.11 Connection
See [H14.10]. The use of the connection option "close" in RTSP
messages SHOULD be limited to error messages when the server is
unable to recover and therefore see it necessary to close the
connection. The reason is that the client has the choice of
continuing using a connection indefinitely, as long as it sends valid
messages.
14.12 Connection-Credentials
The Connection-Credentials response header is used to carry the
credentials of any next hop that need to be approved by the
requestor. It SHALL only be used in server to client responses.
The Connection-Credentials header in an RTSP response SHALL, if
included, contain the credentials information of the next hop that an
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intermediary needs to securely connect to. The credential MUST
include the URI of the next proxy or server and the DER encoded
X.509v3 [15] certificate in base64 [35].
Example:
Connection-Credentials:"rtsps://proxy2.example.com/";MIIDNTCC...
14.13 Content-Base
The Content-Base entity-header field may be used to specify the base
URI for resolving relative URIs within the entity.
Content-Base: rtsp://media.example.com/movie/twister
If no Content-Base field is present, the base URI of an entity is
defined either by its Content-Location (if that Content-Location URI
is an absolute URI) or the URI used to initiate the request, in that
order of precedence. Note, however, that the base URI of the contents
within the entity-body may be redefined within that entity-body.
14.14 Content-Encoding
See [H14.11]
14.15 Content-Language
See [H14.12]
14.16 Content-Length
The Content-Length general-header field contains the length of the
body (entity) of the message (i.e. after the double CRLF following
the last header). Unlike HTTP, it MUST be included in all messages
that carry body beyond the header portion of the message. If it is
missing, a default value of zero is assumed. It is interpreted
according to [H14.13].
14.17 Content-Location
See [H14.14]
14.18 Content-Type
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See [H14.17]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and
parameter-value types.
14.19 CSeq
The CSeq general-header field specifies the sequence number for an
RTSP request-response pair. This field MUST be present in all
requests and responses. For every RTSP request containing the given
sequence number, the corresponding response will have the same
number. Any retransmitted request MUST contain the same sequence
number as the original (i.e. the sequence number is not incremented
for retransmissions of the same request). For each new RTSP request
the CSeq value SHALL be incremented by one. The initial sequence
number MAY be any number, however it is RECOMMENDED to start at 0.
Each sequence number series is unique between each requester and
responder, i.e. the client has one series for its request to a
server and the server has another when sending request to the client.
Each requester and responder is identified with its network address.
Example:
CSeq: 239
14.20 Date
See [H14.18]. An RTSP message containing a body MUST include a Date
header if the sending host has a clock. Servers SHOULD include a Date
header in all other RTSP messages.
14.21 ETag
The ETag response header MAY be included in DESCRIBE or SETUP
responses. The entity tag returned in a DESCRIBE response is for the
included entity, while for SETUP it refers to the media resource just
set up. This differentiation allows for cache validation of both
session description and the media resource associated with the
description. If the ETag is provided both inside the entity, e.g.
within the "a=etag" attribute in SDP, and in the response message,
then both tags SHALL be identical. It is RECOMMENDED that the ETag is
primarily given in the RTSP response message, to ensure that caches
can use the ETag without requiring content inspection.
SETUP and DESCRIBE requests can be made conditional upon the ETag
using the headers If-Match (Section 14.25) and If-None-Match (Section
14.27).
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14.22 Expires
The Expires entity-header field gives a date and time after which the
description or media-stream should be considered stale. The
interpretation depends on the method:
DESCRIBE response: The Expires header indicates a date and time
after which the presentation description (body) SHOULD be
considered stale.
SETUP response: The Expires header indicate a date and time
after which the media stream SHOULD be considered stale.
A stale cache entry may not normally be returned by a cache (either a
proxy cache or an user agent cache) unless it is first validated with
the origin server (or with an intermediate cache that has a fresh
copy of the entity). See section 16 for further discussion of the
expiration model.
The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.
The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format:
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.1 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occurred in the past
(i.e., already expired).
To mark a response as "already expired," an origin server should use
an Expires date that is equal to the Date header value. To mark a
response as "never expires," an origin server SHOULD use an Expires
date approximately one year from the time the response is sent.
RTSP/1.1 servers SHOULD NOT send Expires dates more than one year in
the future.
The presence of an Expires header field with a date value of some
time in the future on a media stream that otherwise would by default
be non-cacheable indicates that the media stream is cacheable, unless
indicated otherwise by a Cache-Control header field (Section 14.10).
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14.23 From
See [H14.22].
14.24 Host
The Host HTTP request header field [H14.23] is not needed for RTSP,
and SHALL NOT be sent. It SHALL be silently ignored if received.
14.25 If-Match
See [H14.24].
The If-Match request-header field is especially useful for ensuring
the integrity of the presentation description, in both the case where
it is fetched via means external to RTSP (such as HTTP), or in the
case where the server implementation is guaranteeing the integrity of
the description between the time of the DESCRIBE message and the
SETUP message. By including the ETag given in or with the session
description in a SETUP request, the client ensures that resources set
up are matching the description. A SETUP request for which the ETag
validation check fails, SHALL responde using 412 (Precondition
Failed).
This validation check is also very useful if a session has been
redirected from one server to another.
14.26 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (Not Modified)
response SHALL be returned without any message-body.
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
14.27 If-None-Match
See [H14.26].
This request header can be used with entity tags to make DESCRIBE
requests conditional. A new session description is retrieved only if
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another entity than the already available would be included. If the
entity available for delivery is matching the one the client already
has, then a 304 (Not Modified) response is given.
14.28 Last-Modified
The Last-Modified entity-header field indicates the date and time at
which the origin server believes the presentation description or
media stream was last modified. See [H14.29]. For the methods
DESCRIBE, the header field indicates the last modification date and
time of the description, for SETUP that of the media stream.
14.29 Location
See [H14.30].
14.30 Proxy-Authenticate
See [H14.33].
14.31 Proxy-Require
The Proxy-Require request-header field is used to indicate proxy-
sensitive features that MUST be supported by the proxy. Any Proxy-
Require header features that are not supported by the proxy MUST be
negatively acknowledged by the proxy to the client using the
Unsupported header. The proxy SHALL use the 551 (Option Not
Supported) status code in the response. Any feature tag included in
the Proxy-Require does not apply to the end-point (server or client).
To ensure that a feature is supported by both proxies and servers the
tag needs to be included in also a Require header.
See Section 14.37 for more details on the mechanics of this message
and a usage example.
Example of use:
Proxy-Require: play.basic
14.32 Proxy-Supported
The Proxy-Supported header field enumerates all the extensions
supported by the proxy using feature tags. The header carries the
intersection of extensions supported by the forwarding proxies. The
Proxy-Supported header MAY be included in any request by a proxy. It
SHALL be added by any proxy if the Supported header is present in a
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request. When present in a request, the receiver MUST in the response
copy the received Proxy-Supported header.
The Proxy-Supported header field contains a list of feature-tags
applicable to proxies, as described in Section 3.7. The list are the
intersection of all feature-tags understood by the proxies. To
achieve an intersection, the proxy adding the Proxy-Supported header
includes all proxy feature-tags it understands. Any proxy receiving a
request with the header, checks the list and removes any feature tag
it do not support. A Proxy-Supported header present in the response
SHALL NOT be touched by the proxies.
Example:
C->P1: OPTIONS rtsp://example.com/ RTSP/1.1
Supported: foo, bar, blech
P1->P2: OPTIONS rtsp://example.com/ RTSP/1.1
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-bar, proxy-blech
Via: 1.1 prox1.example.com
P2->S: OPTIONS rtsp://example.com/ RTSP/1.1
Supported: foo, bar, blech
Proxy-Supported: proxy-foo, proxy-blech
Via: 1.1 prox1.example.com, 1.1 prox2.example.com
S->C: RTSP/1.1 200 OK
Supported: foo, bar, baz
Proxy-Supported: proxy-foo, proxy-blech
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
Via: 1.1 prox1.example.com, 1.1 prox2.example.com
14.33 Public
The Public response header field lists the set of methods supported
by the response sender. This header applies to the general
capabilities of the sender and its only purpose is to indicate the
sender's capabilities to the recipient. The methods listed may or may
not be applicable to the Request-URI; the Allow header field (section
14.7) MAY be used to indicate methods allowed for a particular URI.
Example of use:
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
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In the event that there are proxies between the sender and the
recipient of a response, each intervening proxy MUST modify the
Public header field to remove any methods that are not supported via
that proxy. The resulting Public header field will contain an
intersection of the sender's methods and the methods allowed through
by the intervening proxies.
In general proxies should allow all methods to
transparently pass through from the sending RTSP agent to
the receiving RTSP agent, but there may be cases where this
is not desirable for a given proxy. Modification of the
Public response header field by the intervening proxies
ensures that the request sender gets an accurate response
indicating the methods that can be used on the target agent
via the proxy chain.
14.34 Range
The Range header specifies a time range in PLAY (Section 11.4), PAUSE
(Section 11.5), SETUP (Section 11.3), and REDIRECT (Section 11.9)
requests and/or responses.
The range can be specified in a number of units. This specification
defines smpte (Section 3.4), npt (Section 3.5), and clock (Section
3.6) range units. While byte ranges [H14.35.1] and other extended
units MAY be used, their behavior is unspecified since they are not
normally meaningful in RTSP. Servers supporting the Range header MUST
understand the NPT range format and SHOULD understand the SMPTE range
format. If the Range header is sent in a time format that is not
understood, the recipient SHOULD return 456 (Header Field Not Valid
for Resource) and include an Accept-Ranges header indicating the
supported time formats for the given resource.
The Range header MAY contain a time parameter in UTC, specifying the
time at which the operation is to be made effective. This
functionality SHALL be used only with the REDIRECT method.
Ranges are half-open intervals, including the first point, but
excluding the second point. In other words, a range of A-B starts
exactly at time A, but stops just before B. Only the start time of a
media unit such as a video or audio frame is relevant. For example,
assume that video frames are generated every 40 ms. A range of
10.0-10.1 would include a video frame starting at 10.0 or later time
and would include a video frame starting at 10.08, even though it
lasted beyond the interval. A range of 10.0-10.08, on the other hand,
would exclude the frame at 10.08.
Example:
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Range: clock=19960213T143205Z-;time=19970123T143720Z
The notation is similar to that used for the HTTP/1.1 [3]
byte-range header. It allows clients to select an excerpt
from the media object, and to play from a given point to
the end as well as from the current location to a given
point.
By default, range intervals increase, where the second point is
larger than the first point.
Example:
Range: npt=10-15
However, range intervals can also decrease if the Scale header (see
section 14.39) indicates a negative scale value. For example, this
would be the case when a playback in reverse is desired.
Example:
Scale: -1
Range: npt=15-10
Decreasing ranges are still half open intervals as described above.
Thus, for range A-B, A is closed and B is open. In the above example,
15 is closed and 10 is open. An exception to this rule is the case
when B=0 in a decreasing range. In this case, the range is closed on
both ends, as otherwise there would be no way to reach 0 on a reverse
playback.
Example:
Scale: -1
Range: npt=15-0
In this range both 15 and 0 are closed.
A decreasing range interval without a corresponding negative Scale
header is not valid.
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14.35 Referer
See [H14.36]. The URI refers to that of the presentation description,
typically retrieved via HTTP.
14.36 Retry-After
See [H14.37].
14.37 Require
The Require request-header field is used by clients or servers to
ensure that the other end-point supports features that are required
in respect to this request. It can also be used to query if the other
end-point supports certain features, however the use of the Supported
(Section 14.43) is much more effective in this purpose. The server
MUST respond to this header by using the Unsupported header to
negatively acknowledge those feature-tags which are NOT supported.
The response SHALL use the error code 551 (Option Not Supported).
This header does not apply to proxies, for the same functionality in
respect to proxies see, header Proxy-Require (Section 14.31).
This is to make sure that the client-server interaction
will proceed without delay when all features are understood
by both sides, and only slow down if features are not
understood (as in the example below). For a well-matched
client-server pair, the interaction proceeds quickly,
saving a round-trip often required by negotiation
mechanisms. In addition, it also removes state ambiguity
when the client requires features that the server does not
understand.
Example:
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.1
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/1.1 551 Option not supported
CSeq: 302
Unsupported: funky-feature
In this example, "funky-feature" is the feature-tag which indicates
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to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.
Proxies and other intermediary devices SHALL ignore this header. If a
particular extension requires that intermediate devices support it,
the extension should be tagged in the Proxy-Require field instead
(see Section 14.31).
14.38 RTP-Info
The RTP-Info response-header field is used to set RTP-specific
parameters in the PLAY response. For streams using RTP as transport
protocol the RTP-Info header SHOULD be part of a 200 response to
PLAY.
The exclusion of the RTP-Info in a PLAY response for RTP
transported media will result in that a client needs to
synchronize the media streams using RTCP. This may have
negative impact as the RTCP can be lost, and does not need
to be particulary timely in their arrival. Also
functionality as informing the client from which packet a
seek has occurred is affected.
The RTP-Info MAY also be included in SETUP responses to provide
synchronization information when changing transport parameters, see
11.3.
The header can carry the following parameters:
url: Indicates the stream URI which for which the following RTP
parameters correspond, this URI MUST be the same used in
the SETUP request for this media stream. Any relative URI
SHALL use the Request-URI as base URI. This parameter SHALL
be present.
ssrc: The Synchronization source (SSRC) that the RTP timestamp
and sequence number provide applies to. This parameter
SHALL be present.
seq: Indicates the sequence number of the first packet of the
stream that is direct result of the request. This allows
clients to gracefully deal with packets when seeking. The
client uses this value to differentiate packets that
originated before the seek from packets that originated
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after the seek. Note that a client may not receive the
packet with the expressed sequence number, and instead
packets with a higher sequence number, due to packet loss
or reordering. This parameter is RECOMMENDED to be present.
rtptime: SHALL indicate the RTP timestamp value corresponding to
the start time value in the Range response header, or if
not explicitly given the implied start point. The client
uses this value to calculate the mapping of RTP time to NPT
or other media timescale. This parameter SHOULD be present
to ensure inter-media synchronization is achieved. There
exist no requirement that any received RTP packet will have
the same RTP timestamp value as the one in the parameter
used to establish synchronization.
A mapping from RTP timestamps to NTP timestamps (wall
clock) is available via RTCP. However, this information is
not sufficient to generate a mapping from RTP timestamps to
media clock time (NPT, etc.). Furthermore, in order to
ensure that this information is available at the necessary
time (immediately at startup or after a seek), and that it
is delivered reliably, this mapping is placed in the RTSP
control channel.
In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP, using
initial RTCP sender reports to do the mapping, and later reports to
check drift against the mapping.
Example:
Range:npt=3.25-15
RTP-Info:url="rtsp://example.com/foo/audio" ssrc=0A13C760:seq=45102;
rtptime=12345678,url="rtsp://example.com/foo/video"
ssrc=9A9DE123:seq=30211;rtptime=29567112
Lets assume that audio uses a 16kHz RTP timestamp clock and Video
a 90kHz RTP timestamp clock. Then the media synchronization is
depicted in the following way.
NPT 3.0---3.1---3.2-X-3.3---3.4---3.5---3.6
Audio PA A
Video V PV
X: NPT time value = 3.25, from Range header.
A: RTP timestamp value for Audio from RTP-Info header (12345678).
V: RTP timestamp value for Video from RTP-Info header (29567112).
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PA: RTP audio packet carrying an RTP timestamp of 12344878. Which
corresponds to NPT = (12344878 - A) / 16000 + 3.25 = 3.2
PV: RTP video packet carrying an RTP timestamp of 29573412. Which
corresponds to NPT = (29573412 - V) / 90000 + 3.25 = 3.32
14.39 Scale
A scale value of 1 indicates normal play at the normal forward
viewing rate. If not 1, the value corresponds to the rate with
respect to normal viewing rate. For example, a ratio of 2 indicates
twice the normal viewing rate ("fast forward") and a ratio of 0.5
indicates half the normal viewing rate. In other words, a ratio of 2
has normal play time increase at twice the wallclock rate. For every
second of elapsed (wallclock) time, 2 seconds of content will be
delivered. A negative value indicates reverse direction. For certain
media transports this may require certain considerations to work
consistent, see section B.1 for description on how RTP handles this.
Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected key frames. For audio, it may time-scale
the audio while preserving pitch or, less desirably, deliver
fragments of audio.
The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response
MUST contain the actual scale value chosen by the server.
If the server does not implement the possibility to scale, it will
not return a Scale header. A server supporting Scale operations for
PLAY SHALL indicate this with the use of the "play.scale" feature-
tags.
When indicating a negative scale for a reverse playback, the Range
header MUST indicate a decreasing range as described in section
14.34.
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
Range: npt=15-10
14.40 Speed
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The Speed request-header field requests the server to deliver data to
the client at a particular speed, contingent on the server's ability
and desire to serve the media stream at the given speed.
Implementation by the server is OPTIONAL. The default is the bit rate
of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of
2.0 indicates that data is to be delivered twice as fast as normal. A
speed of zero is invalid. All speeds may not be possible to support.
Therefore the actual used speed MUST be included in the response. The
lack of a response header is indication of lack of support from the
server of this functionality. Support of the speed functionality are
indicated by the "play.speed" featuretag.
Example:
Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It is
meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. Implementors
should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss
rates. If the data transport is performed over public best-effort
networks the sender MUST perform congestion control of the stream(s).
This can result in that the communicated speed is impossible to
maintain.
14.41 Server
See [H14.38], however the header syntax is corrected in section
19.2.3.
14.42 Session
The Session request-header and response-header field identifies an
RTSP session. An RTSP session is created by the server as a result of
a successful SETUP request and in the response the session identifier
is given to the client. The RTSP session exist until destroyed by a
TEARDOWN or timed out by the server.
The session identifier is chosen by the server (see Section 3.3) and
MUST be returned in the SETUP response. Once a client receives a
session identifier, it SHALL be included in any request related to
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that session. This means that the Session header MUST be included in
a request using the following methods: PLAY, PAUSE, and TEARDOWN, and
MAY be included in SETUP, OPTIONS, SET_PARAMETER, GET_PARAMETER, and
REDIRECT, and SHALL NOT be included in DESCRIBE. In an RTSP response
the session header SHALL be included in methods, SETUP, PLAY, and
PAUSE, and MAY be included in methods, TEARDOWN, and REDIRECT, and if
included in the request of the following methods it SHALL also be
included in the response, OPTIONS, GET_PARAMETER, and SET_PARAMETER,
and SHALL NOT be included in DESCRIBE.
The timeout parameter MAY be included in a SETUP response, and SHALL
NOT be included in requests. The server uses it to indicate to the
client how long the server is prepared to wait between RTSP commands
or other signs of life before closing the session due to lack of
activity (see below and Section A). The timeout is measured in
seconds, with a default of 60 seconds (1 minute). The length of the
session timeout SHALL NOT be changed in a established session.
The mechanisms for showing liveness of the client is, any RTSP
request with a Session header, if RTP & RTCP is used an RTCP message,
or through any other used media protocol capable of indicating
liveness of the RTSP client. It is RECOMMENDED that a client does not
wait to the last second of the timeout before trying to send a
liveness message. The RTSP message may be lost or when using reliable
protocols, such as TCP, the message may take some time to arrive
safely at the receiver. To show liveness between RTSP request issued
to accomplish other things, the following mechanisms can be used, in
descending order of preference:
RTCP: If RTP is used for media transport RTCP SHOULD be used. If
RTCP is used to report transport statistics, it SHALL also
work as keep alive. The server can determine the client by
used network address and port together with the fact that
the client is reporting on the servers SSRC(s). A downside
of using RTCP is that it only gives statistical guarantees
to reach the server. However that probability is so low
that it can be ignored in most cases. For example, a
session with 60 seconds timeout and enough bitrate assigned
to RTCP messages to send a message from client to server on
average every 5 seconds. That client have for a network
with 5 % packet loss, the probability to fail showing
liveness sign in that session within the timeout interval
of 2.4*E-16. In sessions with shorter timeout times, or
much higher packet loss, or small RTCP bandwidths SHOULD
also use any of the mechanisms below.
SET_PARAMETER: When using SET_PARAMETER for keep alive, no body
SHOULD be included. This method is the RECOMMENDED RTSP
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method to use in request only intended to perform keep-
alive.
OPTIONS: This method does also work. However it causes the
server to perform unnecessary processing and result in
bigger responses than necessary for the task. The reason
for this is that the server needs to determine what
capabilities that are associated with the media resource to
correctly populate the Public and Allow headers.
Note that a session identifier identifies an RTSP session across
transport sessions or connections. RTSP requests for a given session
can use different URIs (Presentation and media URIs). Note, that
there are restrictions depending on the session which URIs that are
acceptable for a given method. However, multiple "user" sessions for
the same URI from the same client will require use of different
session identifiers.
The session identifier is needed to distinguish several
delivery requests for the same URI coming from the same
client.
The response 454 (Session Not Found) SHALL be returned if the session
identifier is invalid.
14.43 Supported
The Supported header field enumerates all the extensions supported by
the client or server using feature tags. The header carries the
extensions supported by the message sending entity. The Supported
header MAY be included in any request. When present in a request,
the receiver MUST respond with its corresponding Supported header.
Note, also in 4xx and 5xx responses is the supported header included.
The Supported header field contains a list of feature-tags, described
in Section 3.7, that are understood by the client or server.
Example:
C->S: OPTIONS rtsp://example.com/ RTSP/1.1
Supported: foo, bar, blech
S->C: RTSP/1.1 200 OK
Supported: bar, blech, baz
14.44 Timestamp
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The Timestamp general-header field describes when the agent sent the
request. The value of the timestamp is of significance only to the
agent and may use any timescale. The responding agent MUST echo the
exact same value and MAY, if it has accurate information about this,
add a floating point number indicating the number of seconds that has
elapsed since it has received the request. The timestamp is used by
the agent to compute the round-trip time to the responding agent so
that it can adjust the timeout value for retransmissions. It also
resolves retransmission ambiguities for unreliable transport of RTSP.
14.45 Transport
The Transport request and response header field indicates which
transport protocol is to be used and configures its parameters such
as destination address, compression, multicast time-to-live and
destination port for a single stream. It sets those values not
already determined by a presentation description.
Transports are comma separated, listed in order of preference.
Parameters may be added to each transport, separated by a semicolon.
The server SHOULD return a Transport response-header field in the
response to indicate the values actually chosen. The Transport header
field MAY also be used to change certain transport parameters. A
server MAY refuse to change parameters of an existing stream.
A Transport request header field MAY contain a list of transport
options acceptable to the client, in the form of multiple
transportspec entries. In that case, the server MUST return the
single (transport-spec) which was actually chosen. The number of
transportspec entries is expected to be limited as the client will
get guidance on what configurations that are possible from the
presentation description.
A transport-spec transport option may only contain one of any given
parameter within it. Parameters MAY be given in any order.
Additionally, it may only contain the unicast or the multicast
transport type parameter. Unknown parameters SHALL be ignored. The
requester need to ensure that the responder understands the
parameters through the use of feature tags and the Require header.
Any parameters part of future extensions requires clarification if
they are safe to ignore in accordance to this specification, or is
required to be understood. If a parameter is required to be
understood, then a feature tag MUST be defined for the functionality
and used in the Require and/or Proxy-Require headers.
The Transport header field is restricted to describing a
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single media stream. (RTSP can also control multiple
streams as a single entity.) Making it part of RTSP rather
than relying on a multitude of session description formats
greatly simplifies designs of firewalls.
The syntax for the transport specifier is
transport/profile/lower-transport.
The default value for the "lower-transport" parameters is specific to
the profile. For RTP/AVP, the default is UDP.
There are two different methods for how to specify where the media
should be delivered:
o The presence of this parameter and its values indicates the
destination address or addresses (host address and port pairs
for IP flows) necessary for the media transport.
o The lack of the dest_addr parameter indicates that the server
SHALL send media to same address for which the RTSP messages
originates. Does not work for transports requiring explicitly
given destination ports.
The choice of method for indicating where the media is to be
delivered depends on the use case. In many case the only allowed
method will be to use no explicit address indication and have the
server deliver media to the source of the RTSP messages.
An RTSP proxy will need to take care. If the media is not desired to
be routed through the proxy, the proxy will need to introduce the
destination indication.
Below are the configuration parameters associated with transport:
General parameters:
unicast / multicast: This parameter is a mutually exclusive
indication of whether unicast or multicast delivery will be
attempted. One of the two values MUST be specified. Clients
that are capable of handling both unicast and multicast
transmission needs to indicate such capability by including
two full transport-specs with separate parameters for each.
layers: The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses
starting at the dest_addr address.
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dest_addr: A general destination address parameter that can
contain one or more address specifications. Each
combination of Protocol/Profile/Lower Transport needs to
have the format and interpretation of its address
specification defined. For RTP/AVP/UDP and RTP/AVP/TCP,
the address specification is a tuple containing a host
address and port.
The client originating the RTSP request MAY specify the
destination address of the stream recipient with the host
address part of the tuple. When the destination address is
specified, the recipient may be a different party than the
originator of the request. To avoid becoming the unwitting
perpetrator of a remote-controlled denial-of-service
attack, a server MUST perform security checks (see Section
20.1) and SHOULD log such attempts before allowing the
client to direct a media stream to a recipient address not
chosen by the server. Implementations cannot rely on TCP as
reliable means of client identification. If the server does
not allow the host address part of the tuple to be set, it
SHALL return 463 (Destination Prohibited).
The host address part of the tuple MAY be empty, for
example ":58044", in cases when only destination port is
desired to be specified.
src_addr: A general source address parameter that can contain
one or more address specifications. Each combination of
Protocol/Profile/Lower Transport needs to have the format
and interpretation of its address specification defined.
For RTP/AVP/UDP and RTP/AVP/TCP, the address specification
is a tuple containing a host address and port.
This parameter MUST be specified by the server if it
transmits media packets from another address than the one
RTSP messages are sent to. This will allow the client to
verify source address and give it a destination address for
its RTCP feedback packets if RTP is used. The address or
addresses indicated in the src_addr parameter SHOULD be
used both for sending and receiving of the media streams
data packets. The main reasons are threefold: First,
indicating the port and source address(s) lets the receiver
know where from the packets is expected to originate.
Secondly, traversal of NATs are greatly simplified when
traffic is flowing symmetrically over a NAT binding.
Thirdly, certain NAT traversal mechanisms, needs to know to
which address and port to send so called "binding packets"
from the receiver to the sender, thus creating a address
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binding in the NAT that the sender to receiver packet flow
can use.
This information may also be available through SDP.
However, since this is more a feature of transport
than media initialization, the authoritative source
for this information should be in the SETUP response.
mode: The mode parameter indicates the methods to be supported
for this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY. The RECORD value was
defined in RFC 2326 and is deprecated in this
specification.
append: The append parameter was used together with RECORD and
is now deprecated.
interleaved: The interleaved parameter implies mixing the media
stream with the control stream in whatever protocol is
being used by the control stream, using the mechanism
defined in Section 12. The argument provides the channel
number to be used in the $ statement and MUST be present.
This parameter MAY be specified as a range, e.g.,
interleaved=4-5 in cases where the transport choice for the
media stream requires it, e.g. for RTP with RTCP. The
channel number given in the request are only a guidance
from the client to the server on what channel number(s) to
use. The server MAY set any valid channel number in the
response. The declared channel(s) are bi-directional, so
both end-parties MAY send data on the given channel. One
example of such usage is the second channel used for RTCP,
where both server and client sends RTCP packets on the same
channel.
This allows RTP/RTCP to be handled similarly to the
way that it is done with UDP, i.e., one channel for
RTP and the other for RTCP.
Multicast-specific:
ttl: multicast time-to-live.
RTP-specific:
These parameters are MAY only be used if the media transport protocol
is RTP.
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ssrc: The ssrc parameter, if included in a SETUP response,
indicates the RTP SSRC [16] value(s) that will be used by
the media server for RTP packets within the stream. It is
expressed as an eight digit hexadecimal value.
The ssrc parameter SHALL NOT be specified in requests. The
functionality of specifying the ssrc parameter in a SETUP
request is deprecated as it is incompatible with the
specification of RTP in RFC 3550 [16]. If the parameter is
included in the Transport header of a SETUP request, the
server MAY ignore it, and choose an appropriate SSRC for
the stream. The server MAY set the ssrc parameter in the
Transport header of the response.
The combination of transport protocol, profile and lower transport
needs to be defined. A number of combinations are defined in the
appendix B.
Below is a usage example, showing a client advertising the capability
to handle multicast or unicast, preferring multicast. Since this is
a unicast-only stream, the server responds with the proper transport
parameters for unicast.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.1
CSeq: 302
Transport: RTP/AVP;multicast;mode="PLAY",
RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";mode="PLAY"
S->C: RTSP/1.1 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:3456"/
"192.0.2.5:3457";src_addr="192.0.2.224:6256"
/"192.0.2.224:6257";mode="PLAY"
14.46 Unsupported
The Unsupported response-header field lists the features not
supported by the server. In the case where the feature was specified
via the Proxy-Require field (Section 14.31), if there is a proxy on
the path between the client and the server, the proxy MUST send a
response message with a status code of 551 (Option Not Supported).
The request SHALL NOT be forwarded.
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See Section 14.37 for a usage example.
14.47 User-Agent
See [H14.43] for explanation, however the syntax is clarified due to
an error in RFC 2616. A Client SHOULD include this header in all RTSP
messages it sends.
14.48 Vary
See [H14.44]
14.49 Via
See [H14.45].
14.50 WWW-Authenticate
See [H14.47].
15 Proxies
RTSP Proxies are RTSP agents that sit in between a client and a
server. A proxy can take on both the role as a client and as server
depending on what it tries to accomplish. Proxies are also introduced
for several different reasons.
o This type of proxy is used to reduce the workload on servers
and connections. By caching a presentation, both description
and media streams the proxy can serve a client content without
requesting it from the server once it has been cached and
hasn't become stale. See the caching section 16.
o This type of proxy is used to ensure that a RTSP client get
access to servers on an external network. Thus this proxy is
placed on the border between two domains, e.g. a private
address space and the public internet. The proxy performs the
necessary translation, usually addresses, and often also media
stream translation or redirection.
o This type of proxy is used to help facilitate security
functions around RTSP. For example when having a firewalled
network, the security proxy request that the necessary
pinholes in the firewall is opened when a client in the
protected network want to access media streams on the external
side. It can also provide network owners with a logging and
audit point for RTSP sessions, e.g. for corporations that
tracks or limits their employees access to certain type of
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content.
All type of proxies can be used also when using secured communication
with TLS as RTSP 1.1 allows the client to approve certificates for
connection establishment from a proxy, see section 18.3.2.
Access proxies SHOULD NOT be used in equipment like NATs and
firewalls that aren't expected to be regularly maintained, like home
or small office equipment. In these cases it is better to use the NAT
traversal procedures defined for RTSP 1.1 [25]. The reason for these
recommendations is that any extensions of RTSP resulting in new media
transport protocols or profiles, new parameters etc may fail in a
proxy that isn't maintained. Thus resulting in blocking further
development of RTSP and its usage.
The existence of proxies must always be considered when developing
new RTSP extensions. There must be definition of how proxies may
handle the extension, if it is required to understand it, thus
requiring a feature tag to be used in the Proxy-Require header.
16 Caching
In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE.
(Since the responses for anything but DESCRIBE and GET_PARAMETER do
not return any data, caching is not really an issue for these
requests.) However, it is desirable for the continuous media data,
typically delivered out-of-band with respect to RTSP, to be cached,
as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up-to-date by
issuing a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is not
up-to-date, it modifies the SETUP transport parameters as appropriate
and forwards the request to the origin server. Subsequent control
commands such as PLAY or PAUSE then pass the proxy unmodified. The
proxy delivers the continuous media data to the client, while
possibly making a local copy for later reuse. The exact behavior
allowed to the cache is given by the cache-response directives
described in Section 14.10. A cache MUST answer any DESCRIBE requests
if it is currently serving the stream to the requestor, as it is
possible that low-level details of the stream description may have
changed on the origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
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through" variety. Rather than retrieving the whole resource from the
origin server, the cache simply copies the streaming data as it
passes by on its way to the client. Thus, it does not introduce
additional latency.
To the client, an RTSP proxy cache appears like a regular media
server, to the media origin server like a client. Just as an HTTP
cache has to store the content type, content language, and so on for
the objects it caches, a media cache has to store the presentation
description. Typically, a cache eliminates all transport-references
(that is, multicast information) from the presentation description,
since these are independent of the data delivery from the cache to
the client. Information on the encodings remains the same. If the
cache is able to translate the cached media data, it would create a
new presentation description with all the encoding possibilities it
can offer.
17 Examples
This section contains several different examples trying to illustrate
possible ways of using RTSP. The examples can also help with the
understanding of how functions of RTSP work. However remember that
this is examples and the normative and syntax description in the
other sections takes precedence. Please also note that many of the
example MAY contain syntax illegal line breaks to accommodate the
formatting restriction that the RFC series impose.
17.1 Media on Demand (Unicast)
Client C requests a movie distributed from two different media
servers A (audio.example.com ) and V (video.example.com ). The media
description is stored on a web server W. The media description
contains descriptions of the presentation and all its streams,
including the codecs that are available, dynamic RTP payload types,
the protocol stack, and content information such as language or
copyright restrictions. It may also give an indication about the
timeline of the movie.
In this example, the client is only interested in the last part of
the movie.
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.0 200 OK
Date: 23 Jan 1997 15:35:06 GMT
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Content-Type: application/sdp
Content-Length: 264
Expires: 23 Jan 1998 15:35:06 GMT
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
e=adm@example.com
a=range:npt=0-1:49:34
t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.1
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
RTP/AVP/TCP;unicast;interleaved=0-1
A->C: RTSP/1.1 200 OK
CSeq: 1
Session: 12345678
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057";
src_addr="192.0.2.5:5000"/"192.0.2.5:5001"
Date: 23 Jan 1997 15:35:12 GMT
Server: PhonyServer/1.0
Expires: 24 Jan 1997 15:35:12 GMT
Cache-Control: public
Accept-Ranges: NPT, SMPTE
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.1
CSeq: 1
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;dest_addr=":3058"/":3059",
RTP/AVP/TCP;unicast;interleaved=0-1
V->C: RTSP/1.1 200 OK
CSeq: 1
Session: 23456789
Transport: RTP/AVP/UDP;unicast;dest_addr=":3058"/":3059";
src_addr="192.0.2.5:5002"/"192.0.2.5:5003"
Date: 23 Jan 1997 15:35:12 GMT
Server: PhonyServer/1.0
Cache-Control: public
Expires: 24 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT, SMPTE
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C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.1
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 23456789
Range: smpte=0:10:00-
V->C: RTSP/1.1 200 OK
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-1:49:23
RTP-Info: url="rtsp://video.example.com/twister/video"
ssrc=A17E189D:seq=12312232;rtptime=78712811
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:35:13 GMT
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.1
CSeq: 2
User-Agent: PhonyClient/1.2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/1.1 200 OK
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-1:49:23
RTP-Info: url="rtsp://audio.example.com/twister/audio.en"
ssrc=3D124F01:seq=876655;rtptime=1032181
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:13 GMT
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.1
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 12345678
A->C: RTSP/1.1 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:52 GMT
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.1
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 23456789
V->C: RTSP/1.1 200 OK
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CSeq: 3
Server: PhonyServer/2.0
Date: 23 Jan 1997 15:36:52 GMT
Even though the audio and video track are on two different servers,
may start at slightly different times, and may drift with respect to
each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender
reports. Initial synchronization is achieved through the RTP-Info and
Range headers information in the PLAY response.
17.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in
which multiple continuous media types pertaining to the same end-user
presentation are present. In effect, the container file represents an
RTSP presentation, with each of its components being RTSP streams.
Container files are a widely used means to store such presentations.
While the components are transported as independent streams, it is
desirable to maintain a common context for those streams at the
server end.
This enables the server to keep a single storage handle
open easily. It also allows treating all the streams
equally in case of any prioritization of streams by the
server.
It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by the client in order to preserve
the artistic effect of the combined media presentation. Similarly, in
such a tightly bound presentation, it is desirable to be able to
control all the streams via a single control message using an
aggregate URI.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URIs. In a
container file it is also desirable to not write any URI parts which
is not kept, when the container is distributed, like the host and
most of the path element. Therefore this example also uses the "*"
and relative URI in the delivered SDP.
Client C requests a presentation from media server M. The movie is
stored in a container file. The client has obtained an RTSP URI to
the container file.
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C->M: DESCRIBE rtsp://example.com/twister.3gp RTSP/1.1
CSeq: 1
User-Agent: PhonyClient/1.2
M->C: RTSP/1.1 200 OK
CSeq: 1
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 257
Content-Base: rtsp://example.com/twister.3gp/
Expires: 24 Jan 1997 15:35:06 GMT
v=0
o=- 2890844256 2890842807 IN IP4 172.16.2.93
s=RTSP Session
i=An Example of RTSP Session Usage
e=adm@example.com
a=control: *
a=range: npt=0-0:10:34.10
t=0 0
m=audio 0 RTP/AVP 0
a=control: trackID=1
m=video 0 RTP/AVP 26
a=control: trackID=4
C->M: SETUP rtsp://example.com/twister.3gp/trackID=1 RTSP/1.1
CSeq: 2
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001"
M->C: RTSP/1.1 200 OK
CSeq: 2
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;dest_addr=":8000"/":8001;
src_addr="192.0.2.5:9000"/"192.0.2.5:9001"
ssrc=93CB001E
Session: 12345678
Expires: 24 Jan 1997 15:35:12 GMT
Date: 23 Jan 1997 15:35:12 GMT
Accept-Ranges: NPT
C->M: SETUP rtsp://example.com/twister.3gp/trackID=4 RTSP/1.1
CSeq: 3
User-Agent: PhonyClient/1.2
Require: play.basic
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003"
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Session: 12345678
M->C: RTSP/1.1 200 OK
CSeq: 3
Server: PhonyServer/1.0
Transport: RTP/AVP;unicast;dest_addr=":8002"/":8003;
src_addr="192.0.2.5:9002"/"192.0.2.5:9003";
ssrc=A813FC13
Session: 12345678
Expires: 24 Jan 1997 15:35:13 GMT
Date: 23 Jan 1997 15:35:13 GMT
Accept-Range: NPT
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.1
CSeq: 4
User-Agent: PhonyClient/1.2
Range: npt=0-10, npt=30-
Session: 12345678
M->C: RTSP/1.1 200 OK
CSeq: 4
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:14 GMT
Session: 12345678
Range: npt=0-10, npt=30-623.10
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12345;rtptime=3450012,
url="rtsp://example.com/twister.3gp/trackID=1";
ssrc=4F312DD8:seq=54321;rtptime=2876889
C->M: PAUSE rtsp://example.com/twister.3gp/ RTSP/1.1
CSeq: 5
User-Agent: PhonyClient/1.2
Session: 12345678
M->C: RTSP/1.1 200 OK
CSeq: 5
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=34.57-623.10
C->M: PLAY rtsp://example.com/twister.3gp/ RTSP/1.1
CSeq: 6
User-Agent: PhonyClient/1.2
Range: npt=34.57-623.10
Session: 12345678
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M->C: RTSP/1.1 200 OK
CSeq: 6
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:36:01 GMT
Session: 12345678
Range: npt=34.57-623.10
RTP-Info: url="rtsp://example.com/twister.3gp/trackID=4"
ssrc=0D12F123:seq=12555;rtptime=6330012,
url="rtsp://example.com/twister.3gp/trackID=1"
ssrc=4F312DD8:seq=55021;rtptime=3132889
17.3 Single Stream Container Files
Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients SHOULD use the rules set forth in the session
description for Request-URIs, rather than assuming that a consistent
URI may always be used throughout. Below are an example of how a
multi-stream server might expect a single-stream file to be served:
C->S: DESCRIBE rtsp://foo.com/test.wav RTSP/1.1
Accept: application/x-rtsp-mh, application/sdp
CSeq: 1
User-Agent: PhonyClient/1.2
S->C: RTSP/1.1 200 OK
CSeq: 1
Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp
Content-length: 148
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:06 GMT
Expires: 23 Jan 1997 17:00:00 GMT
v=0
o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file
i=audio test
t=0 0
a=control: *
m=audio 0 RTP/AVP 0
a=control:streamid=0
C->S: SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.1
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Transport: RTP/AVP/UDP;unicast;
dest_addr=":6970"/":6971";mode="PLAY"
CSeq: 2
User-Agent: PhonyClient/1.2
S->C: RTSP/1.1 200 OK
Transport: RTP/AVP/UDP;unicast;dest_addr=":6970"/":6971";
src_addr="192.0.2.5:6970"/"192.0.2.5:6971";
mode="PLAY";ssrc=EAB98712
CSeq: 2
Session: 2034820394
Expires: 23 Jan 1997 16:00:00 GMT
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
C->S: PLAY rtsp://foo.com/test.wav/ RTSP/1.1
CSeq: 3
User-Agent: PhonyClient/1.2
Session: 2034820394
S->C: RTSP/1.1 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:08 GMT
Session: 2034820394
Range: npt=0-600
RTP-Info: url="rtsp://foo.com/test.wav/streamid=0"
ssrc=0D12F123:seq=981888;rtptime=3781123
Note the different URI in the SETUP command, and then the switch back
to the aggregate URI in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but is less
than intuitive in the special case where the number of streams is
one. However the server has declared that the aggregated control URI
in the SDP and therefore this is legal.
In this case, it is also required that servers accept implementations
that use the non-aggregated interpretation and use the individual
media URI, like this:
C->S: PLAY rtsp://example.com/test.wav/streamid=0 RTSP/1.1
CSeq: 3
User-Agent: PhonyClient/1.2
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17.4 Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, it
is assumed that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
C->W: GET /sessions.html HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: text/html
<html>
...
<href "Stremed Live Music performance"
src="rtsp://live.example.com/concert/audio">
...
</html>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.1
CSeq: 1
Supported: play.basic, play.scale
M->C: RTSP/1.1 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 182
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:06 GMT
Supported: play.basic
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 3456 RTP/AVP 0
c=IN IP4 224.2.0.1/16
a=control: rtsp://live.example.com/concert/audio
a=range:npt=0-
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.1
CSeq: 2
Transport: RTP/AVP;multicast
M->C: RTSP/1.1 200 OK
CSeq: 2
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:06 GMT
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Transport: RTP/AVP;multicast;dest_addr="224.2.0.1:3456"/"
224.2.0.1:3457";ttl=16
Session: 0456804596
Accept-Ranges: NPT, UTC
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.1
CSeq: 3
Session: 0456804596
M->C: RTSP/1.1 200 OK
CSeq: 3
Server: PhonyServer/1.0
Date: 23 Jan 1997 15:35:07 GMT
Session: 0456804596
Range:npt=1256-
RTP-Info: url="rtsp://live.example.com/concert/audio"
ssrc=0D12F123:seq=1473; rtptime=80000
17.5 Capability Negotiation
This examples illustrate how the client and server determines their
capability to support a special feature, in this case "play.scale".
The server, through the clients request and the included Supported
header, learns that the client is supporting this updated
specification, and also supports the playback time scaling feature of
RTSP. The server's response contains the following feature related
information to the client; it supports the updated specification
(play.basic), the extended functionality of time scaling of content
(play.scale), and one "example.com" proprietary feature
(example.com.flight). The client also learns the methods supported
(Public header) by the server for the indicated resource.
C->S: OPTIONS rtsp://media.example.com/movie/twister.3gp RTSP/1.1
CSeq: 1
Supported: play.basic, play.scale
User-Agent: PhonyClient/1.2
S->C: RTSP/1.1 200 OK
CSeq: 1
Public: OPTIONS, SETUP, PLAY, PAUSE, TEARDOWN
Server: PhonyServer/2.0
Supported: play.basic, play.scale, example.com.flight
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When the client sends its SETUP request it tells the server that it
is requires support of the play.scale feature for this session by
including the Require header.
C->S: SETUP rtsp://media.example.com/twister.3gp/trackID=1 RTSP/1.1
CSeq: 3
User-Agent: PhonyClient/1.2
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057",
RTP/AVP/TCP;unicast;interleaved=0-1
Require: play.scale
S->C: RTSP/1.1 200 OK
CSeq: 3
Session: 12345678
Transport: RTP/AVP/UDP;unicast;dest_addr=":3056"/":3057";
src_addr="192.0.2.5:5000"/"192.0.2.5:5001"
Server: PhonyServer/2.0
Accept-Ranges: NPT, SMPTE
18 Security Framework
The RTSP security framework consists of two high level components:
the pure authentication mechanisms based on HTTP authentication, and
the transport protection based on TLS, which is independent of RTSP.
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security for HTTP is re-used to a large extent.
18.1 RTSP and HTTP Authentication
RTSP and HTTP share common authentication schemes, and thus follow
the same usage guidelines as specified in [7] and also in [H15].
Servers SHOULD implement both basic and digest [7] authentication.
It should be stressed that using the HTTP authentication alone does
not provide full control message security. Therefore, in environments
requiring tighter security for the control messages, TLS SHOULD be
used, see Section 18.2.
18.2 RTSP over TLS
RTSP SHALL follow the same guidelines with regards to TLS [6] usage
as specified for HTTP, see [17]. RTSP over TLS is separated from
unsecured RTSP both on URI level and port level. Instead of using the
"rtsp" scheme identifier in the URI, the "rtsps" scheme identifier
MUST be used to signal RTSP over TLS. If no port is given in a URI
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with the "rtsps" scheme, port 322 SHALL be used for TLS over TCP/IP.
When a client tries to setup an insecure channel to the server (using
the "rtsp" URI), and the policy for the resource requires a secure
channel, the server SHALL redirect the client to the secure service
by sending a 301 redirect response code together with the correct
Location URI (using the "rtsps" scheme).
It should be noted that TLS allows for mutual authentication (when
using both server and client certificates). Still, one of the more
common way TLS is used is to only provide server side authentication
(often to avoid client certificates). TLS is then used in addition to
HTTP authentication, providing transport security and server
authentication, while HTTP Authentication is used to authenticate the
client.
RTSP includes the possibility to keep a TCP session up between the
client and server, throughout the RTSP session lifetime. It may be
convenient to keep the TCP session, not only to save the extra setup
time for TCP, but also the extra setup time for TLS (even if TLS uses
the resume function, there will be almost two extra roundtrips).
Still, when TLS is used, such behavior introduces extra active state
in the server, not only for TCP and RTSP, but also for TLS. This may
increase the vulnerability to DoS attacks.
In addition to these recommendations, Section 18.3 gives further
recommendations of TLS usage with proxies.
18.3 Security and Proxies
The nature of a proxy is often to act as a "man-in-the-middle", while
security is often about preventing the existence of a "man-in-the-
middle". This section provides the clients with the possibility to
use proxies even when applying secure transports (TLS). The client
needs to select between using the below specified procedure or using
a TLS connection directly (by-passing any proxies) to the server. The
choice may be dependent on policies.
There are basically two categories of inspecting proxies, the
transparent proxies (which the client is not aware of) and the non-
transparent proxies (which the client is aware of). An infrastructure
based on proxies requires that the trust model is such that both
client and servers can trust the proxies to handle the RTSP messages
correctly. To be able to trust a proxy, the client and server also
needs to be aware of the proxy. Hence, transparent proxies cannot
generally be seen as trusted and will not work well with security
(unless they work only at transport layer). In the rest of this
section any reference to proxy will be to a non-transparent proxy,
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which requires to inspect/manipulate the RTSP messages.
The HTTP Authentication is built on the assumption of proxies and can
provide user-proxy authentication and proxy-proxy/server
authentication in addition to the client-server authentication.
When TLS is applied and a proxy is used, the client will use the
proxy's destination URI address when sending messages. This implies
that for TLS, the client will authenticate the proxy server and not
the end server. Note that, when the client checks the server
certificate in TLS, it MUST check the proxy's identity (URI or
possibly other known identity) against the proxy's identity as
presented in the proxy's Certificate message.
The problem is that for proxy accepted by the client, it needs to be
provided information on which grounds it should accept the next-hop
certificate. Both the proxy and the user may have rules for this, and
the user have the possibility to select the desired behavior. To
handle this case, the Accept-Credentials header (See Section 14.2) is
used, where the client can force the proxy/proxies to relay back the
certificates used by any intermediate proxies as well as the server.
Given the assumption that the proxies are viewed as trusted, it gives
the user a possibility to enforce policies to each trusted proxy of
whether it should accept the next entity in the chain.
A proxy MUST use TLS for the next hop if the RTSP request includes a
"rtsps" URI. TLS MAY be applied on intermediate links (e.g. between
client and proxy, or between proxy and proxy), even if the resource
and the end server does not require to use it.
18.3.1 Accept-Credentials
The Accept-Credentials header can be used by the client to distribute
simple authorization policies to intermediate proxies. The client
includes the Accept-Credentials header to dictate how the proxy
treats the server/next proxy certificate. There are currently three
methods defined:
Any, which means that the proxy (or proxies) SHALL accept
whatever certificate presented. This is of course not a
recommended option to use, but may be useful in certain
circumstances (such as testing).
Proxy, which means that the proxy (or proxies) MUST use its own
policies to validate the certificate and decide whether to
accept it or not. This is convenient in cases where the
user has a strong trust relation with the proxy. Reason why
a strong trust relation may exist are; personal/company
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proxy, proxy has a out-of-band policy configuration
mechanism.
User, which means that the proxy (or proxies) MUST send
credential information about the next hop to the client for
authorization. The client can then decide whether the proxy
should accept the certificate or not. See section 18.3.2
for further details.
If the Accept-Credentials header is not included in the RTSP request
from the client, the default method used SHALL be "Proxy". If
something else than the "Proxy" method is used, the Accept-
Credentials header SHALL always be included in the RTSP request from
the client. This is because it cannot be assumed that the proxy
always keeps the TLS state or the users previously preference between
different RTSP messages (in particular if the time interval between
the messages is long).
The "Any" and "Proxy" methods does not require the proxy to provide
any specific response, but only apply the policy as defined for
respectively method. If the policy do not accept the credentials of
the next hop, the entity SHALL respond with a message using status
code 471 (Connection Credentials not accepted).
An RTSP request in the direction server to client MUST NOT include
the Accept-Credential header. As for the non-secured communication,
the possibility for these request depends on the presence of a client
established connection. However if the server to client request is
in relation to a session established over a TLS secured channel, if
MUST be sent in a TLS secured connection. That secured connection
MUST also be the one used by the last client to server request. If no
such transport connection exist at the time when the server desire to
send the request, it silently fails.
Further policies MAY be defined and registered, but should be done so
with caution.
18.3.2 User approved TLS procedure
For the "User" method each proxy MUST perform the the following
procedure for each RTSP request:
o Setup the TLS session to the next hop if not already present
(i.e. run the TLS handshake, but do not send the RTSP
request).
o Extract the peer certificate for the TLS session.
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o Check if a matching identity and hash of the peer certificate
is present in the Accept-Credentials header. If present, send
the message to the next hop, and conclude these procedures. If
not, go to the next step.
o The proxy responds to the RTSP request with a 470 or 407
response code. The 407 response code MAY be used when the
proxy requires both user and connection authorization from
user or client. In this message the proxy SHALL include a
Connection-Credentials header, see section 14.12 with the next
hop's identity and certificate.
The client MUST upon receiving a 470 or 407 response with
Connection-Credentials header take the decision on whether to accept
the certificate or not (if it cannot do so, the user SHOULD be
consulted). If the certificate is accepted, the client has to again
send the RTSP request. In that request the client has to include the
Accept-Credentials header including the hash over the DER encoded
certificate for all trusted proxies in the chain.
Example:
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/1.1
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
P->C: RTSP/1.1 470 Connection Authorization Required
CSeq: 2
Connection-Credentials: "rtsps://test.example.org";
MIIDNTCCAp...
C->P: SETUP rtsps://test.example.org/secret/audio RTSP/1.1
CSeq: 2
Transport: RTP/AVP;unicast;dest_addr="192.0.2.5:4588"/
"192.0.2.5:4589"
Accept-Credentials: User "rtsps://test.example.org" ;
dPYD 7txp oGTb AqZZ QJ+v aeOk yH4= ...
One implication of this process is that the connection for secured
RTSP messages may take significantly more round-trip times for the
first message. An complete extra message exchange between the proxy
connecting to the next hop and the client results because of the
process for approval for each hop. However after the first message
exchange the remaining message should not be delayed, if each message
contains the chain of proxies that the requestor accepts. The
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procedure of including the credentials in each request rather than
building state in each proxy, avoids the need for revocation
procedures.
19 Syntax
The RTSP syntax is described in an augmented Backus-Naur Form (BNF)
as defined in RFC 2234 [4]. It uses the basic definitions present in
RFC 2234.
19.1 Base Syntax
RTSP header field values can be folded onto multiple lines if the
continuation line begins with a space or horizontal tab. All linear
white space, including folding, has the same semantics as SP. A
recipient MAY replace any linear white space with a single SP before
interpreting the field value or forwarding the message downstream.
This is intended to behave exactly as HTTP/1.1 as described in RFC
2616 [8]. The SWS construct is used when linear white space is
optional, generally between tokens and separators.
To separate the header name from the rest of value, a colon is used,
which, by the above rule, allows whitespace before, but no line
break, and whitespace after, including a linebreak. The HCOLON
defines this construct.
OCTET = %x00-FF ; any 8-bit sequence of data
CHAR = %x01-7F ; any US-ASCII character (octets 1 - 127)
UPALPHA = %x41-5A ; any US-ASCII uppercase letter "A".."Z"
LOALPHA = %x61-7A ;any US-ASCII lowercase letter "a".."z"
ALPHA = UPALPHA / LOALPHA
DIGIT = %x30-39 ; any US-ASCII digit "0".."9"
CTL = %x00-1F / %x7F ; any US-ASCII control character
; (octets 0 - 31) and DEL (127)
CR = %x0D ; US-ASCII CR, carriage return (13
LF = %x0A ; US-ASCII LF, linefeed (10)
SP = %x20 ; US-ASCII SP, space (32)
HT = %x09 ; US-ASCII HT, horizontal-tab (9)
DQUOTE = %x22 ; US-ASCII double-quote mark (34)
BACKSLASH = %x5C ; US-ASCII backslash (92)
CRLF = CR LF
LWS = [CRLF] 1*( SP / HT )
SWS = [LWS] ; sep whitespace
HCOLON = *( SP / HTAB ) ":" SWS
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TEXT = %x20-7D / %x80-FF ; any OCTET except CTLs
tspecials = "(" / ")" / "<" / ">" / "@"
/ "," / ";" / ":" / BACKSLASH / DQUOTE
/ "/" / "[" / "]" / "?" / "="
/ "{" / "}" / SP / HT
token = 1*(%x21 / %x23-27 / %x2A-2B / %x2D-2E / %x30-39
/ %x41-5A / %x5E-7A / %x7C / %x7E)
; 1*<any CHAR except CTLs or tspecials>
quoted-string = ( DQUOTE *qdtext DQUOTE )
qdtext = %x20-21 / %x23-7D / %x80-FF ; any TEXT except <">
quoted-pair = BACKSLASH CHAR
ctext = %x20-27 / %x2A-7D
/ %x80-FF ; any OCTET except CTLs, "(" and ")"
generic-param = token [ EQUAL gen-value ]
gen-value = token / host / quoted-string
safe = "$" / "-" / "_" / "." / "+"
extra = "!" / "*" / "'" / "(" / ")" / ","
rtsp-extra = "!" / "*" / "'" / "(" / ")"
HEX = DIGIT / "A" / "B" / "C" / "D" / "E" / "F"
/ "a" / "b" / "c" / "d" / "e" / "f"
LHEX = DIGIT / %x61-66 ;lowercase a-f
reserved = ";" / "/" / "?" / ":" / "@" / "&" / "="
unreserved = ALPHA / DIGIT / safe / extra
rtsp-unreserved = ALPHA / DIGIT / safe / rtsp-extra
base64 = *base64-unit [base64-pad]
base64-unit = 4base64-char
base64-pad = (2base64-char "==") / (3base64-char "=")
base64-char = ALPHA / DIGIT / "+" / "/"
SLASH = SWS "/" SWS ; slash
EQUAL = SWS "=" SWS ; equal
LPAREN = SWS "(" SWS ; left parenthesis
RPAREN = SWS ")" SWS ; right parenthesis
COMMA = SWS "," SWS ; comma
SEMI = SWS ";" SWS ; semicolon
COLON = SWS ":" SWS ; colon
LDQUOT = SWS DQUOTE ; open double quotation mark
RDQUOT = DQUOTE SWS ; close double quotation mark
RAQUOT = ">" SWS ; right angle quote
LAQUOT = SWS "<" ; left angle quote
TEXT-UTF8char = %x21-7E / UTF8-NONASCII
UTF8-NONASCII = %xC0-DF 1UTF8-CONT
/ %xE0-EF 2UTF8-CONT
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/ %xF0-F7 3UTF8-CONT
/ %xF8-FB 4UTF8-CONT
/ %xFC-FD 5UTF8-CONT
UTF8-CONT = %x80-BF
19.2 RTSP Protocol Definition
19.2.1 Generic Protocol elements
URI-reference = RTSP-URI / relative-ref
relative-ref = < As defined in RFC 3986 [10]>
RTSP-URI = rtsp-uri-def / rtsps-uri-def / rtspu-uri-def
rtsp-uri-def = "rtsp:" rtsp-uri-rest
rtsps-uri-def = "rtsps:" rtsp-uri-rest
rtspu-uri-def = "rtspu:" rtsp-uri-rest
rtsp-uri-rest = "//" host [":" port] [abs-path-def] [frag-def]
abs-path-def = abs-path ["?" query]
frag-def = "#" fragment
host = <As defined by RFC 3986 [10]>
abs-path = <As defined by RFC 3986 [10]>
port = *DIGIT ; Is expected to be 1*5DIGIT
query = <As defined by RFC 3986 [10]>
fragment = <As defined by RFC 3986 [10]>
absolute-URI = <As defined by RFC 3986 [10]>
IPv4address = 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT "." 1*3DIGIT
IPv6address = hexpart [ ":" IPv4address ]
hexpart = hexseq / hexseq "::" [ hexseq ] / "::" [ hexseq ]
hexseq = hex4 *( ":" hex4)
hex4 = 1*4HEXDIG
smpte-range = smpte-type "=" smpte-range-spec
;Section 3.4
smpte-range-spec = ( smpte-time "-" [ smpte-time ] )
/ ( "-" smpte-time )
smpte-type = "smpte" / "smpte-30-drop"
/ "smpte-25" / smpte-type-extension
; other timecodes may be added
smpte-type-extension = token
smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT
[ ":" 1*2DIGIT [ "." 1*2DIGIT ] ]
npt-range = "npt=" npt-range-spec ; Section 3.5
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npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time )
npt-time = "now" / npt-sec / npt-hhmmss
npt-sec = 1*DIGIT [ "." *DIGIT ]
npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ]
npt-hh = 1*DIGIT ; any positive number
npt-mm = 1*2DIGIT ; 0-59
npt-ss = 1*2DIGIT ; 0-59
utc-range = "clock=" utc-range-spec ; Section 3.6
utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time )
utc-time = utc-date "T" utc-clock "Z"
utc-date = 8DIGIT ; < YYYYMMDD >
utc-clock = 6DIGIT [ "." fraction ]; < HHMMSS.fraction >
fraction = 1*DIGIT
feature-tag = token
session-id = 8*( ALPHA / DIGIT / safe )
extension-header = header-name HCOLON header-value
header-name = token
header-value = *(TEXT-UTF8char / UTF8-CONT / LWS)
19.2.2 Message Syntax
RTSP-message = Request / Response ; RTSP/1.1 messages
Request = Request-Line ; Section 6.1
*( general-header ; Section 5
/ request-header ; Section 6.2
/ entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
Response = Status-Line ; Section 7.1
*( general-header ; Section 5
/ response-header ; Section 7.1.2
/ entity-header ) ; Section 8.1
CRLF
[ message-body ] ; Section 4.3
Request-Line = Method SP Request-URI SP RTSP-Version CRLF
Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF
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Method = "DESCRIBE" ; Section 11.2
/ "GET_PARAMETER" ; Section 11.7
/ "OPTIONS" ; Section 11.1
/ "PAUSE" ; Section 11.5
/ "PLAY" ; Section 11.4
/ "REDIRECT" ; Section 11.9
/ "SETUP" ; Section 11.3
/ "SET_PARAMETER" ; Section 11.8
/ "TEARDOWN" ; Section 11.6
/ extension-method
extension-method = token
Request-URI = "*" / RTSP-URI
RTSP-Version = "RTSP/" 1*DIGIT "." 1*DIGIT
message-body = 1*OCTET
Status-Code = "100" ; Continue
/ "200" ; OK
/ "201" ; Created
/ "250" ; Low on Storage Space
/ "300" ; Multiple Choices
/ "301" ; Moved Permanently
/ "302" ; Moved Temporarily
/ "303" ; See Other
/ "304" ; Not Modified
/ "305" ; Use Proxy
/ "400" ; Bad Request
/ "401" ; Unauthorized
/ "402" ; Payment Required
/ "403" ; Forbidden
/ "404" ; Not Found
/ "405" ; Method Not Allowed
/ "406" ; Not Acceptable
/ "407" ; Proxy Authentication Required
/ "408" ; Request Time-out
/ "410" ; Gone
/ "411" ; Length Required
/ "412" ; Precondition Failed
/ "413" ; Request Entity Too Large
/ "414" ; Request-URI Too Large
/ "415" ; Unsupported Media Type
/ "451" ; Parameter Not Understood
/ "452" ; reserved
/ "453" ; Not Enough Bandwidth
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/ "454" ; Session Not Found
/ "455" ; Method Not Valid in This State
/ "456" ; Header Field Not Valid for Resource
/ "457" ; Invalid Range
/ "458" ; Parameter Is Read-Only
/ "459" ; Aggregate operation not allowed
/ "460" ; Only aggregate operation allowed
/ "461" ; Unsupported transport
/ "462" ; Destination unreachable
/ "463" ; Destination Prohibited
/ "470" ; Connection Authorization Required
/ "471" ; Connection Credentials not accepted
/ "500" ; Internal Server Error
/ "501" ; Not Implemented
/ "502" ; Bad Gateway
/ "503" ; Service Unavailable
/ "504" ; Gateway Time-out
/ "505" ; RTSP Version not supported
/ "551" ; Option not supported
/ extension-code
extension-code = 3DIGIT
Reason-Phrase = *TEXT
general-header = Cache-Control ; Section 14.10
/ Connection ; Section 14.11
/ CSeq ; Section 14.19
/ Date ; Section 14.20
/ Proxy-Supported ; Section 14.32
/ Supported ; Section 14.43
/ Timestamp ; Section 14.44
/ Via ; Section 14.49
/ extension-header
request-header = Accept ; Section 14.1 and [H14.1]
/ Accept-Credentials ; Section 14.2
/ Accept-Encoding ; Section 14.3 and [H14.3]
/ Accept-Language ; Section 14.4 and [H14.4]
/ Authorization ; Section 14.7 and [H14.8]
/ Bandwidth ; Section 14.8
/ Blocksize ; Section 14.9
/ From ; Section 14.23
/ If-Match ; Section 14.25
/ If-Modified-Since ; Section 14.26 and [H14.25]
/ If-None-Match ; Section 14.27
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/ Proxy-Require ; Section 14.31
/ Range ; Section 14.34
/ Referer ; Section 14.35
/ Require ; Section 14.37
/ Scale ; Section 14.39
/ Session ; Section 14.42
/ Speed ; Section 14.40
/ Supported ; Section 14.43
/ Transport ; Section 14.45
/ User-Agent ; Section 14.47
/ extension-header
response-header = Accept-Credentials ; Section 14.2
/ Accept-Ranges ; Section 14.5
/ Connection-Creds ; Section 14.12
/ ETag ; Section 14.21
/ Location ; Section 14.29
/ Proxy-Authenticate ; Section 14.30
/ Public ; Section 14.33
/ Range ; Section 14.34
/ Retry-After ; Section 14.36
/ RTP-Info ; Section 14.38
/ Scale ; Section 14.39
/ Session ; Section 14.42
/ Server ; Section 14.41
/ Speed ; Section 14.40
/ Transport ; Section 14.45
/ Unsupported ; Section 14.46
/ Vary ; Section 14.48
/ WWW-Authenticate ; Section 14.50
/ extension-header
entity-header = Allow ; Section 14.6
/ Content-Base ; Section 14.13
/ Content-Encoding ; Section 14.14
/ Content-Language ; Section 14.15
/ Content-Length ; Section 14.16
/ Content-Location ; Section 14.17
/ Content-Type ; Section 14.18
/ Expires ; Section 14.22 and [H14.21]
/ Last-Modified ; Section 14.28
/ extension-header
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19.2.3 Header Syntax
All header syntaxes not defined in this section are defined in
section 14 of the HTTP 1.1 specification [3].
Accept = "Accept" HCOLON
[ accept-range *(COMMA accept-range) ]
accept-range = media-range *(SEMI accept-param)
media-range = ( "*/*"
/ ( m-type SLASH "*" )
/ ( m-type SLASH m-subtype )
) *( SEMI m-parameter )
accept-param = ("q" EQUAL qvalue) / generic-param
qvalue = ( "0" [ "." *3DIGIT ] )
/ ( "1" [ "." *3("0") ] )
Accept-Credentials = "Accept-Credentials" HCOLON cred-decision CRLF
cred-decision = ("User" COMMA [cred-info])
/ "Proxy"
/ "Any"
/ token ; For future extensions
cred-info = cred-info-data *(COMMA cred-info-data)
cred-info-data = DQUOTE RTSP-URI DQUOTE SEMI base64
Accept-Encoding = "Accept-Encoding" HCOLON
[ encoding *(COMMA encoding) ]
encoding = codings *(SEMI accept-param)
codings = content-coding / "*"
content-coding = token
Accept-Language = "Accept-Language" HCOLON
[ language *(COMMA language) ]
language = language-range *(SEMI accept-param)
language-range = ( ( 1*8ALPHA *( "-" 1*8ALPHA ) ) / "*" )
Accept-Ranges = "Accept-Ranges" HCOLON acceptable-ranges CRLF
acceptable-ranges = (range-unit *(COMMA range-unit))
/ "none"
range-unit = "NPT" / "SMPTE" / "UTC" / extension-format
extension-format = token
Allow = "Allow" HCOLON [Method *(COMMA Method)]
Authorization = "Authorization" HCOLON credentials
credentials = ("Digest" LWS digest-response)
/ other-response
digest-response = dig-resp *(COMMA dig-resp)
dig-resp = username / realm / nonce / digest-uri
/ dresponse / algorithm / cnonce
/ opaque / message-qop
/ nonce-count / auth-param
username = "username" EQUAL username-value
username-value = quoted-string
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digest-uri = "uri" EQUAL LDQUOT digest-uri-value RDQUOT
digest-uri-value = Request-URI
; by HTTP/1.1
message-qop = "qop" EQUAL qop-value
cnonce = "cnonce" EQUAL cnonce-value
cnonce-value = nonce-value
nonce-count = "nc" EQUAL nc-value
nc-value = 8LHEX
dresponse = "response" EQUAL request-digest
request-digest = LDQUOT 32LHEX RDQUOT
auth-param = auth-param-name EQUAL
( token / quoted-string )
auth-param-name = token
other-response = auth-scheme LWS auth-param
*(COMMA auth-param)
auth-scheme = token
Bandwidth = "Bandwidth" HCOLON 1*DIGIT CRLF
Blocksize = "Blocksize" HCOLON 1*DIGIT CRLF
Cache-Control = "Cache-Control" HCOLON cache-directive CRLF
*(COMMA cache-directive)
cache-directive = cache-rqst-directive
/ cache-rspns-directive
cache-rqst-directive = "no-cache"
/ "max-stale" [EQUAL delta-seconds]
/ "min-fresh" EQUAL delta-seconds
/ "only-if-cached"
/ cache-extension
cache-rspns-directive = "public"
/ "private"
/ "no-cache"
/ "no-transform"
/ "must-revalidate"
/ "proxy-revalidate"
/ "max-age" EQUAL delta-seconds
/ cache-extension
cache-extension = token [EQUAL (token / quoted-string)]
delta-seconds = 1*DIGIT
Connection-Creds = "Connection-Credentials" HCOLON cred-info CRLF
Connection = "Connection" HCOLON (connection-token)
*(COMMA connection-token) CRLF
connection-token = token
Content-Base = "Content-Base" HCOLON URI-reference CRLF
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Content-Encoding = "Content-Encoding" HCOLON
content-coding *(COMMA content-coding)
Content-Language = "Content-Language" HCOLON
language-tag *(COMMA language-tag)
language-tag = primary-tag *( "-" subtag )
primary-tag = 1*8ALPHA
subtag = 1*8ALPHA
Content-Length = "Content-Length" HCOLON 1*DIGIT
Content-Location = "Content-Location" HCOLON URI-reference
Content-Type = ( "Content-Type" / "c" ) HCOLON media-type
media-type = m-type SLASH m-subtype *(SEMI m-parameter)
m-type = discrete-type / composite-type
discrete-type = "text" / "image" / "audio" / "video"
/ "application" / extension-token
composite-type = "message" / "multipart" / extension-token
extension-token = ietf-token / x-token
ietf-token = token
x-token = "x-" token
m-subtype = extension-token / iana-token
iana-token = token
m-parameter = m-attribute EQUAL m-value
m-attribute = token
m-value = token / quoted-string
CSeq = "Cseq" HCOLON 1*DIGIT CRLF
Date = "Date" HCOLON RTSP-date
RTSP-date = rfc1123-date ; HTTP-date
rfc1123-date = wkday "," SP date1 SP time SP "GMT"
date1 = 2DIGIT SP month SP 4DIGIT
; day month year (e.g., 02 Jun 1982)
time = 2DIGIT ":" 2DIGIT ":" 2DIGIT
; 00:00:00 - 23:59:59
wkday = "Mon" / "Tue" / "Wed"
/ "Thu" / "Fri" / "Sat" / "Sun"
month = "Jan" / "Feb" / "Mar" / "Apr"
/ "May" / "Jun" / "Jul" / "Aug"
/ "Sep" / "Oct" / "Nov" / "Dec"
ETag = "ETag" HCOLON entity-tag
Expires = "Expires" HCOLON delta-seconds
From = "From" HCOLON from-spec
from-spec = ( name-addr / addr-spec ) *( SEMI from-param )
name-addr = [ display-name ] LAQUOT addr-spec RAQUOT
addr-spec = RTSP-URI / absolute-URI
display-name = *(token LWS)/ quoted-string
from-param = tag-param / generic-param
tag-param = "tag" EQUAL token
If-Match = "If-Match" HCOLON ( "*" / entity-tag-list)
entity-tag-list = entity-tag *(COMMA entity-tag)
entity-tag = [ weak ] opaque-tag
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weak = "W/"
opaque-tag = quoted-string
If-Modified-Since = "If-Modified-Since" HCOLON RTSP-date
If-None-Match = "If-None-Match" HCOLON ("*" / entity-tag-list)
Last-Modified = "Last-Modified" HCOLON RTSP-date
Location = "Location" HCOLON RTSP-URI
Proxy-Authenticate = "Proxy-Authenticate" HCOLON challenge
challenge = ("Digest" LWS digest-cln *(COMMA digest-cln))
/ other-challenge
other-challenge = auth-scheme LWS auth-param
*(COMMA auth-param)
digest-cln = realm / domain / nonce
/ opaque / stale / algorithm
/ qop-options / auth-param
realm = "realm" EQUAL realm-value
realm-value = quoted-string
domain = "domain" EQUAL LDQUOT URI
*( 1*SP URI ) RDQUOT
URI = RTSP-URI / abs-path
nonce = "nonce" EQUAL nonce-value
nonce-value = quoted-string
opaque = "opaque" EQUAL quoted-string
stale = "stale" EQUAL ( "true" / "false" )
algorithm = "algorithm" EQUAL ("MD5" / "MD5-sess" / token)
qop-options = "qop" EQUAL LDQUOT qop-value
*("," qop-value) RDQUOT
qop-value = "auth" / "auth-int" / token
Proxy-Require = "Proxy-Require" HCOLON feature-tag CRLF
*(COMMA feature-tag)
Proxy-Supported = "Proxy-Supported" HCOLON feature-tag
*(COMMA feature-tag) CRLF
Public = "Public" HCOLON Method *(COMMA Method) CRLF
Range = "Range" HCOLON ranges-list [exec-time] CRLF
ranges-list = ranges-spec *(COMMA ranges-spec)
exec-time = SEMI "time" EQUAL utc-time
ranges-spec = npt-range / utc-range / smpte-range
Referer = "Referer" HCOLON URI-reference
Require = "Require" HCOLON feature-tag-list CRLF
feature-tag-list = feature-tag *(COMMA feature-tag)
RTP-Info = "RTP-Info" HCOLON rtsp-info-spec
*(COMMA rtsp-info-spec) CRLF
rtsp-info-spec = stream-url 1*ssrc-parameter
stream-url = "url" EQUAL DQUOTE URI-reference DQUOTE
ssrc-parameter = LWS "ssrc" EQUAL ssrc HCOLON
ri-parameter *(SEMI ri-parameter)
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ri-parameter = "seq" EQUAL 1*DIGIT
/ "rtptime" EQUAL 1*DIGIT
Retry-After = "Retry-After" HCOLON delta-seconds
[ comment ] *( SEMI retry-param )
retry-param = ("duration" EQUAL delta-seconds)
/ generic-param
Scale = "Scale" HCOLON ["-"] 1*DIGIT [ "." *DIGIT ] CRLF
Speed = "Speed" HCOLON 1*DIGIT [ "." *DIGIT ] CRLF
Server = "Server" HCOLON ( product / comment )
*(LWS (product / comment)) CRLF
product = token [SLASH product-version]
product-version = token
comment = LPAREN *( ctext / quoted-pair) RPAREN
Session = "Session" HCOLON session-id
[ SEMI "timeout" EQUAL delta-seconds ] CRLF
Supported = "Supported" HCOLON [feature-tag-list] CRLF
Timestamp = "Timestamp" HCOLON timestamp-value LWS [delay]
timestamp-value = *DIGIT [ "." *DIGIT ]
delay = *DIGIT [ "." *DIGIT ]
Transport = "Transport" HCOLON transport-spec
*(COMMA transport-spec) CRLF
transport-spec = transport-id *tr-parameter
transport-id = trans-id-rtp / other-trans
trans-id-rtp = "RTP/" profile ["/" lower-transport]
; no LWS is allowed inside transport-id
other-trans = token *("/" token)
profile = "AVP" / "SAVP" / "AVPF" / token
lower-transport = "TCP" / "UDP" / token
tr-parameter = SEMI ( "unicast" / "multicast" )
/ SEMI "interleaved" EQUAL channel [ "-" channel ]
/ SEMI "append"
/ SEMI "ttl" EQUAL ttl
/ SEMI "layers" EQUAL 1*DIGIT
/ SEMI "ssrc" EQUAL ssrc *(SLASH ssrc)
/ SEMI "client_ssrc" EQUAL ssrc
/ SEMI "mode" EQUAL mode-spec
/ SEMI "dest_addr" EQUAL addr-list
/ SEMI "src_addr" EQUAL addr-list
/ SEMI trn-param-ext
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trn-param-ext = par-name EQUAL trn-par-value
par-name = token
trn-par-value = *(rtsp-unreserved / DQUOTE *TEXT DQUOTE)
ttl = 1*3DIGIT ; 0 to 255
ssrc = 8HEX
channel = 1*3DIGIT
mode-spec = mode / ( DQUOTE mode *(COMMA mode) DQUOTE )
mode = "PLAY" / "RECORD" / token
addr-list = quoted-addr *(SLASH quoted-addr)
quoted-addr = DQUOTE (host-port / extension-addr) DQUOTE
host-port = host [":" port]
/ ":" port
extension-addr = 1*qdtext
Unsupported = "Unsupported" HCOLON feature-tag-list CRLF
User-Agent = "User-Agent" HCOLON ( product / comment )
0*(LWS (product / comment)) CRLF
Vary = "Vary" HCOLON ( "*" / field-name-list)
field-name-list = field-name *(COMMA field-name)
field-name = token
Via = "Via" HCOLON via-parm *(COMMA via-parm)
via-parm = sent-protocol LWS sent-by *( SEMI via-params )
via-params = via-ttl / via-maddr
/ via-received / via-branch
/ via-extension
via-ttl = "ttl" EQUAL ttl
via-maddr = "maddr" EQUAL host
via-received = "received" EQUAL (IPv4address / IPv6address)
via-branch = "branch" EQUAL token
via-extension = generic-param
sent-protocol = protocol-name SLASH protocol-version
SLASH transport-prot
protocol-name = "RTSP" / token
protocol-version = token
transport-prot = "UDP" / "TCP" / "TLS" / other-transport
other-transport = token
sent-by = host [ COLON port ]
WWW-Authenticate = "WWW-Authenticate" HCOLON challenge
19.3 SDP extension Syntax
This section defines in ABNF the SDP extensions defined for RTSP.
See section C for the definition of the extensions in text.
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control-attribute = "a=control:" *SP RTSP-URI
a-range-def = "a=range:" ranges-spec CRLF
a-etag-def = "a=etag:" etag-string CRLF
etag-string = 1*(%x01-09/%x0B-0C/%x0E-FF)
20 Security Considerations
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [H15]
apply. Specifically, please note the following:
Abuse of Server Log Information: RTSP and HTTP servers will
presumably have similar logging mechanisms, and thus should
be equally guarded in protecting the contents of those
logs, thus protecting the privacy of the users of the
servers. See [H15.1.1] for HTTP server recommendations
regarding server logs.
Transfer of Sensitive Information: There is no reason to believe
that information transferred via RTSP may be any less
sensitive than that normally transmitted via HTTP.
Therefore, all of the precautions regarding the protection
of data privacy and user privacy apply to implementors of
RTSP clients, servers, and proxies. See [H15.1.2] for
further details.
Attacks Based On File and Path Names: Though RTSP URIs are
opaque handles that do not necessarily have file system
semantics, it is anticipated that many implementations will
translate portions of the Request-URIs directly to file
system calls. In such cases, file systems SHOULD follow the
precautions outlined in [H15.5], such as checking for ".."
in path components.
Personal Information: RTSP clients are often privy to the same
information that HTTP clients are (user name, location,
etc.) and thus should be equally sensitive. See [H15.1]
for further recommendations.
Privacy Issues Connected to Accept Headers: Since may of the
same "Accept" headers exist in RTSP as in HTTP, the same
caveats outlined in [H15.1.4] with regards to their use
should be followed.
DNS Spoofing: Presumably, given the longer connection times
typically associated to RTSP sessions relative to HTTP
sessions, RTSP client DNS optimizations should be less
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prevalent. Nonetheless, the recommendations provided in
[H15.3] are still relevant to any implementation which
attempts to rely on a DNS-to-IP mapping to hold beyond a
single use of the mapping.
Location Headers and Spoofing: If a single server supports
multiple organizations that do not trust each another, then
it needs to check the values of Location and Content-
Location header fields in responses that are generated
under control of said organizations to make sure that they
do not attempt to invalidate resources over which they have
no authority. ([H15.4])
In addition to the recommendations in the current HTTP specification
(RFC 2616 [3], as of this writing) and also of the previous RFC2068
[18], future HTTP specifications may provide additional guidance on
security issues.
The following are added considerations for RTSP implementations.
Concentrated denial-of-service attack: The protocol offers the
opportunity for a remote-controlled denial-of-service
attack. See Section .
Session hijacking: Since there is no or little relation between
a transport layer connection and an RTSP session, it is
possible for a malicious client to issue requests with
random session identifiers which would affect unsuspecting
clients. The server SHOULD use a large, random and non-
sequential session identifier to minimize the possibility
of this kind of attack. For real session security, client
authentication needs to be performed.
Authentication: Servers SHOULD implement both basic and digest
[7] authentication. In environments requiring tighter
security for the control messages, the transport layer
mechanism TLS (RFC 2246 [6]) SHOULD be used.
Stream issues: RTSP only provides for stream control. Stream
delivery issues are not covered in this section, nor in the
rest of this draft. RTSP implementations will most likely
rely on other protocols such as RTP, IP multicast, RSVP and
IGMP, and should address security considerations brought up
in those and other applicable specifications.
Persistently suspicious behavior: RTSP servers SHOULD return
error code 403 (Forbidden) upon receiving a single instance
of behavior which is deemed a security risk. RTSP servers
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SHOULD also be aware of attempts to probe the server for
weaknesses and entry points and MAY arbitrarily disconnect
and ignore further requests clients which are deemed to be
in violation of local security policy.
20.1 Remote denial of Service Attack
The attacker may initiate traffic flows to one or more IP addresses
by specifying them as the destination in SETUP requests. While the
attacker's IP address may be known in this case, this is not always
useful in prevention of more attacks or ascertaining the attackers
identity. Thus, an RTSP server MUST only allow client-specified
destinations for RTSP-initiated traffic flows if the server has
ensured that the specified destination address accepts receiving
media through different security mechanisms. Security mechanism that
are acceptable in an increased generality are; verification of the
client's identity, either against a database of known users using
RTSP authentication mechanisms (preferably digest authentication or
stronger); a list of addresses that accept to be media destinations,
especially considering user identity; and media path based
verification.
The server SHOULD NOT allow the destination field to be set unless a
mechanism exists in the system to authorize the request originator to
direct streams to the recipient. It is preferred that this
authorization be performed by the recipient itself and the
credentials passed along to the server. However, in certain cases,
such as when recipient address is a multicast group, or when the
recipient is unable to communicate with the server in an out-of-band
manner, this may not be possible. In these cases server may chose
another method such as a server-resident authorization list to ensure
that the request originator has the proper credentials to request
stream delivery to the recipient.
21 IANA Considerations
This section set up a number of registers for RTSP 1.1 that should be
maintained by IANA. For each registry there is a description on what
it is required to contain, what specification is needed when adding a
entry with IANA, and finally the entries that this document needs to
register. See also the section 1.6 "Extending RTSP". There is also an
IANA registration of two SDP attributes.
The sections describing how to register an item uses some of the
requirements level described in RFC 2434 [19], namely "First Come,
First Served", "Specification Required", and "Standards Action".
A registration request to IANA MUST contain the following
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information:
o A name of the item to register according to the rules
specified by the intended registry.
o Indication of who has change control over the feature (for
example, IETF, ISO, ITU-T, other international standardization
bodies, a consortium, a particular company or group of
companies, or an individual);
o A reference to a further description, if available, for
example (in order of preference) an RFC, a published standard,
a published paper, a patent filing, a technical report,
documented source code or a computer manual;
o For proprietary features, contact information (postal and
email address);
21.1 Feature-tags
21.1.1 Description
When a client and server try to determine what part and functionality
of the RTSP specification and any future extensions that its counter
part implements there is need for a namespace. This registry
contains named entries representing certain functionality.
The usage of feature-tags is explained in section 10 and 11.1.
21.1.2 Registering New Feature-tags with IANA
The registering of feature-tags is done on a first come, first served
basis.
The name of the feature MUST follow these rules: The name may be of
any length, but SHOULD be no more than twenty characters long. The
name MUST NOT contain any spaces, or control characters. The
registration SHALL indicate if the feature tag applies to clients,
servers, or proxies only or any combinations of these. Any
proprietary feature SHALL have as the first part of the name a vendor
tag, which identifies the organization.
21.1.3 Registered entries
The following feature-tags are in this specification defined and
hereby registered. The change control belongs to the Authors and the
IETF MMUSIC WG.
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play.basic: The minimal implementation for playback operations
according to section D. Applies for both clients, servers
and proxies.
play.scale: Support of scale operations for media playback.
Applies only for servers.
play.speed: Support of the speed functionality for playback.
Applies only for servers
21.2 RTSP Methods
21.2.1 Description
What a method is, is described in section 11. Extending the protocol
with new methods allow for totally new functionality.
21.2.2 Registering New Methods with IANA
A new method MUST be registered through an IETF standard track
document. The reason is that new methods may radically change the
protocols behavior and purpose.
A specification for a new RTSP method MUST consist of the following
items:
o A method name which follows the ABNF rules for methods.
o A clear specification on what action and response a request
with the method will result in. Which directions the method is
used, C -> S or S -> C or both. How the use of headers, if
any, modifies the behavior and effect of the method.
o A list or table specifying which of the registered headers
that are allowed to use with the method in request or/and
response.
o Describe how the method relates to network proxies.
21.2.3 Registered Entries
This specification, RFCXXXX, registers 9 methods: DESCRIBE,
GET_PARAMETER, OPTIONS, PAUSE, PLAY, REDIRECT, SETUP, SET_PARAMETER,
and TEARDOWN.
21.3 RTSP Status Codes
21.3.1 Description
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A status code is the three digit numbers used to convey information
in RTSP response messages, see 7. The number space is limited and
care should be taken not to fill the space.
21.3.2 Registering New Status Codes with IANA
A new status code can only be registered by an IETF standards track
document. A specification for a new status code MUST specify the
following:
o The requested number.
o A description what the status code means and the expected
behavior of the sender and receiver of the code.
21.3.3 Registered Entries
RFCXXX, registers the numbered status code defined in the ABNF entry
"Status-Code" except "extension-code" in section 19.2.2.
21.4 RTSP Headers
21.4.1 Description
By specifying new headers a method(s) can be enhanced in many
different ways. An unknown header will be ignored by the receiving
entity. If the new header is vital for a certain functionality, a
feature-tag for the functionality can be created and demanded to be
used by the counter-part with the inclusion of a Require header
carrying the feature-tag.
21.4.2 Registering New Headers with IANA
A public available specification is required to register a header.
The specification SHOULD be a standards document, preferable an IETF
RFC.
The specification MUST contain the following information:
o The name of the header.
o An ABNF specification of the header syntax.
o A list or table specifying when the header may be used,
encompassing all methods, their request or response, the
direction (C -> S or S -> C).
o How the header is to be handled by proxies.
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o A description of the purpose of the header.
21.4.3 Registered entries
All headers specified in section 14 in RFCXXXX are to be registered.
Furthermore the following RTSP headers defined in other
specifications are registered:
o x-wap-profile defined in [36].
o x-wap-profile-diff defined in [36].
o x-wap-profile-warning defined in [36].
o x-predecbufsize defined in [36].
o x-initpredecbufperiod defined in [36].
o x-initpostdecbufperiod defined in [36].
o 3gpp-videopostdecbufsize defined in [36].
o 3GPP-Link-Char defined in [36].
o 3GPP-Adaptation defined in [36].
o 3GPP-QoE-Metrics defined in [36].
o 3GPP-QoE-Feedback defined in [36].
The use of "X-" is NOT RECOMMENDED but the above headers in the
register list was defined prior to the clarification.
21.5 Transport Header registries
The transport header contains a number of parameters which have
possibilities for future extensions. Therefore registries for these
needs to be defined.
21.5.1 Transport Protocols
A registry for the parameter transport-protocol SHALL be defined with
the following rules:
o Registering require an public available standards
specification.
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o A contact person or organization with address and email.
o A value definition that are following the ABNF token
definition.
o A describing text that explains how the registered value are
used in RTSP.
This specification registers 1 value:
o Use of the RTP [16] protocol for media transport. The usage
is explained in RFC XXXX, appendix B.1.
21.5.2 Profile
A registry for the parameter profile SHALL be defined with the
following rules:
o Registering requires public available standards specification.
o A contact person or organization with address and email.
o A value definition that are following the BNF token
definition.
o A definition of which Transport protocol(s) that this profile
is valid for.
o A describing text that explains how the registered value are
used in RTSP.
This specification registers 3 value:
o The "RTP profile for audio and video conferences with minimal
control" [2] MUST only be used when the transport
specification's transport-protocol is "RTP".
o The "Extended RTP Profile for RTCP-based Feedback (RTP/AVPF)"
[20] MUST only be used when the transport specification's
transport-protocol is "RTP"
o The "The Secure Real-time Transport Protocol (SRTP)" [21] MUST
only be used when the transport specification's transport-
protocol is "RTP".
21.5.3 Lower Transport
A registry for the parameter lower-transport SHALL be defined with
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the following rules:
o Registering requires public available standards specification.
o A contact person or organization with address and email.
o A value definition that are following the BNF token
definition.
o A text describing how the registered value are used in RTSP.
This specification registers 2 values:
UDP: Indicates the use of the "User datagram protocol" [8] for
media transport.
TCP: Indicates the use Transmission control protocol [9] for
media transport.
21.5.4 Transport modes
A registry for the transport parameter mode SHALL be defined with the
following rules:
o Registering requires an IETF standard tracks document.
o A contact person or organization with address and email.
o A value definition that are following the ABNF token
definition.
o A describing text that explains how the registered value are
used in RTSP.
This specification registers 1 values:
PLAY: See RFC XXXX.
21.5.5 Transport Parameters
A registry for parameters that may be included in the Transport
header SHALL be defined with the following rules:
o Registering required a Open Standards document
o A value definition that are following the ABNF token
definition.
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o A describing text that explains how the registered value are
used in RTSP.
This specification registers all the transport parameters defined in
Section 14.45.
21.6 Cache Directive Extensions
There exist a number of cache directives which can be sent in the
Cache-Control header. A registry for this cache directives SHALL be
defined with the following rules:
o Registering requires an IETF standard tracks document.
o A registration is required to contain a contact person.
o Name of the directive and a definition of the value, if any.
o Specification if it is an request or response directive.
o A describing text that explains how the cache directive is
used for RTSP controlled media streams.
This specification registers the following values:
no-cache:
public:
private:
no-transform:
only-if-cached:
max-stale:
min-fresh:
must-revalidate:
proxy-revalidate:
max-age:
21.7 Accept-Credentials policies
In section 18.3.1 three policies for how to handle certificates.
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Further policies may be defined and SHALL be registered with IANA
using the following rules:
o Registering requires an IETF standard tracks document.
o A registration is required name a contact person.
o Name of the policy.
o A describing text that explains how the policy works for
handling the certificates.
This specification registers the following values:
Any
Proxy
User
21.8 URI Schemes
This specification defines two URI schemes ("rtsp" and "rtsps") and
reserves a third one ("rtspu").
This will need to be done in accordance with RFC 2717.
21.9 SDP attributes
This specification defines two SDP [1] attributes that it is
requested that IANA register.
SDP Attribute ("att-field"):
Attribute name: range
Long form: Media Range Attribute
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: See ABNF definition.
Attribute name: control
Long form: RTSP control URI
Type of name: att-field
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Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: Absolute or Relative URIs.
Attribute name: etag
Long form: Entity Tag
Type of name: att-field
Type of attribute: Media and session level
Subject to charset: No
Purpose: RFC XXXX
Reference: RFC XXXX
Values: See ABNF definition
A RTSP Protocol State Machine
The RTSP session state machine describes the behavior of the protocol
from RTSP session initialization through RTSP session termination.
The State machine is defined on a per session basis which is uniquely
identified by the RTSP session identifier. The session may contain
one or more media streams depending on state. If a single media
stream is part of the session it is in non-aggregated control. If two
or more is part of the session it is in aggregated control.
The below state machine is a normative description of the protocols
behavior. However, in case of ambiguity with the earlier parts of
this specification, the description in the earlier parts SHALL take
precedence.
A.1 States
The state machine contains three states, described below. For each
state there exist a table which shows which requests and events that
is allowed and if they will result in a state change.
Init: Initial state no session exist.
Ready: Session is ready to start playing.
Play: Session is playing, i.e. sending media stream data in the
direction S -> C.
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A.2 State variables
This representation of the state machine needs more than its state to
work. A small number of variables are also needed and is explained
below.
NRM: The number of media streams part of this session.
RP: Resume point, the point in the presentation time line at
which a request to continue will resume from. A time format
for the variable is not mandated.
A.3 Abbreviations
To make the state tables more compact a number of abbreviations are
used, which are explained below.
IFI: IF Implemented.
md: Media
PP: Pause Point, the point in the presentation time line at
which the presentation was paused.
Prs: Presentation, the complete multimedia presentation.
RedP: Redirect Point, the point in the presentation time line at
which a REDIRECT was specified to occur.
SES: Session.
A.4 State Tables
This section contains a table for each state. The table contains all
the requests and events that this state is allowed to act on. The
events which is method names are, unless noted, requests with the
given method in the direction client to server (C -> S). In some
cases there exist one or more requisite. The response column tells
what type of response actions should be performed. Possible actions
that is requested for an event includes: response codes, e.g. 200,
headers that MUST be included in the response, setting of state
variables, or setting of other session related parameters. The new
state column tells which state the state machine changes to.
The response to valid request meeting the requisites is normally a
2xx (SUCCESS) unless other noted in the response column. The
exceptions needs to be given a response according to the response
column. If the request does not meet the requisite, is erroneous or
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some other type of error occur the appropriate response code MUST be
sent. If the response code is a 4xx the session state is unchanged. A
response code of 3rr will result in that the session is ended and its
state is changed to Init. A response code of 304 results in no state
change. However there exist restrictions to when a 3xx response may
be used. A 5xx response SHALL not result in any change of the session
state, except if the error is not possible to recover from. A
unrecoverable error SHALL result the ending of the session. As it in
the general case can't be determined if it was a unrecoverable error
or not the client will be required to test. In the case that the next
request after a 5xx is responded with 454 (Session Not Found) the
client knows that the session has ended.
The server will timeout the session after the period of time
specified in the SETUP response, if no activity from the client is
detected. Therefore there exist a timeout event for all states
except Init.
In the case that NRM=1 the presentation URI is equal to the media
URI. For NRM>1 the presentation URI MUST be other than any of the
medias that are part of the session. This applies to all states.
Event Prerequisite Response
______________________________________________________________
DESCRIBE Needs REDIRECT 3rr, Redirect
DESCRIBE 200, Session description
OPTIONS Session ID 200, Reset session timeout timer
OPTIONS 200
SET_PARAMETER Valid parameter 200, change value of parameter
GET_PARAMETER Valid parameter 200, return value of parameter
Table 13: None state-machine changing events
The methods in Table 13 do not have any effect on the state machine
or the state variables. However some methods do change other session
related parameters, for example SET_PARAMETER which will set the
parameter(s) specified in its body. Also all of these methods that
allows Session header will also update the keep-alive timer for the
session.
The initial state of the state machine, see Table 14 can only be left
by processing a correct SETUP request. As seen in the table the two
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Action Requisite New State Response
_____________________________________________________________
SETUP Ready NRM=1, RP=0.0
SETUP Needs Redirect Init 3rr Redirect
S -> C:REDIRECT No Session hdr Init Terminate all SES
Table 14: State: Init
state variables are also set by a correct request. This table also
shows that a correct SETUP can in some cases be redirected to another
URI and/or server by a 3rr response.
Action Requisite New State Response
_____________________________________________________________________
SETUP New URI Ready NRM+=1
SETUP Setten up URI Ready Change transport param
TEARDOWN Prs URI,NRM>1 Init No session hdr, NRM=0
TEARDOWN md URI,NRM=1 Init No Session hdr, NRM=0
TEARDOWN md URI,NRM>1 Ready Session hdr, NRM-=1
PLAY Prs URI, No range Play Play from RP
PLAY Prs URI, Range Play According to range
PAUSE Prs URI Ready Return PP
S -> C:REDIRECT Range hdr Ready Set RedP
S -> C:REDIRECT no range hdr Init Session is removed
Timeout Init
RedP reached Init TEARDOWN of session
Table 15: State: Ready
In the Ready state, see Table 15, some of the actions are depending
on the number of media streams (NRM) in the session, i.e. aggregated
or non-aggregated control. A setup request in the ready state can
either add one more media stream to the session or if the media
stream (same URI) already is part of the session change the transport
parameters. TEARDOWN is depending on both the Request-URI and the
number of media stream within the session. If the Request-URI is the
presentations URI the whole session is torn down. If a media URI is
used in the TEARDOWN request and more than one media exist in the
session, the session will remain and a session header MUST be
returned in the response. If only a single media stream remains in
the session when performing a TEARDOWN with a media URI the session
is removed. The number of media streams remaining after tearing down
a media stream determines the new state.
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Action Requisite New State Response
_____________________________________________________________________
PAUSE PrsURI,No range Ready Set RP to present point
PAUSE PrsURI,Range>now Play Set RP & PP to given p.
PAUSE PrsURI,Range<now Ready Set RP to Range Hdr.
PP reached Ready RP = PP
End of media All media Play Set RP = End of media
End of range Play Set RP = End of range
PLAY Prs URI, No range Play Play from present point
PLAY Prs URI, Range Play According to range
SETUP New URI Play 455
SETUP Setuped URI Play 455
SETUP Setuped URI, IFI Play Change transport param.
TEARDOWN Prs URI,NRM>1 Init No session hdr
TEARDOWN md URI,NRM=1 Init No Session hdr, NRM=0
TEARDOWN md URI Play 455
S -> C:REDIRECT Range hdr Play Set RedP
S -> C:REDIRECT no range hdr Init Session is removed
RedP reached Init TEARDOWN of session
Timeout Init Stop Media playout
Table 16: State: Play
The Play state table, see Table 16, is the largest. The table
contains an number of requests that has presentation URI as a
prerequisite on the Request-URI, this is due to the exclusion of
non-aggregated stream control in sessions with more than one media
stream.
To avoid inconsistencies between the client and server, automatic
state transitions are avoided. This can be seen at for example "End
of media" event when all media has finished playing, the session
still remain in Play state. An explicit PAUSE request MUST be sent to
change the state to Ready. It may appear that there exist two
automatic transitions in "RedP reached" and "PP reached", however
they are requested and acknowledge before they take place. The time
at which the transition will happen is known by looking at the range
header. If the client sends request close in time to these
transitions it needs to be prepared for getting error message as the
state may or may not have changed.
B Media Transport Alternatives
This section defines how certain combinations of protocols, profiles
and lower transports are used. This includes the usage of the
Transport header's source and destination address parameters
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"src_addr" and "dest_addr".
B.1 RTP
This section defines the interaction of RTSP with respect to the RTP
protocol [16]. It also defines any necessary media transport
signalling with regards to RTP.
The available RTP profiles and lower layer transports are described
below along with rules on signalling the available combinations.
B.1.1 AVP
The usage of the "RTP Profile for Audio and Video Conferences with
Minimal Control" [2] when using RTP for media transport over
different lower layer transport protocols is defined below in regards
to RTSP.
One such case is defined within this document, the use of embedded
(interleaved) binary data as defined in section 12. The usage of
this method is indicated by include the "interleaved" parameter.
When using embedded binary data the "src_addr" and "dest_addr" SHALL
NOT be used. This addressing and multiplexing is used as defined with
use of channel numbers and the interleaved parameter.
B.1.2 AVP/UDP
This part describes sending of RTP [16] over lower transport layer
UDP [8] according to the profile "RTP Profile for Audio and Video
Conferences with Minimal Control" defined in RFC 3551 [2]. This
profiles requires one or two uni- or bi-directional UDP flows per
media stream. The first UDP flow is for RTP and the second is for
RTCP. Embedding of RTP data with the RTSP messages, in accordance
with section 12, SHOULD NOT be performed when RTSP messages are
transported over unreliable transport protocols, like UDP [8].
The RTP/UDP and RTCP/UDP flows can be established using the Transport
header's "src_addr", and "dest_addr" parameters.
In RTSP PLAY mode, the transmission of RTP packets from client to
server is unspecified. The behavior in regards to such RTP packets
MAY be defined in future.
The "src_addr" and "dest_addr" parameters are used in the following
way for media playback, i.e. Mode=PLAY:
o The "src_addr" and "dest_addr" parameters MUST contain either
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1 or 2 address specifications.
o Each address specification for RTP/AVP/UDP or RTP/AVP/TCP MUST
contain either:
- both an address and a port number, or
- a port number without an address
o The first address and port pair given in either of the
parameters applies to the RTP stream. The second address and
port pair if present applies to the RTCP stream.
o The RTP/UDP packets from the server to the client SHALL be
sent to the address and port given by first address and port
pair of the "dest_addr" parameter.
o The RTCP/UDP packets from the server to the client SHALL be
sent to the address and port given by the second address and
port pair of the "dest_addr" parameter. If no second pair is
given RTCP SHALL NOT be sent.
o The RTCP/UDP packets from the client to the server SHALL be
sent to the address and port given by the second address and
port pair of the "src_addr" parameter. If no second pair is
given RTCP SHALL NOT be sent.
o The RTP/UDP packets from the client to the server SHALL be
sent to the address and port given by the first address and
port pair of the "src_addr" parameter.
o RTP and RTCP Packets SHOULD be sent from the corresponding
receiver port, i.e. RTCP packets from server should be sent
from the "src_addr" parameters second address port pair.
B.1.3 AVP/TCP
Note that this combination is not yet defined using sperate TCP
connections. However the use of embedded (interleaved) binary data
transported on the RTSP connection is possible as specified in
section 12. When using this declared combination of interleaved
binary data the RTSP messages MUST be transported over TCP.
A possible future for this profile would be to define the
use of a combination of the two drafts "Connection-Oriented
Media Transport in SDP" [37] and "Framing RTP and RTCP
Packets over Connection-Oriented Transport" [38]. However
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as this work is not finished, this functionality is
unspecified.
B.1.4 AVPF
The RTP profile "Extended RTP Profile for RTCP-based Feedback
(RTP/AVPF)" [20] MAY be used as RTP profiles in session using RTP.
All that is defined for AVP SHALL also apply for AVPF.
The usage of AVPF is indicated by the media initialization protocol
used. In the case of SDP it is indicated by media lines (m=)
containing the profile RTP/AVPF. That SDP MAY also contain further
AVPF related SDP attributes configuring the AVPF session regarding
reporting interval and feedback messages that shall be used that
SHALL be followed.
B.1.5 SAVP
The RTP profile "The Secure Real-time Transport Protocol (SRTP)" [21]
is an RTP profile (SAVP) that MAY be used in RTSP sessions using RTP.
All that is defined for AVP SHALL also apply for AVPF.
The usage of SRTP requires that a security association is
established. The protocol used are outside of the scope of RTSP,
however a method must exist to enable the usage of the RTP profile
SAVP.
B.1.6 Handling NPT Jumps in the RTP Media Layer
RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP
media layer[16]. Such control allows jumps to be created in NPT
timeline of the RTSP session. For example, jumps in NPT can be caused
by multiple ranges in the range specifier of a PLAY request or
through a "seek" opertaion on an RTSP session which involves a PLAY,
PAUSE, PLAY scenario where a new NPT is set for the session. The
media layer rendering the RTP stream should not be affected by jumps
in NPT. Thus, both RTP sequence numbers and RTP timestamps MUST be
continuous and monotonic across jumps of NPT.
We cannot assume that the RTSP client can communicate with
the RTP media agent, as the two may be independent
processes. If the RTP timestamp shows the same gap as the
NPT, the media agent will assume that there is a pause in
the presentation. If the jump in NPT is large enough, the
RTP timestamp may roll over and the media agent may believe
later packets to be duplicates of packets just played out.
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As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms and an initial sequence number and timestamp of
zero.
C->S: PLAY rtsp://xyz/fizzle RTSP/1.1
CSeq: 4
Session: abcdefg
Range: npt=10-15
S->C: RTSP/1.1 200 OK
CSeq: 4
Session: abcdefg
Range: npt=10-15
RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
. . .
S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
Immediately after the end of the play range, the client follows up
with a request to PLAY from a new NPT.
C->S: PLAY rtsp://xyz/fizzle RTSP/1.1
CSeq: 5
Session: abcdefg
Range: npt=18-20;
S->C: RTSP/1.1 200 OK
CSeq: 5
Session: abcdefg
Range: npt=18-20
RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
ssrc=0D12F123:seq=50;rtptime=40100
The ensuing RTP data stream is depicted below:
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S->C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
S->C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
. . .
S->C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s
In this example, first, NPT 10 through 15 is played, then the client
request the server to skip ahead and play NPT 18 through 20. The
first segment is presented as RTP packets with sequence numbers 0
through 49 and timestamp 0 through 39,200. The second segment
consists of RTP packets with sequence number 50 through 69, with
timestamps 40,100 through 55,200. While there is a gap in the NPT,
there is no gap in the sequence number space of the RTP data stream.
The RTP timestamp gap is present in the above example due to the time
it takes to perform the second play request, in this case 12.5 ms
(100/8000). To avoid this gap in playback due to the time it takes to
perform RTSP requests, a PLAY request with multiple ranges needs to
be specified. That would result in the following example:
C->S: PLAY rtsp://xyz/fizzle RTSP/1.1
CSeq: 4
Session: abcdefg
Range: npt=10-15;npt=18-20
S->C: RTSP/1.1 200 OK
CSeq: 4
Session: abcdefg
Range: npt=10-15
RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
. . .
S -> C: RTP packet - seq = 49, rtptime = 39200, NPT time = 14.9s
S -> C: RTP packet - seq = 50, rtptime = 40100, NPT time = 18s
S -> C: RTP packet - seq = 51, rtptime = 40900, NPT time = 18.1s
. . .
S -> C: RTP packet - seq = 69, rtptime = 55300, NPT time = 19.9s
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B.1.7 Handling RTP Timestamps after PAUSE
During a PAUSE / PLAY interaction in an RTSP session, the duration of
time for which the RTP transmission was halted MUST be reflected in
the RTP timestamp of each RTP stream. The duration can be calculated
for each RTP stream as the time elapsed from when the last RTP packet
was sent before the PAUSE request was received and when the first RTP
packet was sent after the subsequent PLAY request was received. The
duration includes all latency incurred and processing time required
to complete the request.
The RTP RFC [16] states that: The RTP timestamp for each
unit[packet] would be related to the wallclock time at
which the unit becomes current on the virtual presentation
timeline.
In order to satisfy the requirements of [16], the RTP timestamp space
needs to increase continuously with real time. While this is not
optimal for stored media, it is required for RTP and RTCP to function
as intended. Using a continuous RTP timestamp space allows the same
timestamp model for both stored and live media and allows better
opportunity to integrate both types of media under a single control.
As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms and an initial sequence number and timestamp of
zero.
C->S: PLAY rtsp://xyz/fizzle RTSP/1.1
CSeq: 4
Session: abcdefg
Range: npt=10-15;
S->C: RTSP/1.1 200 OK
CSeq: 4
Session: abcdefg
Range: npt=10-15
RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
ssrc=0D12F123:seq=0;rtptime=0
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 0, rtptime = 0, NPT time = 10s
S -> C: RTP packet - seq = 1, rtptime = 800, NPT time = 10.1s
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S -> C: RTP packet - seq = 2, rtptime = 1600, NPT time = 10.2s
S -> C: RTP packet - seq = 3, rtptime = 2400, NPT time = 10.3s
The client then sends a PAUSE request:
C->S: PAUSE rtsp://xyz/fizzle RTSP/1.1
CSeq: 5
Session: abdcdefg
S->C: RTSP/1.1 200 OK
CSeq: 5
Session: abcdefg
Range: npt=10.4-15
20 seconds elapse and then the client sends a PLAY request. In
addition the server requires 15 ms to process the request:
C->S: PLAY rtsp://xyz/fizzle RTSP/1.1
CSeq: 6
Session: abcdefg
S->C: RTSP/1.1 200 OK
CSeq: 6
Session: abcdefg
Range: npt=10.4-15
RTP-Info: url="rtsp://xyz/fizzle/audiotrack"
ssrc=0D12F123:seq=4;rtptime=164400
The ensuing RTP data stream is depicted below:
S -> C: RTP packet - seq = 4, rtptime = 164400, NPT time = 10.4s
S -> C: RTP packet - seq = 5, rtptime = 165200, NPT time = 10.5s
S -> C: RTP packet - seq = 6, rtptime = 166000, NPT time = 10.6s
First, NPT 10 through 10.3 is played, then a PAUSE is received by the
server. After 20 seconds a PLAY is received by the server which take
15ms to process. The duration of time for which the session was
paused is reflected in the RTP timestamp of the RTP packets sent
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after this PLAY request.
A client can use the RTSP range header and RTP-Info header to map NPT
time of a presentation with the RTP timestamp.
Note: In RFC 2326 [24], this matter was not clearly defined and was
misunderstood commonly. However for RTSP 1.1 it is expected that this
will be handled correctly and not exception handling will be
required.
B.1.8 RTSP / RTP Integration
For certain datatypes, tight integration between the RTSP layer and
the RTP layer will be necessary. This by no means precludes the above
restrictions. Combined RTSP/RTP media clients should use the RTP-Info
field to determine whether incoming RTP packets were sent before or
after a seek or before or after a PAUSE.
B.1.9 Scaling with RTP
For scaling (see Section 14.39), RTP timestamps should correspond to
the playback timing. For example, when playing video recorded at 30
frames/second at a scale of two and speed (Section 14.40) of one, the
server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame.
Note: The above scaling puts requirements on the media
codec or a media stream to support it. For example motion
JPEG or other non-predictive video coding can easier handle
the above example.
B.1.10 Maintaining NPT synchronization with RTP timestamps
The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning.
The sequence parameter of the RTP-Info (Section 14.38) header
provides the first sequence number of the next segment.
B.1.11 Continuous Audio
For continuous audio, the server SHOULD set the RTP marker bit at the
beginning of serving a new PLAY request or at jumps in timeline. This
allows the client to perform playout delay adaptation.
B.1.12 Multiple Sources in an RTP Session
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Note that more than one SSRC MAY be sent in the media stream. If it
happens all sources are expected to be rendered simultaneously.
B.1.13 Usage of SSRCs and the RTCP BYE Message During an RTSP Session
The RTCP BYE message indicates the end of use of a given SSRC. If all
sources leave an RTP session, it can, in most cases, be assumed to
have ended. Therefore, a client or server SHALL NOT send a RTCP BYE
message until it has finished using a SSRC. A server SHOULD keep
using a SSRC until the RTP session is terminated. Prolonging the use
of a SSRC allows the established synchronization context associated
with that SSRC to be used to synchronize subsequent PLAY requests
even if the PLAY response is late.
An SSRC collision with the SSRC that transmits media does also have
consequences, as it will force the media sender to change its SSRC in
accordance with the RTP specification [16]. This will result in a
loss of synchronization context, and require any receiver to wait for
RTCP sender reports for all media requiring synchronization before
being able to play out synchronized. Due to these reasons a client
joining a session should take care to not select the same SSRC as the
server. Any SSRC signalled in the Transport header SHOULD be avoided.
A client detecting a collision prior to sending any RTP or RTCP
messages can also select a new SSRC.
B.2 Future Additions
It is the intention that any future protocol or profile regarding
both for media delivery and lower transport should be easy to add to
RTSP. This section provides the necessary steps that needs to be
meet.
The following things needs to be considered when adding a new
protocol of profile for use with RTSP:
o The protocol or profile needs to define a name tag
representing it. This tag is required to be a ABNF "token" to
be possible to use in the Transport header specification.
o The useful combinations of protocol/profile/lower-layer needs
to be defined and for each combination declare the necessary
parameters to use in the Transport header.
o For new media protocols the interaction with RTSP needs to be
addressed. One important factor will be the media
synchronization.
See the IANA section (21) for information how to register new
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attributes.
C Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, RFC 2327 [1]) may be used to
describe streams or presentations in RTSP. This description is
typically returned in reply to a DESCRIBE request on an URI from a
server to a client, or received via HTTP from a server to a client.
This appendix describes how an SDP file determines the operation of
an RTSP session. SDP as is provides no mechanism by which a client
can distinguish, without human guidance, between several media
streams to be rendered simultaneously and a set of alternatives
(e.g., two audio streams spoken in different languages). However the
SDP extension "Grouping of Media Lines in the Session Description
Protocol (SDP)" [39] may provide such functionality depending on
need. Also future grouping semantics may in the future be developed.
C.1 Definitions
The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in
SDP (RFC 2327 [1]):
C.1.1 Control URI
The "a=control:" attribute is used to convey the control URI. This
attribute is used both for the session and media descriptions. If
used for individual media, it indicates the URI to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URI for aggregate control
(presentation URI). The session level URI SHALL be different from any
media level URI. The presence of a session level control attribute
SHALL be interpreted as support for aggregated control. The control
attribute SHALL be present on media level unless the presentation
only contains a single media stream, in which case the attribute MAY
only be present on the session level.
ABNF for the attribute is defined in section 19.3.
Example:
a=control:rtsp://example.com/foo
This attribute MAY contain either relative or absolute URIs,
following the rules and conventions set out in RFC 3986 [10].
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Implementations SHALL look for a base URI in the following order:
1. the RTSP Content-Base field;
2. the RTSP Content-Location field;
3. the RTSP Request-URI.
If this attribute contains only an asterisk (*), then the URI SHALL
be treated as if it were an empty embedded URI, and thus inherit the
entire base URI.
The URI handling for SDPs from container files need special
consideration. For example lets assume that a container file has the
URI: "rtsp://example.com/container.mp4". Lets further assume this URI
is the base URI, and that there is a absolute media level URI:
"rtsp://example.com/container.mp4/trackID=2". A relative media level
URI that resolves in accordance with RFC 3986 [10] to the above given
media URI is: "container.mp4/trackID=2". It is usually not desirable
to need to include in or modify the SDP stored within the container
file with the server local name of the container file. To avoid this,
one can modify the base URI used to include a trailing slash, e.g.
"rtsp://example.com/container.mp4/". In this case the relative URI
for the media will only need to be: "trackID=2". However this will
also mean that using "*" in the SDP will result in control URI
including the trailing slash, i.e.
"rtsp://example.com/container.mp4/".
C.1.2 Media Streams
The "m=" field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate
synchronization. If the session is over multicast, the port number
indicated SHOULD be used for reception. The client MAY try to
override the destination port, through the Transport header. The
servers MAY allow this, the response will indicate if allowed or not.
If the session is unicast, the port number is the ones RECOMMENDED by
the server to the client, about which receiver ports to use; the
client MUST still include its receiver ports in its SETUP request.
The client MAY ignore this recommendation. If the server has no
preference, it SHOULD set the port number value to zero.
The "m=" lines contain information about what transport protocol,
profile, and possibly lower-layer is to be used for the media stream.
The combination of transport, profile and lower layer, like
RTP/AVP/UDP needs to be defined for how to be used with RTSP. The
currently defined combinations are defined in section B, further
combinations MAY be specified.
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Usage of grouping of media lines [39] to determine which media lines
should or should not be included in a RTSP session is unspecified.
Example:
m=audio 0 RTP/AVP 31
C.1.3 Payload Type(s)
The payload type(s) are specified in the "m=" field. In case the
payload type is a static payload type from RFC 3551 [2], no other
information may be required. In case it is a dynamic payload type,
the media attribute "rtpmap" is used to specify what the media is.
The "encoding name" within the "rtpmap" attribute may be one of those
specified in RFC 3551 (Sections 5 and 6), or an MIME type registered
with IANA, or an experimental encoding as specified in SDP (RFC 2327
[1]). Codec-specific parameters are not specified in this field, but
rather in the "fmtp" attribute described below.
C.1.4 Format-Specific Parameters
Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) that the attribute refers to. Note that some of the
format specific parameters may be specified outside of the fmtp
parameters, like for example the "ptime" attribute for most audio
encodings.
C.1.5 Range of Presentation
The "a=range" attribute defines the total time range of the stored
session or an individual media. Non-seekable live sessions can be
indicated, while the length of live sessions can be deduced from the
"t" and "r" SDP parameters.
The attribute is both a session and a media level attribute. For
presentations that contains media streams of the same durations, the
range attribute SHOULD only be used at session-level. In case of
different length the range attribute MUST be given at media level for
all media, and SHOULD NOT be given at session level. If the attribute
is present at both media level and session level the media level
values SHALL be used.
The unit is specified first, followed by the value range. The units
and their values are as defined in Section 3.4, 3.5 and 3.6 and MAY
be extended with further formats. Any open ended range (start-), i.e.
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without stop range, is of unspecified duration and SHALL be
considered as non-seekable content unless this property is
overridden. Multiple instances carrying different clock formats MAY
be included at either session or media level.
ABNF for the attribute is defined in section 19.3.
Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203
Non seekable stream of unknown duration:
a=range:npt=0-
C.1.6 Time of Availability
The "t=" field MUST contain suitable values for the start and stop
times for both aggregate and non-aggregate stream control. The
server SHOULD indicate a stop time value for which it guarantees the
description to be valid, and a start time that is equal to or before
the time at which the DESCRIBE request was received. It MAY also
indicate start and stop times of 0, meaning that the session is
always available.
For sessions that are of live type, i.e. specific start time, unknown
stop time, likely unseekable, the "t=" and "r=" field SHOULD be used
to indicate the start time of the event. The stop time SHOULD be
given so that the live event will with high probability have ended at
that time, while still not be unnecessary long into the future.
C.1.7 Connection Information
In SDP, the "c=" field contains the destination address for the media
stream. For a media destination address that is a IPv6 one, the SDP
extension defined in [22] needs to be used. For on-demand unicast
streams and some multicast streams, the destination address MAY be
specified by the client via the SETUP request, thus overriding any
specified address. To identify streams without a fixed destination
address, where the client is required to specify a destination
address, the "c=" field SHOULD be set to a null value. For addresses
of type "IP4", this value SHALL be "0.0.0.0", and for type "IP6",
this value SHALL be "0:0:0:0:0:0:0:0", i.e. the unspecified address
according to RFC 3513 [23].
C.1.8 Entity Tag
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The optional "a=etag" attribute identifies a version of the session
description. It is opaque to the client. SETUP requests may include
this identifier in the If-Match field (see section 14.25) to only
allow session establishment if this attribute value still corresponds
to that of the current description. The attribute value is opaque
and may contain any character allowed within SDP attribute values.
ABNF for the attribute is defined in section 19.3.
Example:
a=etag:158bb3e7c7fd62ce67f12b533f06b83a
One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would
put constraints on servers that need to support multiple
session description types other than SDP for the same piece
of media content.
C.2 Aggregate Control Not Available
If a presentation does not support aggregate control no session level
"a=control:" attribute is specified. For a SDP with multiple media
sections specified, each section will have its own control URI
specified via the "a=control:" attribute.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I came from a web page
e=adm@example.com
c=IN IP4 0.0.0.0
t=0 0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid
Note that the position of the control URI in the description implies
that the client establishes separate RTSP control sessions to the
servers audio.com and video.com
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It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media
client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.
C.3 Aggregate Control Available
In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both a media-level
"a=control:" attributes, which are used to specify the stream URIs,
and a session-level "a=control:" attribute which is used as the
Request-URI for aggregate control. If the media-level URI is
relative, it is resolved to absolute URIs according to Section C.1.1
above.
Example:
C->M: DESCRIBE rtsp://example.com/movie RTSP/1.1
CSeq: 1
M->C: RTSP/1.1 200 OK
CSeq: 1
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Base: rtsp://example.com/movie/
Content-Length: 228
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I contain
i=<more info>
e=adm@example.com
c=IN IP4 0.0.0.0
t=0 0
a=control:*
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
In this example, the client is required to establish a single RTSP
session to the server, and uses the URIs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to set up the video and audio
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streams, respectively. The URI rtsp://example.com/movie/ , which is
resolved from the "*", controls the whole presentation (movie).
A client is not required to issues SETUP requests for all streams
within an aggregate object. Servers should allow the client to ask
for only a subset of the streams.
C.4 RTSP external SDP delivery
There are some considerations that needs to be made when the session
description is delivered to client outside of RTSP, for example in
HTTP or email.
First of all the SDP needs to contain absolute URIs, relative will in
most cases not work as the delivery will not correctly forward the
base URI. And as SDP might be temporarily stored on file system
before being loaded into an RTSP capable client, thus if possible to
transport the base URI it still would need to be merged into the
file.
The writing of the SDP session availability information, i.e. "t="
and "r=", needs to be carefully considered. When the SDP is fetched
by the DESCRIBE method it is with very high probability that the it
is valid. However the same are much less certain for SDPs distributed
using other methods. Therefore the publisher of the SDP should take
care to follow the recommendations about availability in the SDP
specification [1].
D Minimal RTSP implementation
Note: This section is still under development!
This section defines the minimal implementation requirements for any
client and server. In addition the requirements for supporting the
"play.basic" feature tag is defined here.
D.1 Minimal Core Implementation
The minimal core implementation is what is required to negotiate the
usage of any other features. A minimal core implementation is not
supporting any other feature set will be useless as the minimal
implementation doesn't deliver any service. All feature sets SHALL
include the minimal core.
A minimal core implementation SHALL support the following
functionalities:
o Establishing a connection between RTSP agent using TCP.
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o Implement the reception and response to the OPTIONS method.
o Implement the handling of all headers mandatory or conditional
in regards to the usage of the OPTIONS method. See tables 9
and 10. This include at least the capability to ignore
unknown headers.
o Implement the headers related to capability negotiation and
exchange:
- Require
- Supported
- Proxy-Require
- Proxy-Supported
- Unsupported
D.2 The Basic Playback Feature Support
This section defines what is required to be supported for clients,
proxies and servers to be supporting the "play.basic" feature tag.
D.2.1 Client
A play.basic supporting client SHALL implement the following:
o The RTSP methods as required by Table 7.
o All the RTSP headers that are required required or conditional
in requests or responses to method required to be supported
according to Tables 9, 10, 11, and 12 and in addition the
following headers:
- Content-Base
- Content-Encoding and at least the Identity method.
- Content-Location
- Location
- Range
- RTP-Info
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o Handling of all Status code categories and in addition the
following specific status codes:
o Media delivery using RTP/AVP over UDP.
A play.basic client is RECOMMENDED to support the following:
o RTSP basic and digest authentication: The 401 response, the
WWW-Authenticate and Authorization headers, and both Basic and
Digest authentication methods as defined by [7].
o Expires header
o From header
o Secure Transport as specified by section D.3.
D.2.2 Server
To be written!
D.2.3 Proxy
A play.basic supporting proxy SHALL implement the following:
o Correct handling of the RTSP methods as required by Table 7.
o The handling of all RTSP headers that are required to be
handled by the server and clients supporting "play.basic" and
in addition the following headers:
- Cache-Control
- Expires
- Via
D.3 Secure Transport
Any Client, Proxy or Server supporting secure transport of RTSP
messages and usage of the "rtsps" URI scheme SHALL implement; The
Accept-Credentials and Connection-Credentials headers; TLS over TCP.
D.4 Old Implementation Text
The OLD Text follows from here on and is kept in this revision for
comparison reasons:
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D.5 Client
A client implementation MUST be able to do the following :
o Generate the following requests: SETUP, TEARDOWN, PLAY.
o Include the following headers in requests: CSeq, Connection,
Session, Transport.
o Parse and understand the following headers in responses:
CSeq, Connection, Session, Transport, Content-Language,
Content-Encoding, Content-Length, Content-Type.
o Understand the class of each error code received and notify
the end-user, if one is present, of error codes in classes 4xx
and 5xx. The notification requirement may be relaxed if the
end-user explicitly does not want it for one or all status
codes.
o Expect and respond to asynchronous requests from the server,
such as REDIRECT. This does not necessarily mean that it
should implement the REDIRECT method, merely that it MUST
respond positively or negatively to any request received from
the server.
Though not required, the following are RECOMMENDED.
o Implement RTP/AVP/UDP as a valid transport.
o Inclusion of the User-Agent header.
o Understand SDP session descriptions as defined in Appendix C
o Accept media initialization formats (such as SDP) from
standard input, command line, or other means appropriate to
the operating environment to act as a "helper application" for
other applications (such as web browsers).
There may be RTSP applications different from those
initially envisioned by the contributors to the RTSP
specification for which the requirements above do not make
sense. Therefore, the recommendations above serve only as
guidelines instead of strict requirements.
D.5.1 Basic Playback
To support on-demand playback of media streams, the client MUST
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additionally be able to do the following:
o generate the PAUSE request;
o implement the REDIRECT method, and the Location header.
D.5.2 Authentication-enabled
In order to access media presentations from RTSP servers that require
authentication, the client MUST additionally be able to do the
following:
o recognize the 401 (Unauthorized) status code;
o parse and include the WWW-Authenticate header;
o implement Basic Authentication and Digest Authentication.
D.6 Server
A minimal server implementation MUST be able to do the following:
o Implement the following methods: SETUP, TEARDOWN, OPTIONS,
SET_PARAMETER and PLAY.
o Include the following headers in responses: Connection,
Content-Length, Content-Type, Content-Language, Content-
Encoding, Timestamp, Transport, Proxy-Supported, Public, and
Via, and Unsupported. RTP-compliant implementations MUST also
implement the RTP-Info field.
o Parse and respond appropriately to the following headers in
requests: Connection, Proxy-Require, Session, Transport, and
Require.
o Implement Date header if supporting DESCRIBE
Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial
implementations and/or to be a "good citizen".
o Implement RTP/AVP/UDP as a valid transport.
o Inclusion of the Server, Cache-Control, and Expires headers.
o Implement the DESCRIBE method.
o Generate SDP session descriptions as defined in Appendix C
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There may be RTSP applications different from those
initially envisioned by the contributors to the RTSP
specification for which the requirements above do not make
sense. Therefore, the recommendations above serve only as
guidelines instead of strict requirements.
D.6.1 Basic Playback
To support on-demand playback of media streams, the server MUST
additionally be able to do the following:
o Recognize the Range header, and return an error if seeking is
not supported.
o Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface
features, the following are highly recommended for on-demand media
servers:
o Include and parse the Range header, with NPT units.
Implementation of SMPTE units is recommended.
o Include the length of the media presentation in the media
initialization information.
o Include mappings from data-specific timestamps to NPT. When
RTP is used, the rtptime portion of the RTP-Info field may be
used to map RTP timestamps to NPT.
Client implementations may use the presence of length
information to determine if the clip is seekable, and
visably disable seeking features for clips for which the
length information is unavailable. A common use of the
presentation length is to implement a "slider bar" which
serves as both a progress indicator and a timeline
positioning tool.
Mappings from RTP timestamps to NPT are necessary to ensure correct
positioning of the slider bar.
D.6.2 Authentication-enabled
In order to correctly handle client authentication, the server MUST
additionally be able to do the following:
o Generate the 401 (Unauthorized) status code when
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authentication is required for the resource.
o Parse and include the WWW-Authenticate header
o Implement Basic Authentication and Digest Authentication
E Requirements for Unreliable Transport of RTSP messages
This section provides any one intending to define how to transport of
RTSP messages over a unreliable transport protocol with some
information learned by the attempt in RFC 2326 [24]. RFC 2326 define
both an URI scheme and some basic functionality for transport of RTSP
messages over UDP, however it was not sufficient for reliable usage
and successful interoperability.
The RTSP scheme defined for unreliable transport of RTSP messages was
"rtspu". It has been reserved by this specification as at least one
commercial implementation exist, thus avoiding any collisions in the
name space.
The following considerations should exist for operation of RTSP over
an unreliable transport protocol:
o Request shall be acknowledged by the receiver. If there is no
acknowledgement, the sender may resend the same message after
a timeout of one round-trip time (RTT). Any retransmissions
due to lack of acknowledgement must carry the same sequence
number as the original request.
o The round-trip time can be estimated as in TCP (RFC 1123)
[40], with an initial round-trip value of 500 ms. An
implementation may cache the last RTT measurement as the
initial value for future connections.
o If RTSP is used over a small-RTT LAN, standard procedures for
optimizing initial TCP round trip estimates, such as those
used in T/TCP (RFC 1644) [41], can be beneficial.
o The Timestamp header (Section 14.44) is used to avoid the
retransmission ambiguity problem [42] and obviates the need
for Karn's algorithm.
o The registered default port for RTSP over UDP for the server
is 554.
o RTSP messages can be carried over any lower-layer transport
protocol that is 8-bit clean.
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o RTSP messages are vulnerable to bit errors and should not be
subjected to them.
o Source authentication, or at least validation that RTSP
messages comes from the same entity becomes extremely
important, as session hijacking may be substantially easier
for RTSP message transport using an unreliable protocol like
UDP than for TCP.
There exist two RTSP headers thats primarily are intended for being
used by the unreliable handling of RTSP messages and which will be
maintained:
CSeq See section 14.19
Timestamp See section 14.44
F Backwards Compatibility Considerations
This section contains notes on issues about backwards compatibility
with clients or servers being implemented according to RFC 2326 [24].
A server implementing RTSP/1.1 MUST include a RTSP-Version of
RTSP/1.1 in all responses to requests containing RTSP-Version
RTSP/1.1. If a server receives a RTSP/1.0 request, it MAY respond
with a RTSP/1.0 response if it chooses to support RFC 2326. If the
server chooses not to support RFC 2326, it SHOULD respond with a 505
(RTSP Version not supported) status code. A server MUST NOT respond
to a RTSP-Version RTSP/1.0 request with a RTSP-Version RTSP/1.1
response.
Clients implementing RTSP/1.1 SHOULD use an OPTIONS request with a
RTSP-Version of 1.1 to determine whether a server supports RTSP/1.1.
If the server responds with either a RTSP-Version of 1.0 or a status
code of 505 (RTSP Version not supported), the client MAY use RTSP/1.0
requests if it chooses to support RFC 2326. A client SHOULD NOT send
RTSP/1.1 requests to a server which has previously responded to a
RTSP/1.1 request with a RTSP-Version of 1.0.
F.1 Play Request in Play mode
The behavior in the server when a Play is received in Play mode has
changed (Section 11.4). In RFC 2326, the new PLAY request would be
queued until the current Play completed. Any new PLAY request now
take effect immediately replacing the previous request.
F.2 Using Persistent Connections
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Some server implementations of RFC 2326 maintain a one-to-one
relationship between a connection and an RTSP session. Such
implementations require clients to use a persistent connection to
communicate with the server and when a client closes its connection,
the server may remove the RTSP session. This is worth noting if a
RTSP 1.1 client connects to a 1.0 server.
G Open Issues
This section contains a list of open issues that still needs to be
resolved. However also any open issues in the bug tracker at
http://rtspspec.sourceforge.net should also be considered.
1. Should the Allow header be possible to use optional in
request or responses of DESCRIBE and SETUP besides the now
specified 405 error code and OPTIONS?
2. The minimal implementation chapter is still under
refinement. All shall, must and shoulds needs to be
included in the minimal and relevant feature tags.
Feature-tags for these needs to be defined. Further
feature-tags needs to be discussed.
3. The RTSP state machine is kind of not as useful as one
could desire. Should something be done about this? See
http://www1.ietf.org/mail-
archive/web/mmusic/current/msg03542.html
H Changes
Compared to RTSP 1.0 (RFC 2326), the below changes has been made when
defining RTSP 1.1. Note that this list does not reflect minor changes
in wording or correction of typographical errors.
o The Transport header has been changed in the following way:
- The ABNF has been changed to define that extensions are
possible, and that unknown extension parameters are to be
ignored.
- To prevent backwards compatibility issues, any extension or
new parameter requires the usage of a feature tag combined
with the Require header.
- Syntax unclarities with the Mode parameter has been
resolved.
- Syntax error with ";" for multicast and unicast has been
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resolved.
- Two new addressing parameters has been defined, src_addr and
dest_addr. These replaces the parameters "port",
"client_port", "server_port", "destination", "source".
- Support for IPv6 explicit addresses in all address fields
has been included.
- To handle URI definitions that contain ";" or "," a quoted
URI format has been introduced and is required.
- Defined IANA registries for the transport headers
parameters, transport-protocol, profile, lower-transport,
and mode.
- The transport headers interleaved parameter's text was made
more strict and use formal requirements levels. However no
change on how it is used was made.
- It has been clarified that the client can't request of the
server to use a certain RTP SSRC, using a request with the
transport parameter SSRC.
- Syntax definition for SSRC has been clarified to require 8*8
HEX. It has also been extend to allow multiple values for
clients supporting this version.
- Clarified the text on the transport headers "dest_addr"
parameters regarding what security precautions the server is
required to perform.
- The embedded (interleaved) binary data and its transport
parameter was clarified to being symmetric and that it is
the server that sets the channel numbers.
o The Range formats has been changed in the following way:
- The NPT format has been given a initial NPT identifier that
must now be used.
- All formats now support initial open ended formats of type
"npt=-10".
o RTSP message handling has been changed in the following way:
- RTSP messages now uses URIs rather then URLs.
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- It has been clarified that a 4xx message due to missing CSeq
header shall be returned without a CSeq header.
- Rules for how to handle timing out RTSP messages has been
added.
o The HTTP references has been updated to RFC 2616 and RFC 2617.
This has resulted in that the Public, and the Content-Base
header needed to be defined in the RTSP specification. Known
effects on RTSP due to HTTP clarifications:
- Content-Encoding header can include encoding of type
"identity".
o The state machine section has completely been rewritten. It
includes now more details and are also more clear about the
model used.
o A IANA section has been included with contains a number of
registries and their rules. This will allow us to use IANA to
keep track of all RTSP extensions.
o Than transport of RTSP messages has seen the following
changes:
- The use of UDP for RTSP message transport has been
deprecated due to missing interest and to broken
specification.
- The rules for how TCP connections is to be handled has been
clarified. Now it is made clear that servers should not
close the TCP connection unless they have been unused for
significant time.
- Strong recommendations why server and clients should use
persistent connections has also been added.
- There is now a requirement on the servers to handle non-
persistent connections as this provides fault tolerance.
- Added wording on the usage of Connection:Close for RTSP.
- specified usage of TLS for RTSP messages, including a scheme
to approve a proxies TLS connection to the next hop.
o The following header related changes have been made:
- Accept-Ranges response header is added. This header
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clarifies which range formats that can be used for a
resource.
- Clarified that Range header allows multiple ranges to allow
for creating editing list.
- Fixed the missing definitions for the Cache-Control header.
Also added to the syntax definition the missing delta-
seconds for max-stale and min-fresh parameters.
- Put requirement on CSeq header that the value is increased
by one for each new RTSP request. A Recommendation to start
at 1 has also been added.
- Added requirement that the Date header must be used for all
messages with entity. Also the Server should always include
it.
- Removed possibility of using Range header with Scale header
to indicate when it is to be activated, since it can't work
as defined. Also added rule that lack of Scale header in
response indicates lack of support for the header. Feature-
tags for scaled playback has been defined.
- The Speed header must now be responded to indicate support
and the actual speed going to be used. A feature-tag is
defined. Notes on congestion control was also added.
- The Supported header was borrowed from SIP to help with the
feature negotiation in RTSP.
- Clarified that the Timestamp header can be used to resolve
retransmission ambiguities.
- The Session header text has been expanded with a explanation
on keep alive and which methods to use. SET_PARAMETER is now
recommended to use if only keep-alive within RTSP is
desired.
- It has been clarified how the Range header formats is used
to indicate pause points.
- Clarified that RTP-Info URIs that are relative, uses the
Request-URI as base URI. Also clarified that used URI must
be that one that was used in the SETUP request. They are now
also required to be quoted. The header also expresses the
SSRC for the provided RTP timestamp and sequence number
values.
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Internet Draft RTSP July 18, 2005
- Added text that requires the Range to always be present in
PLAY responses. Clarified what should be sent in case of
live streams.
- The headers table has been updated using a structured
borrowed from SIP. This table carries much more information
and should provide a good overview of the available headers.
- It has been is clarified that any message with a message
body is required to have a Content-Length header. This was
the case in RFC 2326 but could be misinterpreted.
- To resolve functionality around ETag. The ETag and If-None-
Match header has been added from HTTP with necessary
clarification in regards to RTSP operation.
- Imported the Public header from HTTP RFC 2068 [18] since it
has been removed from HTTP due to lack of use. Public is
used quite frequently in RTSP.
- Clarified rules for populating the Public header so that it
is an intersection of the capabilities of all the RTSP
agents in a chain.
o The Protocol Syntax has been changed in the following way:
- All BNF definitions are updated according to the rules
defined in RFC 2234 [4] and has been gathered in a separate
section 19.
- The BNF for the User-Agent and Server headers has been
corrected so now only the description is in the HTTP
specification.
- The definition in the introduction of the RTSP session has
been changed.
- The protocol has been made fully IPv6 capable. Certain of
the functionality, like using explicit IPv6 addresses in
fields requires that the protocol support this updated
specification.
- Added a fragment part to the RTSP URI. This seem to be
indicated by the note below the definition however it was
not part of the BNF.
- The CHAR rule has been changed to exclude NULL.
H. Schulzrinne et. al. [Page 173]
Internet Draft RTSP July 18, 2005
o The Status codes has been changed in the following way:
- The use of status code 303 "See Other" has been deprecated
as it does not make sense to use in RTSP.
- When sending response 451 and 458 the response body should
contain the offending parameters.
- Clarification on when a 3rr redirect status code can be
received has been added. This includes receiving 3rr as a
result of request within a established session. This
provides clarification to a previous unspecified behavior.
- Removed the 250 (Low On Storage Space) status code as it
only is relevant to recording which is deprecated.
o The following functionality has been deprecated from the
protocol:
- The use of Queued Play.
- The use of PLAY method for keep-alive in play state.
- The RECORD and ANNOUNCE methods and all related
functionality. Some of the syntax has been removed.
- The possibility to use timed execution of methods with the
time parameter in the Range header.
- The description on how rtspu works is not part of the core
specification and will require external description. Only
that it exist is defined here and some requirements for the
the transport is provided.
o The following changes has been made in relation to methods:
- The OPTIONS method has been clarified with regards to the
use of the Public and Allow headers.
- The RECORD and ANNOUNCE methods are removed as they are
lacking implementation and not considered necessary in the
core specification. Any work on these methods should be done
as a extension document to RTSP.
- Added text clarifying the usage of SET_PARAMETER for keep-
alive and usage without any body.
o Wrote a new section about how to setup different media
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transport alternatives and their profiles, and lower layer
protocols. This resulted that the appendix on RTP interaction
was moved there instead in the part describing RTP. The
section also includes guidelines what to think of when writing
usage guidelines for new protocols and profiles.
o Added a new section describing the available mechanisms to
determine if functionality is supported, called "Capability
Handling". Renamed option-tags to feature-tags.
o Added a contributors section with people who has contribute
actual text to the specification.
o Added a section Use Cases that describes the major use cases
for RTSP.
o Clarified the usage of a=range and how to indicate live
content that are not seekable with this header.
o Text specifying the special behavior of PLAY for live content.
H.1 Changes needing to be updated
o The minimal implementation specification has been changed:
- Required Timestamp, Via, and Unsupported headers for a minimal
server implementation.
- Recommended that Cache-Control, Expires and Date headers be
supported by server implementations.
I Author Addresses
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
electronic mail: schulzrinne@cs.columbia.edu
Anup Rao
Cisco
USA
electronic mail: anrao@cisco.com
Robert Lanphier
H. Schulzrinne et. al. [Page 175]
Internet Draft RTSP July 18, 2005
RealNetworks
P.O. Box 91123
Seattle, WA 98111-9223
USA
electronic mail: robla@real.com
Magnus Westerlund
Ericsson AB, EAB/TVA/A
Torshamsgatan 23
SE-164 80 STOCKHOLM
SWEDEN
electronic mail: magnus.westerlund@ericsson.com
Aravind Narasimhan
Overture Computing Corp.,
East Windsor, NJ 08520
USA
electronic mail: aravind.narasimhan@gmail.com
J Contributors
The following people have made written contributions that were
included in the specification:
o Tom Marshall contributed text on the usage of 3rr status
codes.
o Thomas Zheng contributed text on the usage of the Range in
PLAY responses.
o Sean Sheedy contributed text on the timeout behavior of RTSP
messages and connections, and the 463 status code.
o Fredrik Lindholm contributed text about the RTSP security
framework.
The following people have provided detailed comments on updated
versions of this specification:
o Stephan Wenger
K Acknowledgements
This draft is based on the functionality of the original RTSP draft
submitted in October 1996. It also borrows format and descriptions
from HTTP/1.1.
This document has benefited greatly from the comments of all those
H. Schulzrinne et. al. [Page 176]
Internet Draft RTSP July 18, 2005
participating in the MMUSIC-WG. In addition to those already
mentioned, the following individuals have contributed to this
specification:
Rahul Agarwal, Jeff Ayars, Milko Boic, Torsten Braun, Brent Browning,
Bruce Butterfield, Steve Casner, Francisco Cortes, Kelly Djahandari,
Martin Dunsmuir, Eric Fleischman, Jay Geagan, Andy Grignon, V.
Guruprasad, Peter Haight, Mark Handley, Brad Hefta-Gaub, Volker Hilt,
John K. Ho, Go Hori, Philipp Hoschka, Anne Jones, Anders Klemets,
Ruth Lang, Stephanie Leif, Jonathan Lennox, Eduardo F. Llach, Thomas
Marshall, Rob McCool, David Oran, Joerg Ott, Maria Papadopouli, Sujal
Patel, Ema Patki, Alagu Periyannan, Colin Perkins, Igor Plotnikov,
Jonathan Sergent, Pinaki Shah, David Singer, Lior Sion, Jeff Smith,
Alexander Sokolsky, Dale Stammen, John Francis Stracke, Maureen
Chesire, David Walker, Geetha Srikantan, Stephan Wenger, Pekka Pessi,
Jae-Hwan Kim and Mela Martti.
L Normative References
[1] M. Handley and V. Jacobson, "SDP: session description protocol,"
RFC 2327, Internet Engineering Task Force, Apr. 1998.
[2] H. Schulzrinne and S. Casner, "RTP profile for audio and video
conferences with minimal control," RFC 3551, Internet Engineering
Task Force, July 2003.
[3] R. Fielding, J. Gettys, J. C. Mogul, H. Frystyk, L. Masinter, P.
J. Leach, and T. Berners-Lee, "Hypertext transfer protocol --
HTTP/1.1," RFC 2616, Internet Engineering Task Force, June 1999.
[4] "Augmented BNF for syntax specifications: ABNF," RFC 2234,
Internet Engineering Task Force, Nov. 1997.
[5] S. Bradner, "Key words for use in RFCs to indicate requirement
levels," RFC 2119, Internet Engineering Task Force, Mar. 1997.
[6] T. Dierks and C. Allen, "The TLS protocol version 1.0," RFC 2246,
Internet Engineering Task Force, Jan. 1999.
[7] J. Franks, P. Hallam-Baker, J. Hostetler, S. Lawrence, P. J.
Leach, A. Luotonen, and L. Stewart, "HTTP authentication: Basic and
digest access authentication," RFC 2617, Internet Engineering Task
Force, June 1999.
[8] J. B. Postel, "User datagram protocol," RFC 768, Internet
Engineering Task Force, Aug. 1980.
[9] J. B. Postel, "Transmission control protocol," RFC 793, Internet
H. Schulzrinne et. al. [Page 177]
Internet Draft RTSP July 18, 2005
Engineering Task Force, Sept. 1981.
[10] R. F. T. Berners-Lee and L. Masinter, "Uniform resource
identifier (uri): Generic syntax," RFC 3986, Internet Engineering
Task Force, Jan. 2005.
[11] R. Elz, "A compact representation of IPv6 addresses," RFC 1924,
Internet Engineering Task Force, Apr. 1996.
[12] R. Hinden, B. E. Carpenter, and L. Masinter, "Format for literal
IPv6 addresses in URL's," RFC 2732, Internet Engineering Task Force,
Dec. 1999.
[13] F. Yergeau, "UTF-8, a transformation format of ISO 10646," RFC
2279, Internet Engineering Task Force, Jan. 1998.
[14] NIST, "Fips pub 180-1:secure hash standard," tech. rep.,
National Institute of Standards and Technology, Apr. 1995.
[15] R. Housley, W. Polk, W. Ford, and D. Solo, "Internet X.509
public key infrastructure certificate and certificate revocation list
(CRL) profile," RFC 3280, Internet Engineering Task Force, Apr. 2002.
[16] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP:
a transport protocol for real-time applications," RFC 3550, Internet
Engineering Task Force, July 2003.
[17] E. Rescorla, "HTTP over TLS," RFC 2818, Internet Engineering
Task Force, May 2000.
[18] R. Fielding, J. Gettys, J. C. Mogul, H. Frystyk, and T.
Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068,
Internet Engineering Task Force, Jan. 1997.
[19] T. Narten and H. Alvestrand, "Guidelines for writing an IANA
considerations section in RFCs," RFC 2434, Internet Engineering Task
Force, Oct. 1998.
[20] e. a. J. Ott, "Extended rtp profile for rtcp-based feedback
(rtp/avpf)," internet draft, Internet Engineering Task Force, Aug.
2004. Work in progress.
[21] M. Baugher, D. McGrew, M. Naslund, E. Carrara, and K. Norrman,
"The secure real-time transport protocol (SRTP)," RFC 3711, Internet
Engineering Task Force, Mar. 2004.
[22] S. Olson, G. Camarillo, and A. B. Roach, "Support for IPv6 in
session description protocol (SDP)," RFC 3266, Internet Engineering
H. Schulzrinne et. al. [Page 178]
Internet Draft RTSP July 18, 2005
Task Force, June 2002.
[23] R. Hinden and S. E. Deering, "Internet protocol version 6 (ipv6)
addressing architecture," RFC 3513, Internet Engineering Task Force,
Apr. 2003.
M Informative References
[24] H. Schulzrinne, A. Rao, and R. Lanphier, "Real time streaming
protocol (RTSP)," RFC 2326, Internet Engineering Task Force, Apr.
1998.
[25] T. Z. M. Westerlund, "How to make real-time streaming protocol
(rtsp) traverse network address translators (nat) and interact with
firewalls.," internet draft, Internet Engineering Task Force, Feb.
2004. Work in progress.
[26] F. Yergeau, G. Nicol, G. C. Adams, and M. Duerst,
"Internationalization of the hypertext markup language," RFC 2070,
Internet Engineering Task Force, Jan. 1997.
[27] H. Schulzrinne, "A comprehensive multimedia control architecture
for the Internet," in Proc. International Workshop on Network and
Operating System Support for Digital Audio and Video (NOSSDAV), (St.
Louis, Missouri), May 1997.
[28] International Telecommunication Union, "Visual telephone systems
and equipment for local area networks which provide a non-guaranteed
quality of service," Recommendation H.323, Telecommunication
Standardization Sector of ITU, Geneva, Switzerland, May 1996.
[29] P. McMahon, "GSS-API authentication method for SOCKS version 5,"
RFC 1961, Internet Engineering Task Force, June 1996.
[30] J. Miller, P. Resnick, and D. Singer, "Rating services and
rating systems (and their machine readable descriptions),"
Recommendation REC-PICS-services-961031, W3C (World Wide Web
Consortium), Boston, Massachusetts, Oct. 1996.
[31] J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS label
distribution label syntax and communication protocols,"
Recommendation REC-PICS-labels-961031, W3C (World Wide Web
Consortium), Boston, Massachusetts, Oct. 1996.
[32] D. L. Mills, "Network time protocol (version 3) specification,
implementation," RFC 1305, Internet Engineering Task Force, Mar.
1992.
H. Schulzrinne et. al. [Page 179]
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[33] ISO/IEC, "Information technology -- generic coding of moving
pictures and associated audio informaiton -- part 6: extension for
digital storage media and control," Draft International Standard ISO
13818-6, International Organization for Standardization ISO/IEC
JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995.
[34] ISO/IEC, "Data elements and interchange formats -- information
interchange -- representation of dates and times," Published standard
ISO 8601, International Organization for Standardization ISO/IEC,
Geneva, Switzerland, Dec. 2000.
[35] S. Josefsson and I. W. Ed., "The base16, base32, and base64 data
encodings," RFC 3548, Internet Engineering Task Force, July 2003.
[36] Third Generation Partnership Project (3GPP), "Transparent end-
to-end packet-switched streaming service (pss); protocols and
codecs," Technical Specification 26.234, Third Generation Partnership
Project (3GPP), Dec. 2002.
[37] D. Yon, "Connection-oriented media transport in sdp," internet
draft, Internet Engineering Task Force, Mar. 2003. Work in progress.
[38] J. Lazzaro, "Framing rtp and rtcp packets over connection-
oriented transport," internet draft, Internet Engineering Task Force,
Oct. 2003. Work in progress.
[39] G. Camarillo, G. Eriksson, J. Holler, and H. Schulzrinne,
"Grouping of media lines in the session description protocol (SDP),"
RFC 3388, Internet Engineering Task Force, Dec. 2002.
[40] "Requirements for Internet hosts - application and support," RFC
1123, Internet Engineering Task Force, Oct. 1989.
[41] R. Braden, "T/TCP -- TCP extensions for transactions functional
specification," RFC 1644, Internet Engineering Task Force, July 1994.
[42] W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
Reading, Massachusetts: Addison-Wesley, 1994.
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H. Schulzrinne et. al. [Page 180]
Internet Draft RTSP July 18, 2005
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