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method direction object requirement
DESCRIBE C->S P,S recommended
ANNOUNCE C->S, S->C P,S optional
GET_PARAMETER C->S, S->C P,S optional
OPTIONS C->S, S->C P,S required
(S->C: optional)
PAUSE C->S P,S recommended
PLAY C->S P,S required
RECORD C->S P,S optional
REDIRECT S->C P,S optional
SETUP C->S S required
SET_PARAMETER C->S, S->C P,S optional
TEARDOWN C->S P,S required
Table 2: Overview of RTSP methods, their direction, and what
objects (P: presentation, S: stream) they operate on
Notes on Table 2: PAUSE is recommended, but not required in that a
fully functional server can be built that does not support this
method, for example, for live feeds. If a server does not support a
particular method, it MUST return "501 Not Implemented" and a client
SHOULD not try this method again for this server.
10.1 OPTIONS
The behavior is equivalent to that described in [H9.2]. An OPTIONS
request may be issued at any time, e.g., if the client is about to
try a nonstandard request. It does not influence server state.
Example:
C->S: OPTIONS * RTSP/1.0
CSeq: 1
Require: implicit-play
Proxy-Require: gzipped-messages
S->C: RTSP/1.0 200 OK
CSeq: 1
Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE
Note that these are necessarily fictional features (one would hope
that we would not purposefully overlook a truly useful feature just
so that we could have a strong example in this section).
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10.2 DESCRIBE
The DESCRIBE method retrieves the description of a presentation or
media object identified by the request URL from a server. It may use
the Accept header to specify the description formats that the client
understands. The server responds with a description of the requested
resource. The DESCRIBE reply-response pair constitutes the media
initialization phase of RTSP.
Example:
C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
Accept: application/sdp, application/rtsl, application/mheg
S->C: RTSP/1.0 200 OK
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
Content-Type: application/sdp
Content-Length: 376
v=0
o=mhandley 2890844526 2890842807 IN IP4 126.16.64.4
s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
[email protected] (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
m=whiteboard 32416 UDP WB
a=orient:portrait
The DESCRIBE response MUST contain all media initialization
information for the resource(s) that it describes. If a media client
obtains a presentation description from a source other than DESCRIBE
and that description contains a complete set of media initialization
parameters, the client SHOULD use those parameters and not then
request a description for the same media via RTSP.
Additionally, servers SHOULD NOT use the DESCRIBE response as a means
of media indirection.
Clear ground rules need to be established so that clients have an
unambiguous means of knowing when to request media initialization
information via DESCRIBE, and when not to. By forcing a DESCRIBE
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response to contain all media initialization for the set of streams
that it describes, and discouraging use of DESCRIBE for media
indirection, we avoid looping problems that might result from other
approaches.
Media initialization is a requirement for any RTSP-based system,
but the RTSP specification does not dictate that this must be done
via the DESCRIBE method. There are three ways that an RTSP client
may receive initialization information:
* via RTSP's DESCRIBE method;
* via some other protocol (HTTP, email attachment, etc.);
* via the command line or standard input (thus working as a browser
helper application launched with an SDP file or other media
initialization format).
In the interest of practical interoperability, it is highly
recommended that minimal servers support the DESCRIBE method, and
highly recommended that minimal clients support the ability to act
as a "helper application" that accepts a media initialization file
from standard input, command line, and/or other means that are
appropriate to the operating environment of the client.
10.3 ANNOUNCE
The ANNOUNCE method serves two purposes:
When sent from client to server, ANNOUNCE posts the description of a
presentation or media object identified by the request URL to a
server. When sent from server to client, ANNOUNCE updates the session
description in real-time.
If a new media stream is added to a presentation (e.g., during a live
presentation), the whole presentation description should be sent
again, rather than just the additional components, so that components
can be deleted.
Example:
C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0
CSeq: 312
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Content-Type: application/sdp
Content-Length: 332
v=0
o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4
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s=SDP Seminar
i=A Seminar on the session description protocol
u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps
[email protected] (Mark Handley)
c=IN IP4 224.2.17.12/127
t=2873397496 2873404696
a=recvonly
m=audio 3456 RTP/AVP 0
m=video 2232 RTP/AVP 31
S->C: RTSP/1.0 200 OK
CSeq: 312
10.4 SETUP
The SETUP request for a URI specifies the transport mechanism to be
used for the streamed media. A client can issue a SETUP request for a
stream that is already playing to change transport parameters, which
a server MAY allow. If it does not allow this, it MUST respond with
error "455 Method Not Valid In This State". For the benefit of any
intervening firewalls, a client must indicate the transport
parameters even if it has no influence over these parameters, for
example, where the server advertises a fixed multicast address.
Since SETUP includes all transport initialization information,
firewalls and other intermediate network devices (which need this
information) are spared the more arduous task of parsing the
DESCRIBE response, which has been reserved for media
initialization.
The Transport header specifies the transport parameters acceptable to
the client for data transmission; the response will contain the
transport parameters selected by the server.
C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Transport: RTP/AVP;unicast;client_port=4588-4589
S->C: RTSP/1.0 200 OK
CSeq: 302
Date: 23 Jan 1997 15:35:06 GMT
Session: 47112344
Transport: RTP/AVP;unicast;
client_port=4588-4589;server_port=6256-6257
The server generates session identifiers in response to SETUP
requests. If a SETUP request to a server includes a session
identifier, the server MUST bundle this setup request into the
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existing session or return error "459 Aggregate Operation Not
Allowed" (see Section 11.3.10).
10.5 PLAY
The PLAY method tells the server to start sending data via the
mechanism specified in SETUP. A client MUST NOT issue a PLAY request
until any outstanding SETUP requests have been acknowledged as
successful.
The PLAY request positions the normal play time to the beginning of
the range specified and delivers stream data until the end of the
range is reached. PLAY requests may be pipelined (queued); a server
MUST queue PLAY requests to be executed in order. That is, a PLAY
request arriving while a previous PLAY request is still active is
delayed until the first has been completed.
This allows precise editing.
For example, regardless of how closely spaced the two PLAY requests
in the example below arrive, the server will first play seconds 10
through 15, then, immediately following, seconds 20 to 25, and
finally seconds 30 through the end.
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 835
Session: 12345678
Range: npt=10-15
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 836
Session: 12345678
Range: npt=20-25
C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0
CSeq: 837
Session: 12345678
Range: npt=30-
See the description of the PAUSE request for further examples.
A PLAY request without a Range header is legal. It starts playing a
stream from the beginning unless the stream has been paused. If a
stream has been paused via PAUSE, stream delivery resumes at the
pause point. If a stream is playing, such a PLAY request causes no
further action and can be used by the client to test server liveness.
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The Range header may also contain a time parameter. This parameter
specifies a time in UTC at which the playback should start. If the
message is received after the specified time, playback is started
immediately. The time parameter may be used to aid in synchronization
of streams obtained from different sources.
For a on-demand stream, the server replies with the actual range that
will be played back. This may differ from the requested range if
alignment of the requested range to valid frame boundaries is
required for the media source. If no range is specified in the
request, the current position is returned in the reply. The unit of
the range in the reply is the same as that in the request.
After playing the desired range, the presentation is automatically
paused, as if a PAUSE request had been issued.
The following example plays the whole presentation starting at SMPTE
time code 0:10:20 until the end of the clip. The playback is to start
at 15:36 on 23 Jan 1997.
C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0
CSeq: 833
Session: 12345678
Range: smpte=0:10:20-;time=19970123T153600Z
S->C: RTSP/1.0 200 OK
CSeq: 833
Date: 23 Jan 1997 15:35:06 GMT
Range: smpte=0:10:22-;time=19970123T153600Z
For playing back a recording of a live presentation, it may be
desirable to use clock units:
C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0
CSeq: 835
Session: 12345678
Range: clock=19961108T142300Z-19961108T143520Z
S->C: RTSP/1.0 200 OK
CSeq: 835
Date: 23 Jan 1997 15:35:06 GMT
A media server only supporting playback MUST support the npt format
and MAY support the clock and smpte formats.
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10.6 PAUSE
The PAUSE request causes the stream delivery to be interrupted
(halted) temporarily. If the request URL names a stream, only
playback and recording of that stream is halted. For example, for
audio, this is equivalent to muting. If the request URL names a
presentation or group of streams, delivery of all currently active
streams within the presentation or group is halted. After resuming
playback or recording, synchronization of the tracks MUST be
maintained. Any server resources are kept, though servers MAY close
the session and free resources after being paused for the duration
specified with the timeout parameter of the Session header in the
SETUP message.
Example:
C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 834
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 834
Date: 23 Jan 1997 15:35:06 GMT
The PAUSE request may contain a Range header specifying when the
stream or presentation is to be halted. We refer to this point as the
"pause point". The header must contain exactly one value rather than
a time range. The normal play time for the stream is set to the pause
point. The pause request becomes effective the first time the server
is encountering the time point specified in any of the currently
pending PLAY requests. If the Range header specifies a time outside
any currently pending PLAY requests, the error "457 Invalid Range" is
returned. If a media unit (such as an audio or video frame) starts
presentation at exactly the pause point, it is not played or
recorded. If the Range header is missing, stream delivery is
interrupted immediately on receipt of the message and the pause point
is set to the current normal play time.
A PAUSE request discards all queued PLAY requests. However, the pause
point in the media stream MUST be maintained. A subsequent PLAY
request without Range header resumes from the pause point.
For example, if the server has play requests for ranges 10 to 15 and
20 to 29 pending and then receives a pause request for NPT 21, it
would start playing the second range and stop at NPT 21. If the pause
request is for NPT 12 and the server is playing at NPT 13 serving the
first play request, the server stops immediately. If the pause
request is for NPT 16, the server stops after completing the first
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play request and discards the second play request.
As another example, if a server has received requests to play ranges
10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE
request for NPT=14 would take effect while the server plays the first
range, with the second PLAY request effectively being ignored,
assuming the PAUSE request arrives before the server has started
playing the second, overlapping range. Regardless of when the PAUSE
request arrives, it sets the NPT to 14.
If the server has already sent data beyond the time specified in the
Range header, a PLAY would still resume at that point in time, as it
is assumed that the client has discarded data after that point. This
ensures continuous pause/play cycling without gaps.
10.7 TEARDOWN
The TEARDOWN request stops the stream delivery for the given URI,
freeing the resources associated with it. If the URI is the
presentation URI for this presentation, any RTSP session identifier
associated with the session is no longer valid. Unless all transport
parameters are defined by the session description, a SETUP request
has to be issued before the session can be played again.
Example:
C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 892
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 892
10.8 GET_PARAMETER
The GET_PARAMETER request retrieves the value of a parameter of a
presentation or stream specified in the URI. The content of the reply
and response is left to the implementation. GET_PARAMETER with no
entity body may be used to test client or server liveness ("ping").
Example:
S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 431
Content-Type: text/parameters
Session: 12345678
Content-Length: 15
packets_received
jitter
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C->S: RTSP/1.0 200 OK
CSeq: 431
Content-Length: 46
Content-Type: text/parameters
packets_received: 10
jitter: 0.3838
The "text/parameters" section is only an example type for
parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be
defined after further experimentation.
10.9 SET_PARAMETER
This method requests to set the value of a parameter for a
presentation or stream specified by the URI.
A request SHOULD only contain a single parameter to allow the client
to determine why a particular request failed. If the request contains
several parameters, the server MUST only act on the request if all of
the parameters can be set successfully. A server MUST allow a
parameter to be set repeatedly to the same value, but it MAY disallow
changing parameter values.
Note: transport parameters for the media stream MUST only be set with
the SETUP command.
Restricting setting transport parameters to SETUP is for the
benefit of firewalls.
The parameters are split in a fine-grained fashion so that there
can be more meaningful error indications. However, it may make
sense to allow the setting of several parameters if an atomic
setting is desirable. Imagine device control where the client does
not want the camera to pan unless it can also tilt to the right
angle at the same time.
Example:
C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 421
Content-length: 20
Content-type: text/parameters
barparam: barstuff
S->C: RTSP/1.0 451 Invalid Parameter
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CSeq: 421
Content-length: 10
Content-type: text/parameters
barparam
The "text/parameters" section is only an example type for
parameter. This method is intentionally loosely defined with the
intention that the reply content and response content will be
defined after further experimentation.
10.10 REDIRECT
A redirect request informs the client that it must connect to another
server location. It contains the mandatory header Location, which
indicates that the client should issue requests for that URL. It may
contain the parameter Range, which indicates when the redirection
takes effect. If the client wants to continue to send or receive
media for this URI, the client MUST issue a TEARDOWN request for the
current session and a SETUP for the new session at the designated
host.
This example request redirects traffic for this URI to the new server
at the given play time:
S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0
CSeq: 732
Location: rtsp://bigserver.com:8001
Range: clock=19960213T143205Z-
10.11 RECORD
This method initiates recording a range of media data according to
the presentation description. The timestamp reflects start and end
time (UTC). If no time range is given, use the start or end time
provided in the presentation description. If the session has already
started, commence recording immediately.
The server decides whether to store the recorded data under the
request-URI or another URI. If the server does not use the request-
URI, the response SHOULD be 201 (Created) and contain an entity which
describes the status of the request and refers to the new resource,
and a Location header.
A media server supporting recording of live presentations MUST
support the clock range format; the smpte format does not make sense.
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In this example, the media server was previously invited to the
conference indicated.
C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0
CSeq: 954
Session: 12345678
Conference: 128.16.64.19/32492374
10.12 Embedded (Interleaved) Binary Data
Certain firewall designs and other circumstances may force a server
to interleave RTSP methods and stream data. This interleaving should
generally be avoided unless necessary since it complicates client and
server operation and imposes additional overhead. Interleaved binary
data SHOULD only be used if RTSP is carried over TCP.
Stream data such as RTP packets is encapsulated by an ASCII dollar
sign (24 hexadecimal), followed by a one-byte channel identifier,
followed by the length of the encapsulated binary data as a binary,
two-byte integer in network byte order. The stream data follows
immediately afterwards, without a CRLF, but including the upper-layer
protocol headers. Each $ block contains exactly one upper-layer
protocol data unit, e.g., one RTP packet.
The channel identifier is defined in the Transport header with the
interleaved parameter(Section 12.39).
When the transport choice is RTP, RTCP messages are also interleaved
by the server over the TCP connection. As a default, RTCP packets are
sent on the first available channel higher than the RTP channel. The
client MAY explicitly request RTCP packets on another channel. This
is done by specifying two channels in the interleaved parameter of
the Transport header(Section 12.39).
RTCP is needed for synchronization when two or more streams are
interleaved in such a fashion. Also, this provides a convenient way
to tunnel RTP/RTCP packets through the TCP control connection when
required by the network configuration and transfer them onto UDP
when possible.
C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0
CSeq: 2
Transport: RTP/AVP/TCP;interleaved=0-1
S->C: RTSP/1.0 200 OK
CSeq: 2
Date: 05 Jun 1997 18:57:18 GMT
Transport: RTP/AVP/TCP;interleaved=0-1
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Session: 12345678
C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0
CSeq: 3
Session: 12345678
S->C: RTSP/1.0 200 OK
CSeq: 3
Session: 12345678
Date: 05 Jun 1997 18:59:15 GMT
RTP-Info: url=rtsp://foo.com/bar.file;
seq=232433;rtptime=972948234
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\000{2 byte length}{"length" bytes data, w/RTP header}
S->C: $\001{2 byte length}{"length" bytes RTCP packet}
11 Status Code Definitions
Where applicable, HTTP status [H10] codes are reused. Status codes
that have the same meaning are not repeated here. See Table 1 for a
listing of which status codes may be returned by which requests.
11.1 Success 2xx
11.1.1 250 Low on Storage Space
The server returns this warning after receiving a RECORD request that
it may not be able to fulfill completely due to insufficient storage
space. If possible, the server should use the Range header to
indicate what time period it may still be able to record. Since other
processes on the server may be consuming storage space
simultaneously, a client should take this only as an estimate.
11.2 Redirection 3xx
See [H10.3].
Within RTSP, redirection may be used for load balancing or
redirecting stream requests to a server topologically closer to the
client. Mechanisms to determine topological proximity are beyond the
scope of this specification.
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11.3 Client Error 4xx
11.3.1 405 Method Not Allowed
The method specified in the request is not allowed for the resource
identified by the request URI. The response MUST include an Allow
header containing a list of valid methods for the requested resource.
This status code is also to be used if a request attempts to use a
method not indicated during SETUP, e.g., if a RECORD request is
issued even though the mode parameter in the Transport header only
specified PLAY.
11.3.2 451 Parameter Not Understood
The recipient of the request does not support one or more parameters
contained in the request.
11.3.3 452 Conference Not Found
The conference indicated by a Conference header field is unknown to
the media server.
11.3.4 453 Not Enough Bandwidth
The request was refused because there was insufficient bandwidth.
This may, for example, be the result of a resource reservation
failure.
11.3.5 454 Session Not Found
The RTSP session identifier in the Session header is missing,
invalid, or has timed out.
11.3.6 455 Method Not Valid in This State
The client or server cannot process this request in its current
state. The response SHOULD contain an Allow header to make error
recovery easier.
11.3.7 456 Header Field Not Valid for Resource
The server could not act on a required request header. For example,
if PLAY contains the Range header field but the stream does not allow
seeking.
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11.3.8 457 Invalid Range
The Range value given is out of bounds, e.g., beyond the end of the
presentation.
11.3.9 458 Parameter Is Read-Only
The parameter to be set by SET_PARAMETER can be read but not
modified.
11.3.10 459 Aggregate Operation Not Allowed
The requested method may not be applied on the URL in question since
it is an aggregate (presentation) URL. The method may be applied on a
stream URL.
11.3.11 460 Only Aggregate Operation Allowed
The requested method may not be applied on the URL in question since
it is not an aggregate (presentation) URL. The method may be applied
on the presentation URL.
11.3.12 461 Unsupported Transport
The Transport field did not contain a supported transport
specification.
11.3.13 462 Destination Unreachable
The data transmission channel could not be established because the
client address could not be reached. This error will most likely be
the result of a client attempt to place an invalid Destination
parameter in the Transport field.
11.3.14 551 Option not supported
An option given in the Require or the Proxy-Require fields was not
supported. The Unsupported header should be returned stating the
option for which there is no support.
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12 Header Field Definitions
HTTP/1.1 [2] or other, non-standard header fields not listed here
currently have no well-defined meaning and SHOULD be ignored by the
recipient.
Table 3 summarizes the header fields used by RTSP. Type "g"
designates general request headers to be found in both requests and
responses, type "R" designates request headers, type "r" designates
response headers, and type "e" designates entity header fields.
Fields marked with "req." in the column labeled "support" MUST be
implemented by the recipient for a particular method, while fields
marked "opt." are optional. Note that not all fields marked "req."
will be sent in every request of this type. The "req." means only
that client (for response headers) and server (for request headers)
MUST implement the fields. The last column lists the method for which
this header field is meaningful; the designation "entity" refers to
all methods that return a message body. Within this specification,
DESCRIBE and GET_PARAMETER fall into this class.
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Header type support methods
Accept R opt. entity
Accept-Encoding R opt. entity
Accept-Language R opt. all
Allow r opt. all
Authorization R opt. all
Bandwidth R opt. all
Blocksize R opt. all but OPTIONS, TEARDOWN
Cache-Control g opt. SETUP
Conference R opt. SETUP
Connection g req. all
Content-Base e opt. entity
Content-Encoding e req. SET_PARAMETER
Content-Encoding e req. DESCRIBE, ANNOUNCE
Content-Language e req. DESCRIBE, ANNOUNCE
Content-Length e req. SET_PARAMETER, ANNOUNCE
Content-Length e req. entity
Content-Location e opt. entity
Content-Type e req. SET_PARAMETER, ANNOUNCE
Content-Type r req. entity
CSeq g req. all
Date g opt. all
Expires e opt. DESCRIBE, ANNOUNCE
From R opt. all
If-Modified-Since R opt. DESCRIBE, SETUP
Last-Modified e opt. entity
Proxy-Authenticate
Proxy-Require R req. all
Public r opt. all
Range R opt. PLAY, PAUSE, RECORD
Range r opt. PLAY, PAUSE, RECORD
Referer R opt. all
Require R req. all
Retry-After r opt. all
RTP-Info r req. PLAY
Scale Rr opt. PLAY, RECORD
Session Rr req. all but SETUP, OPTIONS
Server r opt. all
Speed Rr opt. PLAY
Transport Rr req. SETUP
Unsupported r req. all
User-Agent R opt. all
Via g opt. all
WWW-Authenticate r opt. all
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Overview of RTSP header fields
12.1 Accept
The Accept request-header field can be used to specify certain
presentation description content types which are acceptable for the
response.
The "level" parameter for presentation descriptions is properly
defined as part of the MIME type registration, not here.
See [H14.1] for syntax.
Example of use:
Accept: application/rtsl, application/sdp;level=2
12.2 Accept-Encoding
See [H14.3]
12.3 Accept-Language
See [H14.4]. Note that the language specified applies to the
presentation description and any reason phrases, not the media
content.
12.4 Allow
The Allow response header field lists the methods supported by the
resource identified by the request-URI. The purpose of this field is
to strictly inform the recipient of valid methods associated with the
resource. An Allow header field must be present in a 405 (Method not
allowed) response.
Example of use:
Allow: SETUP, PLAY, RECORD, SET_PARAMETER
12.5 Authorization
See [H14.8]
12.6 Bandwidth
The Bandwidth request header field describes the estimated bandwidth
available to the client, expressed as a positive integer and measured
in bits per second. The bandwidth available to the client may change
during an RTSP session, e.g., due to modem retraining.
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Bandwidth = "Bandwidth" ":" 1*DIGIT
Example:
Bandwidth: 4000
12.7 Blocksize
This request header field is sent from the client to the media server
asking the server for a particular media packet size. This packet
size does not include lower-layer headers such as IP, UDP, or RTP.
The server is free to use a blocksize which is lower than the one
requested. The server MAY truncate this packet size to the closest
multiple of the minimum, media-specific block size, or override it
with the media-specific size if necessary. The block size MUST be a
positive decimal number, measured in octets. The server only returns
an error (416) if the value is syntactically invalid.
12.8 Cache-Control
The Cache-Control general header field is used to specify directives
that MUST be obeyed by all caching mechanisms along the
request/response chain.
Cache directives must be passed through by a proxy or gateway
application, regardless of their significance to that application,
since the directives may be applicable to all recipients along the
request/response chain. It is not possible to specify a cache-
directive for a specific cache.
Cache-Control should only be specified in a SETUP request and its
response. Note: Cache-Control does not govern the caching of
responses as for HTTP, but rather of the stream identified by the
SETUP request. Responses to RTSP requests are not cacheable, except
for responses to DESCRIBE.
Cache-Control = "Cache-Control" ":" 1#cache-directive
cache-directive = cache-request-directive
| cache-response-directive
cache-request-directive = "no-cache"
| "max-stale"
| "min-fresh"
| "only-if-cached"
| cache-extension
cache-response-directive = "public"
| "private"
| "no-cache"
| "no-transform"
| "must-revalidate"
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| "proxy-revalidate"
| "max-age" "=" delta-seconds
| cache-extension
cache-extension = token [ "=" ( token | quoted-string ) ]
no-cache:
Indicates that the media stream MUST NOT be cached anywhere.
This allows an origin server to prevent caching even by caches
that have been configured to return stale responses to client
requests.
public:
Indicates that the media stream is cacheable by any cache.
private:
Indicates that the media stream is intended for a single user
and MUST NOT be cached by a shared cache. A private (non-
shared) cache may cache the media stream.
no-transform:
An intermediate cache (proxy) may find it useful to convert
the media type of a certain stream. A proxy might, for
example, convert between video formats to save cache space or
to reduce the amount of traffic on a slow link. Serious
operational problems may occur, however, when these
transformations have been applied to streams intended for
certain kinds of applications. For example, applications for
medical imaging, scientific data analysis and those using
end-to-end authentication all depend on receiving a stream
that is bit-for-bit identical to the original entity-body.
Therefore, if a response includes the no-transform directive,
an intermediate cache or proxy MUST NOT change the encoding of
the stream. Unlike HTTP, RTSP does not provide for partial
transformation at this point, e.g., allowing translation into
a different language.
only-if-cached:
In some cases, such as times of extremely poor network
connectivity, a client may want a cache to return only those
media streams that it currently has stored, and not to receive
these from the origin server. To do this, the client may
include the only-if-cached directive in a request. If it
receives this directive, a cache SHOULD either respond using a
cached media stream that is consistent with the other
constraints of the request, or respond with a 504 (Gateway
Timeout) status. However, if a group of caches is being
operated as a unified system with good internal connectivity,
such a request MAY be forwarded within that group of caches.
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max-stale:
Indicates that the client is willing to accept a media stream
that has exceeded its expiration time. If max-stale is
assigned a value, then the client is willing to accept a
response that has exceeded its expiration time by no more than
the specified number of seconds. If no value is assigned to
max-stale, then the client is willing to accept a stale
response of any age.
min-fresh:
Indicates that the client is willing to accept a media stream
whose freshness lifetime is no less than its current age plus
the specified time in seconds. That is, the client wants a
response that will still be fresh for at least the specified
number of seconds.
must-revalidate:
When the must-revalidate directive is present in a SETUP
response received by a cache, that cache MUST NOT use the
entry after it becomes stale to respond to a subsequent
request without first revalidating it with the origin server.
That is, the cache must do an end-to-end revalidation every
time, if, based solely on the origin server's Expires, the
cached response is stale.)
12.9 Conference
This request header field establishes a logical connection between a
pre-established conference and an RTSP stream. The conference-id must
not be changed for the same RTSP session.
Conference = "Conference" ":" conference-id Example:
Conference: [email protected]%20Starr
A response code of 452 (452 Conference Not Found) is returned if the
conference-id is not valid.
12.10 Connection
See [H14.10]
12.11 Content-Base
See [H14.11]
12.12 Content-Encoding
See [H14.12]
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12.13 Content-Language
See [H14.13]
12.14 Content-Length
This field contains the length of the content of the method (i.e.
after the double CRLF following the last header). Unlike HTTP, it
MUST be included in all messages that carry content beyond the header
portion of the message. If it is missing, a default value of zero is
assumed. It is interpreted according to [H14.14].
12.15 Content-Location
See [H14.15]
12.16 Content-Type
See [H14.18]. Note that the content types suitable for RTSP are
likely to be restricted in practice to presentation descriptions and
parameter-value types.
12.17 CSeq
The CSeq field specifies the sequence number for an RTSP request-
response pair. This field MUST be present in all requests and
responses. For every RTSP request containing the given sequence
number, there will be a corresponding response having the same
number. Any retransmitted request must contain the same sequence
number as the original (i.e. the sequence number is not incremented
for retransmissions of the same request).
12.18 Date
See [H14.19].
12.19 Expires
The Expires entity-header field gives a date and time after which the
description or media-stream should be considered stale. The
interpretation depends on the method:
DESCRIBE response:
The Expires header indicates a date and time after which the
description should be considered stale.
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A stale cache entry may not normally be returned by a cache (either a
proxy cache or an user agent cache) unless it is first validated with
the origin server (or with an intermediate cache that has a fresh
copy of the entity). See section 13 for further discussion of the
expiration model.
The presence of an Expires field does not imply that the original
resource will change or cease to exist at, before, or after that
time.
The format is an absolute date and time as defined by HTTP-date in
[H3.3]; it MUST be in RFC1123-date format:
Expires = "Expires" ":" HTTP-date
An example of its use is
Expires: Thu, 01 Dec 1994 16:00:00 GMT
RTSP/1.0 clients and caches MUST treat other invalid date formats,
especially including the value "0", as having occurred in the past
(i.e., "already expired").
To mark a response as "already expired," an origin server should use
an Expires date that is equal to the Date header value. To mark a
response as "never expires," an origin server should use an Expires
date approximately one year from the time the response is sent.
RTSP/1.0 servers should not send Expires dates more than one year in
the future.
The presence of an Expires header field with a date value of some
time in the future on a media stream that otherwise would by default
be non-cacheable indicates that the media stream is cacheable, unless
indicated otherwise by a Cache-Control header field (Section 12.8).
12.20 From
See [H14.22].
12.21 Host
This HTTP request header field is not needed for RTSP. It should be
silently ignored if sent.
12.22 If-Match
See [H14.25].
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This field is especially useful for ensuring the integrity of the
presentation description, in both the case where it is fetched via
means external to RTSP (such as HTTP), or in the case where the
server implementation is guaranteeing the integrity of the
description between the time of the DESCRIBE message and the SETUP
message.
The identifier is an opaque identifier, and thus is not specific to
any particular session description language.
12.23 If-Modified-Since
The If-Modified-Since request-header field is used with the DESCRIBE
and SETUP methods to make them conditional. If the requested variant
has not been modified since the time specified in this field, a
description will not be returned from the server (DESCRIBE) or a
stream will not be set up (SETUP). Instead, a 304 (not modified)
response will be returned without any message-body.
If-Modified-Since = "If-Modified-Since" ":" HTTP-date
An example of the field is:
If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT
12.24 Last-Modified
The Last-Modified entity-header field indicates the date and time at
which the origin server believes the presentation description or
media stream was last modified. See [H14.29]. For the methods
DESCRIBE or ANNOUNCE, the header field indicates the last
modification date and time of the description, for SETUP that of the
media stream.
12.25 Location
See [H14.30].
12.26 Proxy-Authenticate
See [H14.33].
12.27 Proxy-Require
The Proxy-Require header is used to indicate proxy-sensitive features
that MUST be supported by the proxy. Any Proxy-Require header
features that are not supported by the proxy MUST be negatively
acknowledged by the proxy to the client if not supported. Servers
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should treat this field identically to the Require field.
See Section 12.32 for more details on the mechanics of this message
and a usage example.
12.28 Public
See [H14.35].
12.29 Range
This request and response header field specifies a range of time.
The range can be specified in a number of units. This specification
defines the smpte (Section 3.5), npt (Section 3.6), and clock
(Section 3.7) range units. Within RTSP, byte ranges [H14.36.1] are
not meaningful and MUST NOT be used. The header may also contain a
time parameter in UTC, specifying the time at which the operation is
to be made effective. Servers supporting the Range header MUST
understand the NPT range format and SHOULD understand the SMPTE range
format. The Range response header indicates what range of time is
actually being played or recorded. If the Range header is given in a
time format that is not understood, the recipient should return "501
Not Implemented".
Ranges are half-open intervals, including the lower point, but
excluding the upper point. In other words, a range of a-b starts
exactly at time a, but stops just before b. Only the start time of a
media unit such as a video or audio frame is relevant. As an example,
assume that video frames are generated every 40 ms. A range of 10.0-
10.1 would include a video frame starting at 10.0 or later time and
would include a video frame starting at 10.08, even though it lasted
beyond the interval. A range of 10.0-10.08, on the other hand, would
exclude the frame at 10.08.
Range = "Range" ":" 1\#ranges-specifier
[ ";" "time" "=" utc-time ]
ranges-specifier = npt-range | utc-range | smpte-range
Example:
Range: clock=19960213T143205Z-;time=19970123T143720Z
The notation is similar to that used for the HTTP/1.1 [2] byte-
range header. It allows clients to select an excerpt from the media
object, and to play from a given point to the end as well as from
the current location to a given point. The start of playback can be
scheduled for any time in the future, although a server may refuse
to keep server resources for extended idle periods.
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12.30 Referer
See [H14.37]. The URL refers to that of the presentation description,
typically retrieved via HTTP.
12.31 Retry-After
See [H14.38].
12.32 Require
The Require header is used by clients to query the server about
options that it may or may not support. The server MUST respond to
this header by using the Unsupported header to negatively acknowledge
those options which are NOT supported.
This is to make sure that the client-server interaction will
proceed without delay when all options are understood by both
sides, and only slow down if options are not understood (as in the
case above). For a well-matched client-server pair, the interaction
proceeds quickly, saving a round-trip often required by negotiation
mechanisms. In addition, it also removes state ambiguity when the
client requires features that the server does not understand.
Require = "Require" ":" 1#option-tag
Example:
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 302
Require: funky-feature
Funky-Parameter: funkystuff
S->C: RTSP/1.0 551 Option not supported
CSeq: 302
Unsupported: funky-feature
C->S: SETUP rtsp://server.com/foo/bar/baz.rm RTSP/1.0
CSeq: 303
S->C: RTSP/1.0 200 OK
CSeq: 303
In this example, "funky-feature" is the feature tag which indicates
to the client that the fictional Funky-Parameter field is required.
The relationship between "funky-feature" and Funky-Parameter is not
communicated via the RTSP exchange, since that relationship is an
immutable property of "funky-feature" and thus should not be
transmitted with every exchange.
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Proxies and other intermediary devices SHOULD ignore features that
are not understood in this field. If a particular extension requires
that intermediate devices support it, the extension should be tagged
in the Proxy-Require field instead (see Section 12.27).
12.33 RTP-Info
This field is used to set RTP-specific parameters in the PLAY
response.
url:
Indicates the stream URL which for which the following RTP
parameters correspond.
seq:
Indicates the sequence number of the first packet of the
stream. This allows clients to gracefully deal with packets
when seeking. The client uses this value to differentiate
packets that originated before the seek from packets that
originated after the seek.
rtptime:
Indicates the RTP timestamp corresponding to the time value in
the Range response header. (Note: For aggregate control, a
particular stream may not actually generate a packet for the
Range time value returned or implied. Thus, there is no
guarantee that the packet with the sequence number indicated
by seq actually has the timestamp indicated by rtptime.) The
client uses this value to calculate the mapping of RTP time to
NPT.
A mapping from RTP timestamps to NTP timestamps (wall clock) is
available via RTCP. However, this information is not sufficient to
generate a mapping from RTP timestamps to NPT. Furthermore, in
order to ensure that this information is available at the necessary
time (immediately at startup or after a seek), and that it is
delivered reliably, this mapping is placed in the RTSP control
channel.
In order to compensate for drift for long, uninterrupted
presentations, RTSP clients should additionally map NPT to NTP,
using initial RTCP sender reports to do the mapping, and later
reports to check drift against the mapping.
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Syntax:
RTP-Info = "RTP-Info" ":" 1#stream-url 1*parameter
stream-url = "url" "=" url
parameter = ";" "seq" "=" 1*DIGIT
| ";" "rtptime" "=" 1*DIGIT
Example:
RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102,
url=rtsp://foo.com/bar.avi/streamid=1;seq=30211
12.34 Scale
A scale value of 1 indicates normal play or record at the normal
forward viewing rate. If not 1, the value corresponds to the rate
with respect to normal viewing rate. For example, a ratio of 2
indicates twice the normal viewing rate ("fast forward") and a ratio
of 0.5 indicates half the normal viewing rate. In other words, a
ratio of 2 has normal play time increase at twice the wallclock rate.
For every second of elapsed (wallclock) time, 2 seconds of content
will be delivered. A negative value indicates reverse direction.
Unless requested otherwise by the Speed parameter, the data rate
SHOULD not be changed. Implementation of scale changes depends on the
server and media type. For video, a server may, for example, deliver
only key frames or selected key frames. For audio, it may time-scale
the audio while preserving pitch or, less desirably, deliver
fragments of audio.
The server should try to approximate the viewing rate, but may
restrict the range of scale values that it supports. The response
MUST contain the actual scale value chosen by the server.
If the request contains a Range parameter, the new scale value will
take effect at that time.
Scale = "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ]
Example of playing in reverse at 3.5 times normal rate:
Scale: -3.5
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12.35 Speed
This request header fields parameter requests the server to deliver
data to the client at a particular speed, contingent on the server's
ability and desire to serve the media stream at the given speed.
Implementation by the server is OPTIONAL. The default is the bit rate
of the stream.
The parameter value is expressed as a decimal ratio, e.g., a value of
2.0 indicates that data is to be delivered twice as fast as normal. A
speed of zero is invalid. If the request contains a Range parameter,
the new speed value will take effect at that time.
Speed = "Speed" ":" 1*DIGIT [ "." *DIGIT ]
Example:
Speed: 2.5
Use of this field changes the bandwidth used for data delivery. It is
meant for use in specific circumstances where preview of the
presentation at a higher or lower rate is necessary. Implementors
should keep in mind that bandwidth for the session may be negotiated
beforehand (by means other than RTSP), and therefore re-negotiation
may be necessary. When data is delivered over UDP, it is highly
recommended that means such as RTCP be used to track packet loss
rates.
12.36 Server
See [H14.39]
12.37 Session
This request and response header field identifies an RTSP session
started by the media server in a SETUP response and concluded by
TEARDOWN on the presentation URL. The session identifier is chosen by
the media server (see Section 3.4). Once a client receives a Session
identifier, it MUST return it for any request related to that
session. A server does not have to set up a session identifier if it
has other means of identifying a session, such as dynamically
generated URLs.
Session = "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ]
The timeout parameter is only allowed in a response header. The
server uses it to indicate to the client how long the server is
prepared to wait between RTSP commands before closing the session due
to lack of activity (see Section A). The timeout is measured in
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seconds, with a default of 60 seconds (1 minute).
Note that a session identifier identifies a RTSP session across
transport sessions or connections. Control messages for more than one
RTSP URL may be sent within a single RTSP session. Hence, it is
possible that clients use the same session for controlling many
streams constituting a presentation, as long as all the streams come
from the same server. (See example in Section 14). However, multiple
"user" sessions for the same URL from the same client MUST use
different session identifiers.
The session identifier is needed to distinguish several delivery
requests for the same URL coming from the same client.
The response 454 (Session Not Found) is returned if the session
identifier is invalid.
12.38 Timestamp
The timestamp general header describes when the client sent the
request to the server. The value of the timestamp is of significance
only to the client and may use any timescale. The server MUST echo
the exact same value and MAY, if it has accurate information about
this, add a floating point number indicating the number of seconds
that has elapsed since it has received the request. The timestamp is
used by the client to compute the round-trip time to the server so
that it can adjust the timeout value for retransmissions.
Timestamp = "Timestamp" ":" *(DIGIT) [ "." *(DIGIT) ] [ delay ]
delay = *(DIGIT) [ "." *(DIGIT) ]
12.39 Transport
This request header indicates which transport protocol is to be used
and configures its parameters such as destination address,
compression, multicast time-to-live and destination port for a single
stream. It sets those values not already determined by a presentation
description.
Transports are comma separated, listed in order of preference.
Parameters may be added to each transport, separated by a semicolon.
The Transport header MAY also be used to change certain transport
parameters. A server MAY refuse to change parameters of an existing
stream.
The server MAY return a Transport response header in the response to
indicate the values actually chosen.
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A Transport request header field may contain a list of transport
options acceptable to the client. In that case, the server MUST
return a single option which was actually chosen.
The syntax for the transport specifier is
transport/profile/lower-transport.
The default value for the "lower-transport" parameters is specific to
the profile. For RTP/AVP, the default is UDP.
Below are the configuration parameters associated with transport:
General parameters:
unicast | multicast:
mutually exclusive indication of whether unicast or multicast
delivery will be attempted. Default value is multicast.
Clients that are capable of handling both unicast and
multicast transmission MUST indicate such capability by
including two full transport-specs with separate parameters
for each.
destination:
The address to which a stream will be sent. The client may
specify the multicast address with the destination parameter.
To avoid becoming the unwitting perpetrator of a remote-
controlled denial-of-service attack, a server SHOULD
authenticate the client and SHOULD log such attempts before
allowing the client to direct a media stream to an address not
chosen by the server. This is particularly important if RTSP
commands are issued via UDP, but implementations cannot rely
on TCP as reliable means of client identification by itself. A
server SHOULD not allow a client to direct media streams to an
address that differs from the address commands are coming
from.
source:
If the source address for the stream is different than can be
derived from the RTSP endpoint address (the server in playback
or the client in recording), the source MAY be specified.
This information may also be available through SDP. However, since
this is more a feature of transport than media initialization, the
authoritative source for this information should be in the SETUP
response.
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layers:
The number of multicast layers to be used for this media
stream. The layers are sent to consecutive addresses starting
at the destination address.
mode:
The mode parameter indicates the methods to be supported for
this session. Valid values are PLAY and RECORD. If not
provided, the default is PLAY.
append:
If the mode parameter includes RECORD, the append parameter
indicates that the media data should append to the existing
resource rather than overwrite it. If appending is requested
and the server does not support this, it MUST refuse the
request rather than overwrite the resource identified by the
URI. The append parameter is ignored if the mode parameter
does not contain RECORD.
interleaved:
The interleaved parameter implies mixing the media stream with
the control stream in whatever protocol is being used by the
control stream, using the mechanism defined in Section 10.12.
The argument provides the channel number to be used in the $
statement. This parameter may be specified as a range, e.g.,
interleaved=4-5 in cases where the transport choice for the
media stream requires it.
This allows RTP/RTCP to be handled similarly to the way that it is
done with UDP, i.e., one channel for RTP and the other for RTCP.
Multicast specific:
ttl:
multicast time-to-live
RTP Specific:
port:
This parameter provides the RTP/RTCP port pair for a multicast
session. It is specified as a range, e.g., port=3456-3457.
client_port:
This parameter provides the unicast RTP/RTCP port pair on
which the client has chosen to receive media data and control
information. It is specified as a range, e.g.,
client_port=3456-3457.
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server_port:
This parameter provides the unicast RTP/RTCP port pair on
which the server has chosen to receive media data and control
information. It is specified as a range, e.g.,
server_port=3456-3457.
ssrc:
The ssrc parameter indicates the RTP SSRC [24, Sec. 3] value
that should be (request) or will be (response) used by the
media server. This parameter is only valid for unicast
transmission. It identifies the synchronization source to be
associated with the media stream.
Transport = "Transport" ":"
1\#transport-spec
transport-spec = transport-protocol/profile[/lower-transport]
*parameter
transport-protocol = "RTP"
profile = "AVP"
lower-transport = "TCP" | "UDP"
parameter = ( "unicast" | "multicast" )
| ";" "destination" [ "=" address ]
| ";" "interleaved" "=" channel [ "-" channel ]
| ";" "append"
| ";" "ttl" "=" ttl
| ";" "layers" "=" 1*DIGIT
| ";" "port" "=" port [ "-" port ]
| ";" "client_port" "=" port [ "-" port ]
| ";" "server_port" "=" port [ "-" port ]
| ";" "ssrc" "=" ssrc
| ";" "mode" = <"> 1\#mode <">
ttl = 1*3(DIGIT)
port = 1*5(DIGIT)
ssrc = 8*8(HEX)
channel = 1*3(DIGIT)
address = host
mode = <"> *Method <"> | Method
Example:
Transport: RTP/AVP;multicast;ttl=127;mode="PLAY",
RTP/AVP;unicast;client_port=3456-3457;mode="PLAY"
The Transport header is restricted to describing a single RTP
stream. (RTSP can also control multiple streams as a single
entity.) Making it part of RTSP rather than relying on a multitude
of session description formats greatly simplifies designs of
firewalls.
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12.40 Unsupported
The Unsupported response header lists the features not supported by
the server. In the case where the feature was specified via the
Proxy-Require field (Section 12.32), if there is a proxy on the path
between the client and the server, the proxy MUST insert a message
reply with an error message "551 Option Not Supported".
See Section 12.32 for a usage example.
12.41 User-Agent
See [H14.42]
12.42 Vary
See [H14.43]
12.43 Via
See [H14.44].
12.44 WWW-Authentica
See [H14.46].
13 Caching
In HTTP, response-request pairs are cached. RTSP differs
significantly in that respect. Responses are not cacheable, with the
exception of the presentation description returned by DESCRIBE or
included with ANNOUNCE. (Since the responses for anything but
DESCRIBE and GET_PARAMETER do not return any data, caching is not
really an issue for these requests.) However, it is desirable for the
continuous media data, typically delivered out-of-band with respect
to RTSP, to be cached, as well as the session description.
On receiving a SETUP or PLAY request, a proxy ascertains whether it
has an up-to-date copy of the continuous media content and its
description. It can determine whether the copy is up-to-date by
issuing a SETUP or DESCRIBE request, respectively, and comparing the
Last-Modified header with that of the cached copy. If the copy is not
up-to-date, it modifies the SETUP transport parameters as appropriate
and forwards the request to the origin server. Subsequent control
commands such as PLAY or PAUSE then pass the proxy unmodified. The
proxy delivers the continuous media data to the client, while
possibly making a local copy for later reuse. The exact behavior
allowed to the cache is given by the cache-response directives
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described in Section 12.8. A cache MUST answer any DESCRIBE requests
if it is currently serving the stream to the requestor, as it is
possible that low-level details of the stream description may have
changed on the origin-server.
Note that an RTSP cache, unlike the HTTP cache, is of the "cut-
through" variety. Rather than retrieving the whole resource from the
origin server, the cache simply copies the streaming data as it
passes by on its way to the client. Thus, it does not introduce
additional latency.
To the client, an RTSP proxy cache appears like a regular media
server, to the media origin server like a client. Just as an HTTP
cache has to store the content type, content language, and so on for
the objects it caches, a media cache has to store the presentation
description. Typically, a cache eliminates all transport-references
(that is, multicast information) from the presentation description,
since these are independent of the data delivery from the cache to
the client. Information on the encodings remains the same. If the
cache is able to translate the cached media data, it would create a
new presentation description with all the encoding possibilities it
can offer.
14 Examples
The following examples refer to stream description formats that are
not standards, such as RTSL. The following examples are not to be
used as a reference for those formats.
14.1 Media on Demand (Unicast)
Client C requests a movie from media servers A ( audio.example.com)
and V (video.example.com). The media description is stored on a web
server W . The media description contains descriptions of the
presentation and all its streams, including the codecs that are
available, dynamic RTP payload types, the protocol stack, and content
information such as language or copyright restrictions. It may also
give an indication about the timeline of the movie.
In this example, the client is only interested in the last part of
the movie.
C->W: GET /twister.sdp HTTP/1.1
Host: www.example.com
Accept: application/sdp
W->C: HTTP/1.0 200 OK
Content-Type: application/sdp
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RFC 2326 Real Time Streaming Protocol April 1998
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 0 RTP/AVP 0
a=control:rtsp://audio.example.com/twister/audio.en
m=video 0 RTP/AVP 31
a=control:rtsp://video.example.com/twister/video
C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057
A->C: RTSP/1.0 200 OK
CSeq: 1
Session: 12345678
Transport: RTP/AVP/UDP;unicast;client_port=3056-3057;
server_port=5000-5001
C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 1
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059
V->C: RTSP/1.0 200 OK
CSeq: 1
Session: 23456789
Transport: RTP/AVP/UDP;unicast;client_port=3058-3059;
server_port=5002-5003
C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-
V->C: RTSP/1.0 200 OK
CSeq: 2
Session: 23456789
Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://video.example.com/twister/video;
seq=12312232;rtptime=78712811
C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 2
Session: 12345678
Range: smpte=0:10:00-
A->C: RTSP/1.0 200 OK
CSeq: 2
Session: 12345678
Schulzrinne, et. al. Standards Track [Page 64]
RFC 2326 Real Time Streaming Protocol April 1998
Range: smpte=0:10:00-0:20:00
RTP-Info: url=rtsp://audio.example.com/twister/audio.en;
seq=876655;rtptime=1032181
C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0
CSeq: 3
Session: 12345678
A->C: RTSP/1.0 200 OK
CSeq: 3
C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0
CSeq: 3
Session: 23456789
V->C: RTSP/1.0 200 OK
CSeq: 3
Even though the audio and video track are on two different servers,
and may start at slightly different times and may drift with respect
to each other, the client can synchronize the two using standard RTP
methods, in particular the time scale contained in the RTCP sender
reports.
14.2 Streaming of a Container file
For purposes of this example, a container file is a storage entity in
which multiple continuous media types pertaining to the same end-user
presentation are present. In effect, the container file represents an
RTSP presentation, with each of its components being RTSP streams.
Container files are a widely used means to store such presentations.
While the components are transported as independent streams, it is
desirable to maintain a common context for those streams at the
server end.
This enables the server to keep a single storage handle open
easily. It also allows treating all the streams equally in case of
any prioritization of streams by the server.
It is also possible that the presentation author may wish to prevent
selective retrieval of the streams by the client in order to preserve
the artistic effect of the combined media presentation. Similarly, in
such a tightly bound presentation, it is desirable to be able to
control all the streams via a single control message using an
aggregate URL.
The following is an example of using a single RTSP session to control
multiple streams. It also illustrates the use of aggregate URLs.
