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Client C requests a presentation from media server M . The movie is
stored in a container file. The client has obtained an RTSP URL to
the container file.
C->M: DESCRIBE rtsp://foo/twister RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 164
v=0
o=- 2890844256 2890842807 IN IP4 172.16.2.93
s=RTSP Session
i=An Example of RTSP Session Usage
a=control:rtsp://foo/twister
t=0 0
m=audio 0 RTP/AVP 0
a=control:rtsp://foo/twister/audio
m=video 0 RTP/AVP 26
a=control:rtsp://foo/twister/video
C->M: SETUP rtsp://foo/twister/audio RTSP/1.0
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;unicast;client_port=8000-8001;
server_port=9000-9001
Session: 12345678
C->M: SETUP rtsp://foo/twister/video RTSP/1.0
CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 3
Transport: RTP/AVP;unicast;client_port=8002-8003;
server_port=9004-9005
Session: 12345678
C->M: PLAY rtsp://foo/twister RTSP/1.0
CSeq: 4
Range: npt=0-
Session: 12345678
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M->C: RTSP/1.0 200 OK
CSeq: 4
Session: 12345678
RTP-Info: url=rtsp://foo/twister/video;
seq=9810092;rtptime=3450012
C->M: PAUSE rtsp://foo/twister/video RTSP/1.0
CSeq: 5
Session: 12345678
M->C: RTSP/1.0 460 Only aggregate operation allowed
CSeq: 5
C->M: PAUSE rtsp://foo/twister RTSP/1.0
CSeq: 6
Session: 12345678
M->C: RTSP/1.0 200 OK
CSeq: 6
Session: 12345678
C->M: SETUP rtsp://foo/twister RTSP/1.0
CSeq: 7
Transport: RTP/AVP;unicast;client_port=10000
M->C: RTSP/1.0 459 Aggregate operation not allowed
CSeq: 7
In the first instance of failure, the client tries to pause one
stream (in this case video) of the presentation. This is disallowed
for that presentation by the server. In the second instance, the
aggregate URL may not be used for SETUP and one control message is
required per stream to set up transport parameters.
This keeps the syntax of the Transport header simple and allows
easy parsing of transport information by firewalls.
14.3 Single Stream Container Files
Some RTSP servers may treat all files as though they are "container
files", yet other servers may not support such a concept. Because of
this, clients SHOULD use the rules set forth in the session
description for request URLs, rather than assuming that a consistent
URL may always be used throughout. Here's an example of how a multi-
stream server might expect a single-stream file to be served:
Accept: application/x-rtsp-mh, application/sdp
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CSeq: 1
S->C RTSP/1.0 200 OK
CSeq: 1
Content-base: rtsp://foo.com/test.wav/
Content-type: application/sdp
Content-length: 48
v=0
o=- 872653257 872653257 IN IP4 172.16.2.187
s=mu-law wave file
i=audio test
t=0 0
m=audio 0 RTP/AVP 0
a=control:streamid=0
C->S SETUP rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
Transport: RTP/AVP/UDP;unicast;
client_port=6970-6971;mode=play
CSeq: 2
S->C RTSP/1.0 200 OK
Transport: RTP/AVP/UDP;unicast;client_port=6970-6971;
server_port=6970-6971;mode=play
CSeq: 2
Session: 2034820394
C->S PLAY rtsp://foo.com/test.wav RTSP/1.0
CSeq: 3
Session: 2034820394
S->C RTSP/1.0 200 OK
CSeq: 3
Session: 2034820394
RTP-Info: url=rtsp://foo.com/test.wav/streamid=0;
seq=981888;rtptime=3781123
Note the different URL in the SETUP command, and then the switch back
to the aggregate URL in the PLAY command. This makes complete sense
when there are multiple streams with aggregate control, but is less
than intuitive in the special case where the number of streams is
one.
In this special case, it is recommended that servers be forgiving of
implementations that send:
C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0
CSeq: 3
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In the worst case, servers should send back:
S->C RTSP/1.0 460 Only aggregate operation allowed
CSeq: 3
One would also hope that server implementations are also forgiving of
the following:
C->S SETUP rtsp://foo.com/test.wav RTSP/1.0
Transport: rtp/avp/udp;client_port=6970-6971;mode=play
CSeq: 2
Since there is only a single stream in this file, it's not ambiguous
what this means.
14.4 Live Media Presentation Using Multicast
The media server M chooses the multicast address and port. Here, we
assume that the web server only contains a pointer to the full
description, while the media server M maintains the full description.
C->W: GET /concert.sdp HTTP/1.1
Host: www.example.com
W->C: HTTP/1.1 200 OK
Content-Type: application/x-rtsl
<session>
<track src="rtsp://live.example.com/concert/audio">
</session>
C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 1
M->C: RTSP/1.0 200 OK
CSeq: 1
Content-Type: application/sdp
Content-Length: 44
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
m=audio 3456 RTP/AVP 0
a=control:rtsp://live.example.com/concert/audio
c=IN IP4 224.2.0.1/16
C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 2
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Transport: RTP/AVP;multicast
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=224.2.0.1;
port=3456-3457;ttl=16
Session: 0456804596
C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0
CSeq: 3
Session: 0456804596
M->C: RTSP/1.0 200 OK
CSeq: 3
Session: 0456804596
14.5 Playing media into an existing session
A conference participant C wants to have the media server M play back
a demo tape into an existing conference. C indicates to the media
server that the network addresses and encryption keys are already
given by the conference, so they should not be chosen by the server.
The example omits the simple ACK responses.
C->M: DESCRIBE rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 1
Accept: application/sdp
M->C: RTSP/1.0 200 1 OK
Content-type: application/sdp
Content-Length: 44
v=0
o=- 2890844526 2890842807 IN IP4 192.16.24.202
s=RTSP Session
i=See above
t=0 0
m=audio 0 RTP/AVP 0
C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
port=7000-7001;ttl=127
Conference: [email protected]%20Starr
M->C: RTSP/1.0 200 OK
CSeq: 2
Transport: RTP/AVP;multicast;destination=225.219.201.15;
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port=7000-7001;ttl=127
Session: 91389234234
Conference: [email protected]%20Starr
C->M: PLAY rtsp://server.example.com/demo/548/sound RTSP/1.0
CSeq: 3
Session: 91389234234
M->C: RTSP/1.0 200 OK
CSeq: 3
14.6 Recording
The conference participant client C asks the media server M to record
the audio and video portions of a meeting. The client uses the
ANNOUNCE method to provide meta-information about the recorded
session to the server.
C->M: ANNOUNCE rtsp://server.example.com/meeting RTSP/1.0
CSeq: 90
Content-Type: application/sdp
Content-Length: 121
v=0
o=camera1 3080117314 3080118787 IN IP4 195.27.192.36
s=IETF Meeting, Munich - 1
i=The thirty-ninth IETF meeting will be held in Munich, Germany
u=http://www.ietf.org/meetings/Munich.html
e=IETF Channel 1 <[email protected]>
p=IETF Channel 1 +49-172-2312 451
c=IN IP4 224.0.1.11/127
t=3080271600 3080703600
a=tool:sdr v2.4a6
a=type:test
m=audio 21010 RTP/AVP 5
c=IN IP4 224.0.1.11/127
a=ptime:40
m=video 61010 RTP/AVP 31
c=IN IP4 224.0.1.12/127
M->C: RTSP/1.0 200 OK
CSeq: 90
C->M: SETUP rtsp://server.example.com/meeting/audiotrack RTSP/1.0
CSeq: 91
Transport: RTP/AVP;multicast;destination=224.0.1.11;
port=21010-21011;mode=record;ttl=127
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M->C: RTSP/1.0 200 OK
CSeq: 91
Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.11;
port=21010-21011;mode=record;ttl=127
C->M: SETUP rtsp://server.example.com/meeting/videotrack RTSP/1.0
CSeq: 92
Session: 50887676
Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127
M->C: RTSP/1.0 200 OK
CSeq: 92
Transport: RTP/AVP;multicast;destination=224.0.1.12;
port=61010-61011;mode=record;ttl=127
C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0
CSeq: 93
Session: 50887676
Range: clock=19961110T1925-19961110T2015
M->C: RTSP/1.0 200 OK
CSeq: 93
15 Syntax
The RTSP syntax is described in an augmented Backus-Naur form (BNF)
as used in RFC 2068 [2].
15.1 Base Syntax
OCTET = <any 8-bit sequence of data>
CHAR = <any US-ASCII character (octets 0 - 127)>
UPALPHA = <any US-ASCII uppercase letter "A".."Z">
LOALPHA = <any US-ASCII lowercase letter "a".."z">
ALPHA = UPALPHA | LOALPHA
DIGIT = <any US-ASCII digit "0".."9">
CTL = <any US-ASCII control character
(octets 0 - 31) and DEL (127)>
CR = <US-ASCII CR, carriage return (13)>
LF = <US-ASCII LF, linefeed (10)>
SP = <US-ASCII SP, space (32)>
HT = <US-ASCII HT, horizontal-tab (9)>
<"> = <US-ASCII double-quote mark (34)>
CRLF = CR LF
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LWS = [CRLF] 1*( SP | HT )
TEXT = <any OCTET except CTLs>
tspecials = "(" | ")" | "<" | ">" | "@"
| "," | ";" | ":" | "\" | <">
| "/" | "[" | "]" | "?" | "="
| "{" | "}" | SP | HT
token = 1*<any CHAR except CTLs or tspecials>
quoted-string = ( <"> *(qdtext) <"> )
qdtext = <any TEXT except <">>
quoted-pair = "\" CHAR
message-header = field-name ":" [ field-value ] CRLF
field-name = token
field-value = *( field-content | LWS )
field-content = <the OCTETs making up the field-value and
consisting of either *TEXT or
combinations of token, tspecials, and
quoted-string>
safe = "\$" | "-" | "_" | "." | "+"
extra = "!" | "*" | "$'$" | "(" | ")" | ","
hex = DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
"a" | "b" | "c" | "d" | "e" | "f"
escape = "\%" hex hex
reserved = ";" | "/" | "?" | ":" | "@" | "&" | "="
unreserved = alpha | digit | safe | extra
xchar = unreserved | reserved | escape
16 Security Considerations
Because of the similarity in syntax and usage between RTSP servers
and HTTP servers, the security considerations outlined in [H15]
apply. Specifically, please note the following:
Authentication Mechanisms:
RTSP and HTTP share common authentication schemes, and thus
should follow the same prescriptions with regards to
authentication. See [H15.1] for client authentication issues,
and [H15.2] for issues regarding support for multiple
authentication mechanisms.
Abuse of Server Log Information:
RTSP and HTTP servers will presumably have similar logging
mechanisms, and thus should be equally guarded in protecting
the contents of those logs, thus protecting the privacy of the
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users of the servers. See [H15.3] for HTTP server
recommendations regarding server logs.
Transfer of Sensitive Information:
There is no reason to believe that information transferred via
RTSP may be any less sensitive than that normally transmitted
via HTTP. Therefore, all of the precautions regarding the
protection of data privacy and user privacy apply to
implementors of RTSP clients, servers, and proxies. See
[H15.4] for further details.
Attacks Based On File and Path Names:
Though RTSP URLs are opaque handles that do not necessarily
have file system semantics, it is anticipated that many
implementations will translate portions of the request URLs
directly to file system calls. In such cases, file systems
SHOULD follow the precautions outlined in [H15.5], such as
checking for ".." in path components.
Personal Information:
RTSP clients are often privy to the same information that HTTP
clients are (user name, location, etc.) and thus should be
equally. See [H15.6] for further recommendations.
Privacy Issues Connected to Accept Headers:
Since may of the same "Accept" headers exist in RTSP as in
HTTP, the same caveats outlined in [H15.7] with regards to
their use should be followed.
DNS Spoofing:
Presumably, given the longer connection times typically
associated to RTSP sessions relative to HTTP sessions, RTSP
client DNS optimizations should be less prevalent.
Nonetheless, the recommendations provided in [H15.8] are still
relevant to any implementation which attempts to rely on a
DNS-to-IP mapping to hold beyond a single use of the mapping.
Location Headers and Spoofing:
If a single server supports multiple organizations that do not
trust one another, then it must check the values of Location
and Content-Location headers in responses that are generated
under control of said organizations to make sure that they do
not attempt to invalidate resources over which they have no
authority. ([H15.9])
In addition to the recommendations in the current HTTP specification
(RFC 2068 [2], as of this writing), future HTTP specifications may
provide additional guidance on security issues.
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The following are added considerations for RTSP implementations.
Concentrated denial-of-service attack:
The protocol offers the opportunity for a remote-controlled
denial-of-service attack. The attacker may initiate traffic
flows to one or more IP addresses by specifying them as the
destination in SETUP requests. While the attacker's IP address
may be known in this case, this is not always useful in
prevention of more attacks or ascertaining the attackers
identity. Thus, an RTSP server SHOULD only allow client-
specified destinations for RTSP-initiated traffic flows if the
server has verified the client's identity, either against a
database of known users using RTSP authentication mechanisms
(preferably digest authentication or stronger), or other
secure means.
Session hijacking:
Since there is no relation between a transport layer
connection and an RTSP session, it is possible for a malicious
client to issue requests with random session identifiers which
would affect unsuspecting clients. The server SHOULD use a
large, random and non-sequential session identifier to
minimize the possibility of this kind of attack.
Authentication:
Servers SHOULD implement both basic and digest [8]
authentication. In environments requiring tighter security for
the control messages, the RTSP control stream may be
encrypted.
Stream issues:
RTSP only provides for stream control. Stream delivery issues
are not covered in this section, nor in the rest of this memo.
RTSP implementations will most likely rely on other protocols
such as RTP, IP multicast, RSVP and IGMP, and should address
security considerations brought up in those and other
applicable specifications.
Persistently suspicious behavior:
RTSP servers SHOULD return error code 403 (Forbidden) upon
receiving a single instance of behavior which is deemed a
security risk. RTSP servers SHOULD also be aware of attempts
to probe the server for weaknesses and entry points and MAY
arbitrarily disconnect and ignore further requests clients
which are deemed to be in violation of local security policy.
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Appendix A: RTSP Protocol State Machines
The RTSP client and server state machines describe the behavior of
the protocol from RTSP session initialization through RTSP session
termination.
State is defined on a per object basis. An object is uniquely
identified by the stream URL and the RTSP session identifier. Any
request/reply using aggregate URLs denoting RTSP presentations
composed of multiple streams will have an effect on the individual
states of all the streams. For example, if the presentation /movie
contains two streams, /movie/audio and /movie/video, then the
following command:
PLAY rtsp://foo.com/movie RTSP/1.0
CSeq: 559
Session: 12345678
will have an effect on the states of movie/audio and movie/video.
This example does not imply a standard way to represent streams in
URLs or a relation to the filesystem. See Section 3.2.
The requests OPTIONS, ANNOUNCE, DESCRIBE, GET_PARAMETER,
SET_PARAMETER do not have any effect on client or server state and
are therefore not listed in the state tables.
A.1 Client State Machine
The client can assume the following states:
Init:
SETUP has been sent, waiting for reply.
Ready:
SETUP reply received or PAUSE reply received while in Playing
state.
Playing:
PLAY reply received
Recording:
RECORD reply received
In general, the client changes state on receipt of replies to
requests. Note that some requests are effective at a future time or
position (such as a PAUSE), and state also changes accordingly. If no
explicit SETUP is required for the object (for example, it is
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available via a multicast group), state begins at Ready. In this
case, there are only two states, Ready and Playing. The client also
changes state from Playing/Recording to Ready when the end of the
requested range is reached.
The "next state" column indicates the state assumed after receiving a
success response (2xx). If a request yields a status code of 3xx, the
state becomes Init, and a status code of 4xx yields no change in
state. Messages not listed for each state MUST NOT be issued by the
client in that state, with the exception of messages not affecting
state, as listed above. Receiving a REDIRECT from the server is
equivalent to receiving a 3xx redirect status from the server.
state message sent next state after response
Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
RECORD Recording
TEARDOWN Init
SETUP Ready
Playing PAUSE Ready
TEARDOWN Init
PLAY Playing
SETUP Playing (changed transport)
Recording PAUSE Ready
TEARDOWN Init
RECORD Recording
SETUP Recording (changed transport)
A.2 Server State Machine
The server can assume the following states:
Init:
The initial state, no valid SETUP has been received yet.
Ready:
Last SETUP received was successful, reply sent or after
playing, last PAUSE received was successful, reply sent.
Playing:
Last PLAY received was successful, reply sent. Data is being
sent.
Recording:
The server is recording media data.
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In general, the server changes state on receiving requests. If the
server is in state Playing or Recording and in unicast mode, it MAY
revert to Init and tear down the RTSP session if it has not received
"wellness" information, such as RTCP reports or RTSP commands, from
the client for a defined interval, with a default of one minute. The
server can declare another timeout value in the Session response
header (Section 12.37). If the server is in state Ready, it MAY
revert to Init if it does not receive an RTSP request for an interval
of more than one minute. Note that some requests (such as PAUSE) may
be effective at a future time or position, and server state changes
at the appropriate time. The server reverts from state Playing or
Recording to state Ready at the end of the range requested by the
client.
The REDIRECT message, when sent, is effective immediately unless it
has a Range header specifying when the redirect is effective. In such
a case, server state will also change at the appropriate time.
If no explicit SETUP is required for the object, the state starts at
Ready and there are only two states, Ready and Playing.
The "next state" column indicates the state assumed after sending a
success response (2xx). If a request results in a status code of 3xx,
the state becomes Init. A status code of 4xx results in no change.
state message received next state
Init SETUP Ready
TEARDOWN Init
Ready PLAY Playing
SETUP Ready
TEARDOWN Init
RECORD Recording
Playing PLAY Playing
PAUSE Ready
TEARDOWN Init
SETUP Playing
Recording RECORD Recording
PAUSE Ready
TEARDOWN Init
SETUP Recording
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Appendix B: Interaction with RTP
RTSP allows media clients to control selected, non-contiguous
sections of media presentations, rendering those streams with an RTP
media layer[24]. The media layer rendering the RTP stream should not
be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
timestamps MUST be continuous and monotonic across jumps of NPT.
As an example, assume a clock frequency of 8000 Hz, a packetization
interval of 100 ms and an initial sequence number and timestamp of
zero. First we play NPT 10 through 15, then skip ahead and play NPT
18 through 20. The first segment is presented as RTP packets with
sequence numbers 0 through 49 and timestamp 0 through 39,200. The
second segment consists of RTP packets with sequence number 50
through 69, with timestamps 40,000 through 55,200.
We cannot assume that the RTSP client can communicate with the RTP
media agent, as the two may be independent processes. If the RTP
timestamp shows the same gap as the NPT, the media agent will
assume that there is a pause in the presentation. If the jump in
NPT is large enough, the RTP timestamp may roll over and the media
agent may believe later packets to be duplicates of packets just
played out.
For certain datatypes, tight integration between the RTSP layer and
the RTP layer will be necessary. This by no means precludes the
above restriction. Combined RTSP/RTP media clients should use the
RTP-Info field to determine whether incoming RTP packets were sent
before or after a seek.
For continuous audio, the server SHOULD set the RTP marker bit at the
beginning of serving a new PLAY request. This allows the client to
perform playout delay adaptation.
For scaling (see Section 12.34), RTP timestamps should correspond to
the playback timing. For example, when playing video recorded at 30
frames/second at a scale of two and speed (Section 12.35) of one, the
server would drop every second frame to maintain and deliver video
packets with the normal timestamp spacing of 3,000 per frame, but NPT
would increase by 1/15 second for each video frame.
The client can maintain a correct display of NPT by noting the RTP
timestamp value of the first packet arriving after repositioning. The
sequence parameter of the RTP-Info (Section 12.33) header provides
the first sequence number of the next segment.
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Appendix C: Use of SDP for RTSP Session Descriptions
The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
describe streams or presentations in RTSP. Such usage is limited to
specifying means of access and encoding(s) for:
aggregate control:
A presentation composed of streams from one or more servers
that are not available for aggregate control. Such a
description is typically retrieved by HTTP or other non-RTSP
means. However, they may be received with ANNOUNCE methods.
non-aggregate control:
A presentation composed of multiple streams from a single
server that are available for aggregate control. Such a
description is typically returned in reply to a DESCRIBE
request on a URL, or received in an ANNOUNCE method.
This appendix describes how an SDP file, retrieved, for example,
through HTTP, determines the operation of an RTSP session. It also
describes how a client should interpret SDP content returned in reply
to a DESCRIBE request. SDP provides no mechanism by which a client
can distinguish, without human guidance, between several media
streams to be rendered simultaneously and a set of alternatives
(e.g., two audio streams spoken in different languages).
C.1 Definitions
The terms "session-level", "media-level" and other key/attribute
names and values used in this appendix are to be used as defined in
SDP (RFC 2327 [6]):
C.1.1 Control URL
The "a=control:" attribute is used to convey the control URL. This
attribute is used both for the session and media descriptions. If
used for individual media, it indicates the URL to be used for
controlling that particular media stream. If found at the session
level, the attribute indicates the URL for aggregate control.
Example:
a=control:rtsp://example.com/foo
This attribute may contain either relative and absolute URLs,
following the rules and conventions set out in RFC 1808 [25].
Implementations should look for a base URL in the following order:
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1. The RTSP Content-Base field
2. The RTSP Content-Location field
3. The RTSP request URL
If this attribute contains only an asterisk (*), then the URL is
treated as if it were an empty embedded URL, and thus inherits the
entire base URL.
C.1.2 Media streams
The "m=" field is used to enumerate the streams. It is expected that
all the specified streams will be rendered with appropriate
synchronization. If the session is unicast, the port number serves as
a recommendation from the server to the client; the client still has
to include it in its SETUP request and may ignore this
recommendation. If the server has no preference, it SHOULD set the
port number value to zero.
Example:
m=audio 0 RTP/AVP 31
C.1.3 Payload type(s)
The payload type(s) are specified in the "m=" field. In case the
payload type is a static payload type from RFC 1890 [1], no other
information is required. In case it is a dynamic payload type, the
media attribute "rtpmap" is used to specify what the media is. The
"encoding name" within the "rtpmap" attribute may be one of those
specified in RFC 1890 (Sections 5 and 6), or an experimental encoding
with a "X-" prefix as specified in SDP (RFC 2327 [6]). Codec-
specific parameters are not specified in this field, but rather in
the "fmtp" attribute described below. Implementors seeking to
register new encodings should follow the procedure in RFC 1890 [1].
If the media type is not suited to the RTP AV profile, then it is
recommended that a new profile be created and the appropriate profile
name be used in lieu of "RTP/AVP" in the "m=" field.
C.1.4 Format-specific parameters
Format-specific parameters are conveyed using the "fmtp" media
attribute. The syntax of the "fmtp" attribute is specific to the
encoding(s) that the attribute refers to. Note that the packetization
interval is conveyed using the "ptime" attribute.
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C.1.5 Range of presentation
The "a=range" attribute defines the total time range of the stored
session. (The length of live sessions can be deduced from the "t" and
"r" parameters.) Unless the presentation contains media streams of
different durations, the range attribute is a session-level
attribute. The unit is specified first, followed by the value range.
The units and their values are as defined in Section 3.5, 3.6 and
3.7.
Examples:
a=range:npt=0-34.4368
a=range:clock=19971113T2115-19971113T2203
C.1.6 Time of availability
The "t=" field MUST contain suitable values for the start and stop
times for both aggregate and non-aggregate stream control. With
aggregate control, the server SHOULD indicate a stop time value for
which it guarantees the description to be valid, and a start time
that is equal to or before the time at which the DESCRIBE request was
received. It MAY also indicate start and stop times of 0, meaning
that the session is always available. With non-aggregate control, the
values should reflect the actual period for which the session is
available in keeping with SDP semantics, and not depend on other
means (such as the life of the web page containing the description)
for this purpose.
C.1.7 Connection Information
In SDP, the "c=" field contains the destination address for the media
stream. However, for on-demand unicast streams and some multicast
streams, the destination address is specified by the client via the
SETUP request. Unless the media content has a fixed destination
address, the "c=" field is to be set to a suitable null value. For
addresses of type "IP4", this value is "0.0.0.0".
C.1.8 Entity Tag
The optional "a=etag" attribute identifies a version of the session
description. It is opaque to the client. SETUP requests may include
this identifier in the If-Match field (see section 12.22) to only
allow session establishment if this attribute value still corresponds
to that of the current description. The attribute value is opaque and
may contain any character allowed within SDP attribute values.
Example:
a=etag:158bb3e7c7fd62ce67f12b533f06b83a
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One could argue that the "o=" field provides identical
functionality. However, it does so in a manner that would put
constraints on servers that need to support multiple session
description types other than SDP for the same piece of media
content.
C.2 Aggregate Control Not Available
If a presentation does not support aggregate control and multiple
media sections are specified, each section MUST have the control URL
specified via the "a=control:" attribute.
Example:
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I came from a web page
t=0 0
c=IN IP4 0.0.0.0
m=video 8002 RTP/AVP 31
a=control:rtsp://audio.com/movie.aud
m=audio 8004 RTP/AVP 3
a=control:rtsp://video.com/movie.vid
Note that the position of the control URL in the description implies
that the client establishes separate RTSP control sessions to the
servers audio.com and video.com.
It is recommended that an SDP file contains the complete media
initialization information even if it is delivered to the media
client through non-RTSP means. This is necessary as there is no
mechanism to indicate that the client should request more detailed
media stream information via DESCRIBE.
C.3 Aggregate Control Available
In this scenario, the server has multiple streams that can be
controlled as a whole. In this case, there are both media-level
"a=control:" attributes, which are used to specify the stream URLs,
and a session-level "a=control:" attribute which is used as the
request URL for aggregate control. If the media-level URL is
relative, it is resolved to absolute URLs according to Section C.1.1
above.
If the presentation comprises only a single stream, the media-level
"a=control:" attribute may be omitted altogether. However, if the
presentation contains more than one stream, each media stream section
MUST contain its own "a=control" attribute.
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Example:
v=0
o=- 2890844256 2890842807 IN IP4 204.34.34.32
s=I contain
i=<more info>
t=0 0
c=IN IP4 0.0.0.0
a=control:rtsp://example.com/movie/
m=video 8002 RTP/AVP 31
a=control:trackID=1
m=audio 8004 RTP/AVP 3
a=control:trackID=2
In this example, the client is required to establish a single RTSP
session to the server, and uses the URLs
rtsp://example.com/movie/trackID=1 and
rtsp://example.com/movie/trackID=2 to set up the video and audio
streams, respectively. The URL rtsp://example.com/movie/ controls the
whole movie.
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Appendix D: Minimal RTSP implementation
D.1 Client
A client implementation MUST be able to do the following :
* Generate the following requests: SETUP, TEARDOWN, and one of PLAY
(i.e., a minimal playback client) or RECORD (i.e., a minimal
recording client). If RECORD is implemented, ANNOUNCE must be
implemented as well.
* Include the following headers in requests: CSeq, Connection,
Session, Transport. If ANNOUNCE is implemented, the capability to
include headers Content-Language, Content-Encoding, Content-
Length, and Content-Type should be as well.
* Parse and understand the following headers in responses: CSeq,
Connection, Session, Transport, Content-Language, Content-
Encoding, Content-Length, Content-Type. If RECORD is implemented,
the Location header must be understood as well. RTP-compliant
implementations should also implement RTP-Info.
* Understand the class of each error code received and notify the
end-user, if one is present, of error codes in classes 4xx and
5xx. The notification requirement may be relaxed if the end-user
explicitly does not want it for one or all status codes.
* Expect and respond to asynchronous requests from the server, such
as ANNOUNCE. This does not necessarily mean that it should
implement the ANNOUNCE method, merely that it MUST respond
positively or negatively to any request received from the server.
Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial
implementations and/or to be a "good citizen".
* Implement RTP/AVP/UDP as a valid transport.
* Inclusion of the User-Agent header.
* Understand SDP session descriptions as defined in Appendix C
* Accept media initialization formats (such as SDP) from standard
input, command line, or other means appropriate to the operating
environment to act as a "helper application" for other
applications (such as web browsers).
There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict
requirements.
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D.1.1 Basic Playback
To support on-demand playback of media streams, the client MUST
additionally be able to do the following:
* generate the PAUSE request;
* implement the REDIRECT method, and the Location header.
D.1.2 Authentication-enabled
In order to access media presentations from RTSP servers that require
authentication, the client MUST additionally be able to do the
following:
* recognize the 401 status code;
* parse and include the WWW-Authenticate header;
* implement Basic Authentication and Digest Authentication.
D.2 Server
A minimal server implementation MUST be able to do the following:
* Implement the following methods: SETUP, TEARDOWN, OPTIONS and
either PLAY (for a minimal playback server) or RECORD (for a
minimal recording server). If RECORD is implemented, ANNOUNCE
should be implemented as well.
* Include the following headers in responses: Connection,
Content-Length, Content-Type, Content-Language, Content-Encoding,
Transport, Public. The capability to include the Location header
should be implemented if the RECORD method is. RTP-compliant
implementations should also implement the RTP-Info field.
* Parse and respond appropriately to the following headers in
requests: Connection, Session, Transport, Require.
Though not required, the following are highly recommended at the time
of publication for practical interoperability with initial
implementations and/or to be a "good citizen".
* Implement RTP/AVP/UDP as a valid transport.
* Inclusion of the Server header.
* Implement the DESCRIBE method.
* Generate SDP session descriptions as defined in Appendix C
There may be RTSP applications different from those initially
envisioned by the contributors to the RTSP specification for which
the requirements above do not make sense. Therefore, the
recommendations above serve only as guidelines instead of strict
requirements.
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D.2.1 Basic Playback
To support on-demand playback of media streams, the server MUST
additionally be able to do the following:
* Recognize the Range header, and return an error if seeking is not
supported.
* Implement the PAUSE method.
In addition, in order to support commonly-accepted user interface
features, the following are highly recommended for on-demand media
servers:
* Include and parse the Range header, with NPT units.
Implementation of SMPTE units is recommended.
* Include the length of the media presentation in the media
initialization information.
* Include mappings from data-specific timestamps to NPT. When RTP
is used, the rtptime portion of the RTP-Info field may be used to
map RTP timestamps to NPT.
Client implementations may use the presence of length information
to determine if the clip is seekable, and visibly disable seeking
features for clips for which the length information is unavailable.
A common use of the presentation length is to implement a "slider
bar" which serves as both a progress indicator and a timeline
positioning tool.
Mappings from RTP timestamps to NPT are necessary to ensure correct
positioning of the slider bar.
D.2.2 Authentication-enabled
In order to correctly handle client authentication, the server MUST
additionally be able to do the following:
* Generate the 401 status code when authentication is required for
the resource.
* Parse and include the WWW-Authenticate header
* Implement Basic Authentication and Digest Authentication
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Appendix E: Authors' Addresses
Henning Schulzrinne
Dept. of Computer Science
Columbia University
1214 Amsterdam Avenue
New York, NY 10027
USA
EMail: [email protected]
Anup Rao
Netscape Communications Corp.
501 E. Middlefield Road
Mountain View, CA 94043
USA
EMail: [email protected]
Robert Lanphier
RealNetworks
1111 Third Avenue Suite 2900
Seattle, WA 98101
USA
EMail: [email protected]
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Appendix F: Acknowledgements
This memo is based on the functionality of the original RTSP document
submitted in October 96. It also borrows format and descriptions from
HTTP/1.1.
This document has benefited greatly from the comments of all those
participating in the MMUSIC-WG. In addition to those already
mentioned, the following individuals have contributed to this
specification:
Rahul Agarwal, Torsten Braun, Brent Browning, Bruce Butterfield,
Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Dunsmuir,
Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter
Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka,
Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan
Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Papadopouli,
Sujal Patel, Ema Patki, Alagu Periyannan, Igor Plotnikov, Pinaki
Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and
John Francis Stracke.
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References
1 Schulzrinne, H., "RTP profile for audio and video conferences
with minimal control", RFC 1890, January 1996.
2 Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
Berners-Lee, "Hypertext transfer protocol - HTTP/1.1", RFC
2068, January 1997.
3 Yergeau, F., Nicol, G., Adams, G., and M. Duerst,
"Internationalization of the hypertext markup language", RFC
2070, January 1997.
4 Bradner, S., "Key words for use in RFCs to indicate
requirement levels", BCP 14, RFC 2119, March 1997.
5 ISO/IEC, "Information technology - generic coding of moving
pictures and associated audio information - 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.
6 Handley, M., and V. Jacobson, "SDP: Session Description
Protocol", RFC 2327, April 1998.
7 Franks, J., Hallam-Baker, P., and J. Hostetler, "An extension to
HTTP: digest access authentication", RFC 2069, January 1997.
8 Postel, J., "User Datagram Protocol", STD 6, RFC 768, August
1980.
9 Hinden, B. and C. Partridge, "Version 2 of the reliable data
protocol (RDP)", RFC 1151, April 1990.
10 Postel, J., "Transmission control protocol", STD 7, RFC 793,
September 1981.
11 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.
12 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.
Schulzrinne, et. al. Standards Track [Page 90]
RFC 2326 Real Time Streaming Protocol April 1998
13 McMahon, P., "GSS-API authentication method for SOCKS version
5", RFC 1961, June 1996.
14 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.
15 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.
16 Crocker, D. and P. Overell, "Augmented BNF for syntax
specifications: ABNF", RFC 2234, November 1997.
17 Braden, B., "Requirements for internet hosts - application and
support", STD 3, RFC 1123, October 1989.
18 Elz, R., "A compact representation of IPv6 addresses", RFC
1924, April 1996.
19 Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform
resource locators (URL)", RFC 1738, December 1994.
20 Yergeau, F., "UTF-8, a transformation format of ISO 10646",
RFC 2279, January 1998.
22 Braden, B., "T/TCP - TCP extensions for transactions
functional specification", RFC 1644, July 1994.
22 W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
Reading, Massachusetts: Addison-Wesley, 1994.
23 Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
"RTP: a transport protocol for real-time applications", RFC
1889, January 1996.
24 Fielding, R., "Relative uniform resource locators", RFC 1808,
June 1995.
Schulzrinne, et. al. Standards Track [Page 91]
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Full Copyright Statement
Copyright (C) The Internet Society (1998). All Rights Reserved.
This document and translations of it may be copied and furnished to
others, and derivative works that comment on or otherwise explain it
or assist in its implementation may be prepared, copied, published
and distributed, in whole or in part, without restriction of any
kind, provided that the above copyright notice and this paragraph are
included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
the copyright notice or references to the Internet Society or other
Internet organizations, except as needed for the purpose of
developing Internet standards in which case the procedures for
copyrights defined in the Internet Standards process must be
followed, or as required to translate it into languages other than
English.
The limited permissions granted above are perpetual and will not be
revoked by the Internet Society or its successors or assigns.
This document and the information contained herein is provided on an
"AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING
TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Schulzrinne, et. al. Standards Track [Page 92]
