Schulzrinne, et. al. Standards Track [Page 29]RFC 2326 Real Time Streaming Protocol April 1998 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).Schulzrinne, et. al. Standards Track [Page 30]RFC 2326 Real Time Streaming Protocol April 199810.2 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 e=mjh@isi.edu (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 DESCRIBESchulzrinne, et. al. Standards Track [Page 31]RFC 2326 Real Time Streaming Protocol April 1998 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 p_w_upload, 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 126.16.64.4Schulzrinne, et. al. Standards Track [Page 32]RFC 2326 Real Time Streaming Protocol April 1998 s=SDP Seminar i=A Seminar on the session description protocol u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps e=mjh@isi.edu (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: 31210.4 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 theSchulzrinne, et. al. Standards Track [Page 33]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 34]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 35]RFC 2326 Real Time Streaming Protocol April 199810.6 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 firstSchulzrinne, et. al. Standards Track [Page 36]RFC 2326 Real Time Streaming Protocol April 1998 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: 89210.8 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 jitterSchulzrinne, et. al. Standards Track [Page 37]RFC 2326 Real Time Streaming Protocol April 1998 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 ParameterSchulzrinne, et. al. Standards Track [Page 38]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 39]RFC 2326 Real Time Streaming Protocol April 1998 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: 128.16.64.19/3249237410.12 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=0-1Schulzrinne, et. al. Standards Track [Page 40]RFC 2326 Real Time Streaming Protocol April 1998 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 2xx11.1.1 250 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.Schulzrinne, et. al. Standards Track [Page 41]RFC 2326 Real Time Streaming Protocol April 199811.3 Client Error 4xx11.3.1 405 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.Schulzrinne, et. al. Standards Track [Page 42]RFC 2326 Real Time Streaming Protocol April 199811.3.8 457 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.Schulzrinne, et. al. Standards Track [Page 43]RFC 2326 Real Time Streaming Protocol April 199812 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.Schulzrinne, et. al. Standards Track [Page 44]RFC 2326 Real Time Streaming Protocol April 1998 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. allSchulzrinne, et. al. Standards Track [Page 45]RFC 2326 Real Time Streaming Protocol April 1998 Overview of RTSP header fields12.1 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=212.2 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_PARAMETER12.5 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.Schulzrinne, et. al. Standards Track [Page 46]RFC 2326 Real Time Streaming Protocol April 1998 Bandwidth = "Bandwidth" ":" 1*DIGIT Example: Bandwidth: 400012.7 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"Schulzrinne, et. al. Standards Track [Page 47]RFC 2326 Real Time Streaming Protocol April 1998 | "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.Schulzrinne, et. al. Standards Track [Page 48]RFC 2326 Real Time Streaming Protocol April 1998 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: 199702170042.SAA08642@obiwan.arl.wustl.edu%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]Schulzrinne, et. al. Standards Track [Page 49]RFC 2326 Real Time Streaming Protocol April 199812.13 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.Schulzrinne, et. al. Standards Track [Page 50]RFC 2326 Real Time Streaming Protocol April 1998 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].Schulzrinne, et. al. Standards Track [Page 51]RFC 2326 Real Time Streaming Protocol April 1998 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 GMT12.24 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. ServersSchulzrinne, et. al. Standards Track [Page 52]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 53]RFC 2326 Real Time Streaming Protocol April 199812.30 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.Schulzrinne, et. al. Standards Track [Page 54]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 55]RFC 2326 Real Time Streaming Protocol April 1998 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=3021112.34 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: -3.5Schulzrinne, et. al. Standards Track [Page 56]RFC 2326 Real Time Streaming Protocol April 199812.35 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 inSchulzrinne, et. al. Standards Track [Page 57]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 58]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 59]RFC 2326 Real Time Streaming Protocol April 1998 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.Schulzrinne, et. al. Standards Track [Page 60]RFC 2326 Real Time Streaming Protocol April 1998 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. ***c: The ***c parameter indicates the RTP ×××C [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 ] | ";" "***c" "=" ***c | ";" "mode" = <"> 1\#mode <"> ttl = 1*3(DIGIT) port = 1*5(DIGIT) ***c = 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.Schulzrinne, et. al. Standards Track [Page 61]RFC 2326 Real Time Streaming Protocol April 199812.40 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 directivesSchulzrinne, et. al. Standards Track [Page 62]RFC 2326 Real Time Streaming Protocol April 1998 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/sdpSchulzrinne, et. al. Standards Track [Page 63]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: 12345678Schulzrinne, 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.