Internet Engineering Task Force MMUSIC WG Internet Draft H. Schulzrinne, A. Rao, R. Lanphier ietf-mmusic-rtsp-rev-00.txt Columbia U./Cisco/RealNetworks May 28, 1999 Expires: November, 1999 Real Time Streaming Protocol (RTSP) STATUS OF THIS MEMO This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress". To view the list Internet-Draft Shadow Directories, see http://www.ietf.org/shadow.html. Abstract The Real Time Streaming Protocol, or RTSP, is an application-level pro- tocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 1889). 1 Introduction 1.1 Purpose The Real-Time Streaming Protocol (RTSP) establishes and controls either a single or several time-synchronized streams of continuous media such as audio and video. It does not typically deliver the continuous streams itself, although interleaving of the continuous media stream with the control stream is possible (see Section 10.12). In other words, RTSP H. Schulzrinne, A. Rao, R. Lanphier [Page 1] Internet Draft RTSP May 28, 1999 acts as a "network remote control" for multimedia servers. The set of streams to be controlled is defined by a presentation description. This memorandum does not define a format for a presentation description. There is no notion of an RTSP connection; instead, a server maintains a session labeled by an identifier. An RTSP session is in no way tied to a transport-level connection such as a TCP connection. During an RTSP ses- sion, an RTSP client may open and close many reliable transport connec- tions to the server to issue RTSP requests. Alternatively, it may use a connectionless transport protocol such as UDP. The streams controlled by RTSP may use RTP [1], but the operation of RTSP does not depend on the transport mechanism used to carry continuous media. The protocol is intentionally similar in syntax and operation to HTTP/1.1 [2] so that extension mechanisms to HTTP can in most cases also be added to RTSP. However, RTSP differs in a number of important aspects from HTTP: + RTSP introduces a number of new methods and has a different pro- tocol identifier. + An RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. + Both an RTSP server and client can issue requests. + Data is carried out-of-band by a different protocol. (There is an exception to this.) + RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, consistent with current HTML internationalization efforts [3]. + The Request-URI always contains the absolute URI. Because of backward compatibility with a historical blunder, HTTP/1.1 [2] carries only the absolute path in the request and puts the host name in a separate header field. This makes "virtual hosting" easier, where a single host with one IP address hosts several document trees. The protocol supports the following operations: H. Schulzrinne, A. Rao, R. Lanphier [Page 2] Internet Draft RTSP May 28, 1999 Retrieval of media from media server: The client can request a pre- sentation description via HTTP or some other method. If the presentation is being multicast, the presentation description contains the multicast addresses and ports to be used for the continuous media. If the presentation is to be sent only to the client via unicast, the client provides the destination for security reasons. Invitation of a media server to a conference: A media server can be "invited" to join an existing conference, either to play back media into the presentation or to record all or a subset of the media in a presentation. This mode is useful for dis- tributed teaching applications. Several parties in the confer- ence may take turns "pushing the remote control buttons". Addition of media to an existing presentation: Particularly for live presentations, it is useful if the server can tell the client about additional media becoming available. RTSP requests may be handled by proxies, tunnels and caches as in HTTP/1.1 [2]. 1.2 Requirements The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [4]. 1.3 Terminology Some of the terminology has been adopted from HTTP/1.1 [2]. Terms not listed here are defined as in HTTP/1.1. Aggregate control: The control of the multiple streams using a sin- gle timeline by the server. For audio/video feeds, this means that the client may issue a single play or pause message to control both the audio and video feeds. Conference: a multiparty, multimedia presentation, where "multi" implies greater than or equal to one. Client: The client requests continuous media data from the media server. Connection: A transport layer virtual circuit established between two programs for the purpose of communication. H. Schulzrinne, A. Rao, R. Lanphier [Page 3] Internet Draft RTSP May 28, 1999 Container file: A file which may contain multiple media streams which often comprise a presentation when played together. RTSP servers may offer aggregate control on these files, though the concept of a container file is not embedded in the protocol. Continuous media: Data where there is a timing relationship between source and sink; that is, the sink must reproduce the timing relationship that existed at the source. The most common exam- ples of continuous media are audio and motion video. Continu- ous media can be real-time (interactive) , where there is a "tight" timing relationship between source and sink, or streaming (playback) , where the relationship is less strict. Entity: The information transferred as the payload of a request or response. An entity consists of metainformation in the form of entity-header fields and content in the form of an entity- body, as described in Section 8. Media initialization: Datatype/codec specific initialization. This includes such things as clockrates, color tables, etc. Any transport-independent information which is required by a client for playback of a media stream occurs in the media ini- tialization phase of stream setup. Media parameter: Parameter specific to a media type that may be changed before or during stream playback. Media server: The server providing playback or recording services for one or more media streams. Different media streams within a presentation may originate from different media servers. A media server may reside on the same or a different host as the web server the presentation is invoked from. Media server indirection: Redirection of a media client to a dif- ferent media server. (Media) stream: A single media instance, e.g., an audio stream or a video stream as well as a single whiteboard or shared applica- tion group. When using RTP, a stream consists of all RTP and RTCP packets created by a source within an RTP session. This is equivalent to the definition of a DSM-CC stream([5]). Message: The basic unit of RTSP communication, consisting of a structured sequence of octets matching the syntax defined in Section 15 and transmitted via a connection or a connection- less protocol. H. Schulzrinne, A. Rao, R. Lanphier [Page 4] Internet Draft RTSP May 28, 1999 Participant: Member of a conference. A participant may be a machine, e.g., a media record or playback server. Presentation: A set of one or more streams presented to the client as a complete media feed, using a presentation description as defined below. In most cases in the RTSP context, this implies aggregate control of those streams, but does not have to. Presentation description: A presentation description contains information about one or more media streams within a presenta- tion, such as the set of encodings, network addresses and information about the content. Other IETF protocols such as SDP (RFC 2327 [6]) use the term "session" for a live presenta- tion. The presentation description may take several different formats, including but not limited to the session description format SDP. Response: An RTSP response. If an HTTP response is meant, that is indicated explicitly. Request: An RTSP request. If an HTTP request is meant, that is indicated explicitly. RTSP session: A complete RTSP "transaction", e.g., the viewing of a movie. A session typically consists of a client setting up a transport mechanism for the continuous media stream ( SETUP), starting the stream with PLAY or RECORD, and closing the stream with TEARDOWN. Transport initialization: The negotiation of transport information (e.g., port numbers, transport protocols) between the client and the server. 1.4 Protocol Properties RTSP has the following properties: Extendable: New methods and parameters can be easily added to RTSP. Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers. Secure: RTSP re-uses web security mechanisms, either at the trans- port level (TLS, RFC 2246 ) or within the protocol itself. All HTTP authentication mechanisms such as basic (RFC 2068 [2]) and digest authentication (RFC 2069 [8]) are directly applica- ble. H. Schulzrinne, A. Rao, R. Lanphier [Page 5] Internet Draft RTSP May 28, 1999 Transport-independent: RTSP may use either an unreliable datagram protocol (UDP) (RFC 768 [9]), a reliable datagram protocol (RDP, RFC 1151, not widely used [10]) or a reliable stream protocol such as TCP (RFC 793 [11]) as it implements applica- tion-level reliability. Multi-server capable: Each media stream within a presentation can reside on a different server. The client automatically estab- lishes several concurrent control sessions with the different media servers. Media synchronization is performed at the transport level. Control of recording devices: The protocol can control both record- ing and playback devices, as well as devices that can alter- nate between the two modes ("VCR"). Separation of stream control and conference initiation: Stream con- trol is divorced from inviting a media server to a conference. The only requirement is that the conference initiation proto- col either provides or can be used to create a unique confer- ence identifier. In particular, SIP [12] or H.323 [13] may be used to invite a server to a conference. Suitable for professional applications: RTSP supports frame-level accuracy through SMPTE time stamps to allow remote digital editing. Presentation description neutral: The protocol does not impose a particular presentation description or metafile format and can convey the type of format to be used. However, the presenta- tion description must contain at least one RTSP URI. Proxy and firewall friendly: The protocol should be readily handled by both application and transport-layer (SOCKS [14]) fire- walls. A firewall may need to understand the SETUP method to open a "hole" for the UDP media stream. HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so that the existing infrastructure can be reused. This infrastructure includes PICS (Platform for Internet Content Selection [15,16]) for associating labels with content. However, RTSP does not just add methods to HTTP since the controlling con- tinuous media requires server state in most cases. Appropriate server control: If a client can start a stream, it must be able to stop a stream. Servers should not start streaming to clients in such a way that clients cannot stop the stream. H. Schulzrinne, A. Rao, R. Lanphier [Page 6] Internet Draft RTSP May 28, 1999 Transport negotiation: The client can negotiate the transport method prior to actually needing to process a continuous media stream. Capability negotiation: If basic features are disabled, there must be some clean mechanism for the client to determine which methods are not going to be implemented. This allows clients to present the appropriate user interface. For example, if seeking is not allowed, the user interface must be able to disallow moving a sliding position indicator. An earlier requirement in RTSP was multi-client capability. However, it was determined that a better approach was to make sure that the protocol is easily extensible to the multi- client scenario. Stream identifiers can be used by several control streams, so that "passing the remote" would be possi- ble. The protocol would not address how several clients nego- tiate access; this is left to either a "social protocol" or some other floor control mechanism. 1.5 Extending RTSP Since not all media servers have the same functionality, media servers by necessity will support different sets of requests. For example: + A server may only be capable of playback thus has no need to sup- port the RECORD request. + A server may not be capable of seeking (absolute positioning) if it is to support live events only. + Some servers may not support setting stream parameters and thus not support GET_PARAMETER and SET_PARAMETER. A server SHOULD implement all header fields described in Section 12. It is up to the creators of presentation descriptions not to ask the impossible of a server. This situation is similar in HTTP/1.1 [2], where the methods described in [H19.6] are not likely to be supported across all servers. RTSP can be extended in three ways, listed here in order of the magni- tude of changes supported: + Existing methods can be extended with new parameters, as long as these parameters can be safely ignored by the recipient. (This is equivalent to adding new parameters to an HTML tag.) If the H. Schulzrinne, A. Rao, R. Lanphier [Page 7] Internet Draft RTSP May 28, 1999 client needs negative acknowledgement when a method extension is not supported, a tag corresponding to the extension may be added in the Require: field (see Section 12.33). + New methods can be added. If the recipient of the message does not understand the request, it responds with error code 501 (Not Implemented) and the sender should not attempt to use this method again. A client may also use the OPTIONS method to inquire about methods supported by the server. The server SHOULD list the methods it supports using the Public response header. + A new version of the protocol can be defined, allowing almost all aspects (except the position of the protocol version number) to change. 1.6 Overall Operation Each presentation and media stream may be identified by an RTSP URL. The overall presentation and the properties of the media the presenta- tion is made up of are defined by a presentation description file, the format of which is outside the scope of this specification. The presen- tation description file may be obtained by the client using HTTP or other means such as email and may not necessarily be stored on the media server. For the purposes of this specification, a presentation description is assumed to describe one or more presentations, each of which maintains a common time axis. For simplicity of exposition and without loss of gen- erality, it is assumed that the presentation description contains exactly one such presentation. A presentation may contain several media streams. The presentation description file contains a description of the media streams making up the presentation, including their encodings, language, and other parameters that enable the client to choose the most appropri- ate combination of media. In this presentation description, each media stream that is individually controllable by RTSP is identified by an RTSP URL, which points to the media server handling that particular media stream and names the stream stored on that server. Several media streams can be located on different servers; for example, audio and video streams can be split across servers for load sharing. The description also enumerates which transport methods the server is capa- ble of. Besides the media parameters, the network destination address and port need to be determined. Several modes of operation can be distinguished: H. Schulzrinne, A. Rao, R. Lanphier [Page 8] Internet Draft RTSP May 28, 1999 Unicast: The media is transmitted to the source of the RTSP request, with the port number chosen by the client. Alterna- tively, the media is transmitted on the same reliable stream as RTSP. Multicast, server chooses address: The media server picks the mul- ticast address and port. This is the typical case for a live or near-media-on-demand transmission. Multicast, client chooses address: If the server is to participate in an existing multicast conference, the multicast address, port and encryption key are given by the conference descrip- tion, established by means outside the scope of this specifi- cation. 1.7 RTSP States RTSP controls a stream which may be sent via a separate protocol, inde- pendent of the control channel. For example, RTSP control may occur on a TCP connection while the data flows via UDP. Thus, data delivery contin- ues even if no RTSP requests are received by the media server. Also, during its lifetime, a single media stream may be controlled by RTSP requests issued sequentially on different TCP connections. Therefore, the server needs to maintain "session state" to be able to correlate RTSP requests with a stream. The state transitions are described in Sec- tion A. Many methods in RTSP do not contribute to state. However, the following play a central role in defining the allocation and usage of stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and TEARDOWN. SETUP: Causes the server to allocate resources for a stream and start an RTSP session. PLAY and RECORD: Starts data transmission on a stream allocated via SETUP. PAUSE: Temporarily halts a stream without freeing server resources. TEARDOWN: Frees resources associated with the stream. The RTSP session ceases to exist on the server. RTSP methods that contribute to state use the Session header field (Section 12.38) to identify the RTSP session whose state is being manipulated. The server generates session identifiers in response to SETUP requests (Section 10.4). H. Schulzrinne, A. Rao, R. Lanphier [Page 9] Internet Draft RTSP May 28, 1999 1.8 Relationship with Other Protocols RTSP has some overlap in functionality with HTTP. It also may interact with HTTP in that the initial contact with streaming content is often to be made through a web page. The current protocol specification aims to allow different hand-off points between a web server and the media server implementing RTSP. For example, the presentation description can be retrieved using HTTP or RTSP, which reduces roundtrips in web- browser-based scenarios, yet also allows for standalone RTSP servers and clients which do not rely on HTTP at all. However, RTSP differs fundamentally from HTTP in that data delivery takes place out-of-band in a different protocol. HTTP is an asymmetric protocol where the client issues requests and the server responds. In RTSP, both the media client and media server can issue requests. RTSP requests are also not stateless; they may set parameters and continue to control a media stream long after the request has been acknowledged. Re-using HTTP functionality has advantages in at least two areas, namely security and proxies. The requirements are very similar, so having the ability to adopt HTTP work on caches, proxies and authentication is valuable. While most real-time media will use RTP as a transport protocol, RTSP is not tied to RTP. RTSP assumes the existence of a presentation description format that can express both static and temporal properties of a presentation containing several media streams. 2 Notational Conventions Since many of the definitions and syntax are identical to HTTP/1.1, this specification only points to the section where they are defined rather than copying it. For brevity, [HX.Y] is to be taken to refer to Section X.Y of the current HTTP/1.1 specification (RFC 2068 [2]). All the mechanisms specified in this document are described in both prose and an augmented Backus-Naur form (BNF) similar to that used in [H2.1]. It is described in detail in RFC 2234 [17], with the difference that this RTSP specification maintains the "1#" notation for comma-sepa- rated lists. In this draft, we use indented and smaller-type paragraphs to provide background and motivation. This is intended to give readers who were not involved with the formulation of the specification an understanding of why things are the way that they are in RTSP. H. Schulzrinne, A. Rao, R. Lanphier [Page 10] Internet Draft RTSP May 28, 1999 3 Protocol Parameters 3.1 RTSP Version applies, with HTTP replaced by RTSP. 3.2 RTSP URL The "rtsp", "rtspu" and "rtsps" schemes are used to refer to network resources via the RTSP protocol. This section defines the scheme-spe- cific syntax and semantics for RTSP URLs. rtsp_URL--- ( "rtsp:" | "rtspu:" | "rtsps:" ) "//" host [ ":" port ] [ abs_path ] host --- port --- *DIGIT abs_path is defined in [H3.2.1]. Note that fragment and query identifiers do not have a well- defined meaning at this time, with the interpretation left to the RTSP server. The scheme rtsp requires that commands are issued via a reliable proto- col (within the Internet, TCP), while the scheme rtspu identifies an unreliable protocol (within the Internet, UDP). The scheme rtsps indi- cates that a TCP connection secured by TLS (RFC 2246) must be used. If the port is empty or not given, port 554 is assumed. The semantics are that the identified resource can be controlled by RTSP at the server listening for TCP (scheme "rtsp") connections or UDP (scheme "rtspu") packets on that port of host, and the Request-URI for the resource is rtsp_URL. The use of IP addresses in URLs SHOULD be avoided whenever possible (see RFC 1924 [19]). A presentation or a stream is identified by a textual media identifier, using the character set and escape conventions [H3.2] of URLs (RFC 1738 [20]). URLs may refer to a stream or an aggregate of streams, i.e., a presentation. Accordingly, requests described in Section 10 can apply to either the whole presentation or an individual stream within the presen- tation. Note that some request methods can only be applied to streams, H. Schulzrinne, A. Rao, R. Lanphier [Page 11] Internet Draft RTSP May 28, 1999 not presentations and vice versa. For example, the RTSP URL: rtsp://media.example.com:554/twister/audiotrack identifies the audio stream within the presentation "twister", which can be controlled via RTSP requests issued over a TCP connection to port 554 of host media.example.com Also, the RTSP URL: rtsp://media.example.com:554/twister identifies the presentation "twister", which may be composed of audio and video streams. This does not imply a standard way to reference streams in URLs. The presentation description defines the hierarchical relationships in the presentation and the URLs for the indi- vidual streams. A presentation description may name a stream "a.mov" and the whole presentation "b.mov". The path components of the RTSP URL are opaque to the client and do not imply any particular file system structure for the server. This decoupling also allows presentation descriptions to be used with non-RTSP media control protocols simply by replacing the scheme in the URL. 3.3 Conference Identifiers Conference identifiers are opaque to RTSP and are encoded using standard URI encoding methods (i.e., LWS is escaped with %). They can contain any octet value. The conference identifier MUST be globally unique. For H.323, the conferenceID value is to be used. conference-id--- 1*xchar Conference identifiers are used to allow RTSP sessions to obtain parameters from multimedia conferences the media server H. Schulzrinne, A. Rao, R. Lanphier [Page 12] Internet Draft RTSP May 28, 1999 is participating in. These conferences are created by proto- cols outside the scope of this specification, e.g., H.323 [13] or SIP [12]. Instead of the RTSP client explicitly providing transport information, for example, it asks the media server to use the values in the conference description instead. 3.4 Session Identifiers Session identifiers are opaque strings of arbitrary length. Linear white space must be URL-escaped. A session identifier MUST be chosen randomly and MUST be at least eight octets long to make guessing it more diffi- cult. (See Section 16.) session-id--- 8*( ALPHA | DIGIT | safe ) 3.5 SMPTE Relative Timestamps A SMPTE relative timestamp expresses time relative to the start of the clip. Relative timestamps are expressed as SMPTE time codes for frame- level access accuracy. The time code has the format hours:minutes:seconds:frames.subframes , with the origin at the start of the clip. The default smpte format is"SMPTE 30 drop" format, with frame rate is 29.97 frames per second. Other SMPTE codes MAY be supported (such as "SMPTE 25") through the use of alternative use of "smpte time". For the "frames" field in the time value can assume the values 0 through 29. The difference between 30 and 29.97 frames per second is handled by dropping the first two frame indices (values 00 and 01) of every minute, except every tenth minute. If the frame value is zero, it may be omitted. Subframes are measured in one-hundredth of a frame. smpte-range --- smpte-type "=" smpte-time "-" [ smpte-time ] smpte-type --- "smpte" | "smpte-30-drop" | "smpte-25" ; other timecodes may be added smpte-time --- 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ ":" 1*2DIGIT ] [ "." 1*2DIGIT ] Examples: smpte=10:12:33:20- smpte=10:07:33- smpte=10:07:00-10:07:33:05.01 smpte-25=10:07:00-10:07:33:05.01 H. Schulzrinne, A. Rao, R. Lanphier [Page 13] Internet Draft RTSP May 28, 1999 3.6 Normal Play Time Normal play time (NPT) indicates the stream absolute position relative to the beginning of the presentation. The timestamp consists of a deci- mal fraction. The part left of the decimal may be expressed in either seconds or hours, minutes, and seconds. The part right of the decimal point measures fractions of a second. The beginning of a presentation corresponds to 0.0 seconds. Negative values are not defined. The special constant now is defined as the cur- rent instant of a live event. It may be used only for live events. NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the viewer associates with a program. It is often digitally displayed on a VCR. NPT advances normally when in normal play mode (scale = 1), advances at a faster rate when in fast scan forward (high positive scale ratio), decrements when in scan reverse (high negative scale ratio) and is fixed in pause mode. NPT is (logically) equivalent to SMPTE time codes." [5] npt-range --- ( npt-time "-" [ npt-time ] ) | ( "-" npt-time ) npt-time --- "now" | npt-sec | npt-hhmmss npt-sec --- 1*DIGIT [ "." *DIGIT ] npt-hhmmss--- npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ] npt-hh --- 1*DIGIT ; any positive number npt-mm --- 1*2DIGIT ; 0-59 npt-ss --- 1*2DIGIT ; 0-59 Examples: npt=123.45-125 npt=12:05:35.3- npt=now- The syntax conforms to ISO 8601. The npt-sec notation is opti- mized for automatic generation, the ntp-hhmmss notation for consumption by human readers. The "now" constant allows clients to request to receive the live feed rather than the stored or time-delayed version. This is needed since neither absolute time nor zero time are appropriate for this case. 3.7 Absolute Time Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Fractions of a second may be indicated. H. Schulzrinne, A. Rao, R. Lanphier [Page 14] Internet Draft RTSP May 28, 1999 utc-range--- "clock" "=" utc-time "-" [ utc-time ] utc-time --- utc-date "T" utc-time "Z" utc-date --- 8DIGIT ; < YYYYMMDD > utc-time --- 6DIGIT [ "." fraction ] ; < HHMMSS.fraction > Example for November 8, 1996 at 14h37 and 20 and a quarter seconds UTC: 19961108T143720.25Z 3.8 Option Tags Option tags are unique identifiers used to designate new options in RTSP. These tags are used in in Require (Section 12.33) and Proxy- Require (Section 12.28) header fields. Syntax: option-tag--- token The creator of a new RTSP option should either prefix the option with a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name for a feature whose inventor can be reached at "foo.com"), or register the new option with the Internet Assigned Numbers Authority (IANA). 3.8.1 Registering New Option Tags with IANA When registering a new RTSP option, the following information should be provided: + Name and description of option. The name may be of any length, but SHOULD be no more than twenty characters long. The name MUST not contain any spaces, control characters or periods. + Indication of who has change control over the option (for exam- ple, IETF, ISO, ITU-T, other international standardization bod- ies, a consortium or a particular company or group of companies); + A reference to a further description, if available, for example (in order of preference) an RFC, a published paper, a patent fil- ing, a technical report, documented source code or a computer manual; + For proprietary options, contact information (postal and email address); H. Schulzrinne, A. Rao, R. Lanphier [Page 15] Internet Draft RTSP May 28, 1999 4 RTSP Message RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2279 [21]). Lines are terminated by CRLF, but receivers should be prepared to also interpret CR and LF by themselves as line terminators. Text-based protocols make it easier to add optional parameters in a self-describing manner. Since the number of parameters and the frequency of commands is low, processing efficiency is not a concern. Text-based protocols, if done carefully, also allow easy implementation of research prototypes in scripting languages such as Tcl, Visual Basic and Perl. The 10646 character set avoids tricky character set switching, but is invisible to the application as long as US-ASCII is being used. This is also the encoding used for RTCP. ISO 8859-1 translates directly into Unicode with a high-order octet of zero. ISO 8859-1 characters with the most-significant bit set are represented as 1100001x 10xxxxxx. (See RFC 2279 [21]) RTSP messages can be carried over any lower-layer transport protocol that is 8-bit clean. Requests contain methods, the object the method is operating upon and parameters to further describe the method. Methods are idempotent, unless otherwise noted. Methods are also designed to require little or no state maintenance at the media server. 4.1 Message Types See [H4.1] 4.2 Message Headers See [H4.2] 4.3 Message Body See [H4.3] 4.4 Message Length When a message body is included with a message, the length of that body is determined by one of the following (in order of precedence): H. Schulzrinne, A. Rao, R. Lanphier [Page 16] Internet Draft RTSP May 28, 1999 1. Any response message which MUST NOT include a message body (such as the 1xx, 204, and 304 responses) is always terminated by the first empty line after the header fields, regardless of the entity-header fields present in the message. (Note: An empty line consists of only CRLF.) 2. If a Content-Length header field (section 12.15) is present, its value in bytes represents the length of the message-body. If this header field is not present, a value of zero is assumed. 3. By the server closing the connection. (Closing the connection cannot be used to indicate the end of a request body, since that would leave no possibility for the server to send back a response.) Note that RTSP does not (at present) support the HTTP/1.1 "chunked" transfer coding(see [H3.6]) and requires the presence of the Content- Length header field. Given the moderate length of presentation descriptions returned, the server should always be able to determine its length, even if it is generated dynamically, making the chun- ked transfer encoding unnecessary. Even though Content-Length must be present if there is any entity body, the rules ensure reasonable behavior even if the length is not given explic- itly. 5 General Header Fields See [H4.5], except that Pragma, Transfer-Encoding and Upgrade headers are not defined: general-header = Cache-Control ; Section 12.9 | Connection ; Section 12.11 | CSeq ; Section 12.18 | Date ; Section 12.19 | Via ; Section 12.44 6 Request A request message from a client to a server or vice versa includes, within the first line of that message, the method to be applied to the resource, the identifier of the resource, and the protocol version in use. H. Schulzrinne, A. Rao, R. Lanphier [Page 17] Internet Draft RTSP May 28, 1999 Request = Request-Line ; Section 6.1 *( general-header ; Section 5 | request-header ; Section 6.2 | entity-header ) ; Section 8.1 CRLF [ message-body ] ; Section 4.3 6.1 Request Line Request-Line--- Method SP Request-URI SP RTSP-Version CRLF Method = "DESCRIBE" ; Section 10.2 | "ANNOUNCE" ; Section 10.3 | "GET_PARAMETER" ; Section 10.8 | "OPTIONS" ; Section 10.1 | "PAUSE" ; Section 10.6 | "PLAY" ; Section 10.5 | "RECORD" ; Section 10.11 | "REDIRECT" ; Section 10.10 | "SETUP" ; Section 10.4 | "SET_PARAMETER" ; Section 10.9 | "TEARDOWN" ; Section 10.7 | extension-method extension-method--- token Request-URI --- "*" | absolute_URI RTSP-Version --- "RTSP" "/" 1*DIGIT "." 1*DIGIT 6.2 Request Header Fields request-header = Accept ; Section 12.1 | Accept-Encoding ; Section 12.2 | Accept-Language ; Section 12.3 | Authorization ; Section 12.6 | Bandwidth ; Section 12.7 | Blocksize ; Section 12.8 | Conference ; Section 12.10 | From ; Section 12.21 | If-Modified-Since ; Section 12.24 H. Schulzrinne, A. Rao, R. Lanphier [Page 18] Internet Draft RTSP May 28, 1999 | Proxy-Require ; Section 12.28 | Range ; Section 12.30 | Referer ; Section 12.31 | Require ; Section 12.33 | Scale ; Section 12.35 | Session ; Section 12.38 | Speed ; Section 12.36 | Transport ; Section 12.40 | User-Agent ; Section 12.42 Note that in contrast to HTTP/1.1 [2], RTSP requests always contain the absolute URL (that is, including the scheme, host and port) rather than just the absolute path. HTTP/1.1 requires servers to understand the absolute URL, but clients are supposed to use the Host request header. This is purely needed for backward-compatibility with HTTP/1.0 servers, a consideration that does not apply to RTSP. The asterisk "*" in the Request-URI means that the request does not apply to a particular resource, but to the server itself, and is only allowed when the method used does not necessarily apply to a resource. One example would be: OPTIONS * RTSP/1.0 7 Response [H6] applies except that HTTP-Version is replaced by RTSP-Version. Also, RTSP defines additional status codes and does not define some HTTP codes. The valid response codes and the methods they can be used with are defined in Table 1. After receiving and interpreting a request message, the recipient responds with an RTSP response message. Response = Status-Line ; Section 7.1 *( general-header ; Section 5 | response-header ; Section 7.1.2 | entity-header ) ; Section 8.1 CRLF H. Schulzrinne, A. Rao, R. Lanphier [Page 19] Internet Draft RTSP May 28, 1999 [ message-body ] ; Section 4.3 7.1 Status-Line The first line of a Response message is the Status-Line, consisting of the protocol version followed by a numeric status code, and the textual phrase associated with the status code, with each element separated by SP characters. No CR or LF is allowed except in the final CRLF sequence. Status-Line--- RTSP-Version SP Status-Code SP Reason-Phrase CRLF 7.1.1 Status Code and Reason Phrase The Status-Code element is a 3-digit integer result code of the attempt to understand and satisfy the request. These codes are fully defined in Section 11. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata and the Reason-Phrase is intended for the human user. The client is not required to examine or display the Reason-Phrase. The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. There are 5 values for the first digit: + 1xx: Informational - Request received, continuing process + 2xx: Success - The action was successfully received, understood, and accepted + 3xx: Redirection - Further action must be taken in order to com- plete the request + 4xx: Client Error - The request contains bad syntax or cannot be fulfilled + 5xx: Server Error - The server failed to fulfill an apparently valid request The individual values of the numeric status codes defined for RTSP/1.0, and an example set of corresponding Reason-Phrase's, are presented below. The reason phrases listed here are only recommended -- they may be replaced by local equivalents without affecting the protocol. Note that RTSP adopts most HTTP/1.1 [2] status codes and adds RTSP-specific status codes starting at x50 to avoid conflicts with newly defined HTTP H. Schulzrinne, A. Rao, R. Lanphier [Page 20] Internet Draft RTSP May 28, 1999 status codes. Status-Code = "100" ; Continue | "200" ; OK | "201" ; Created | "250" ; Low on Storage Space | "300" ; Multiple Choices | "301" ; Moved Permanently | "302" ; Moved Temporarily | "303" ; See Other | "304" ; Not Modified | "305" ; Use Proxy | "400" ; Bad Request | "401" ; Unauthorized | "402" ; Payment Required | "403" ; Forbidden | "404" ; Not Found | "405" ; Method Not Allowed | "406" ; Not Acceptable | "407" ; Proxy Authentication Required | "408" ; Request Time-out | "410" ; Gone | "411" ; Length Required | "412" ; Precondition Failed | "413" ; Request Entity Too Large | "414" ; Request-URI Too Large | "415" ; Unsupported Media Type | "451" ; Parameter Not Understood | "452" ; Conference Not Found | "453" ; Not Enough Bandwidth | "454" ; Session Not Found | "455" ; Method Not Valid in This State | "456" ; Header Field Not Valid for Resource | "457" ; Invalid Range | "458" ; Parameter Is Read-Only | "459" ; Aggregate operation not allowed | "460" ; Only aggregate operation allowed | "461" ; Unsupported transport | "462" ; Destination unreachable | "500" ; Internal Server Error | "501" ; Not Implemented | "502" ; Bad Gateway | "503" ; Service Unavailable | "504" ; Gateway Time-out | "505" ; RTSP Version not supported H. Schulzrinne, A. Rao, R. Lanphier [Page 21] Internet Draft RTSP May 28, 1999 | "551" ; Option not supported | extension-code extension-code = 3DIGIT Reason-Phrase = * RTSP status codes are extensible. RTSP applications are not required to understand the meaning of all registered status codes, though such understanding is obviously desirable. However, applications MUST under- stand the class of any status code, as indicated by the first digit, and treat any unrecognized response as being equivalent to the x00 status code of that class, with the exception that an unrecognized response MUST NOT be cached. For example, if an unrecognized status code of 431 is received by the client, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 status code. In such cases, user agents SHOULD present to the user the entity returned with the response, since that entity is likely to include human-readable information which will explain the unusual sta- tus. 7.1.2 Response Header Fields The response-header fields allow the request recipient to pass addi- tional information about the response which cannot be placed in the Sta- tus-Line. These header fields give information about the server and about further access to the resource identified by the Request-URI. response-header = Location ; Section 12.26 | Proxy-Authenticate ; Section 12.27 | Public ; Section 12.29 | Range ; Section 12.30 | Retry-After ; Section 12.32 | RTP-Info ; Section 12.34 | Scale ; Section 12.35 | Session ; Section 12.38 | Server ; Section 12.37 | Speed ; Section 12.36 | Transport ; Section 12.40 | Unsupported ; Section 12.41 | Vary ; Section 12.43 | WWW-Authenticate ; Section 12.45 H. Schulzrinne, A. Rao, R. Lanphier [Page 22] Internet Draft RTSP May 28, 1999 Code reason -------------------------------------------------------- 100 Continue all -------------------------------------------------------- 200 OK all 201 Created RECORD 250 Low on Storage Space RECORD -------------------------------------------------------- 300 Multiple Choices all 301 Moved Permanently all 302 Moved Temporarily all 303 See Other all 305 Use Proxy all -------------------------------------------------------- 400 Bad Request all 401 Unauthorized all 402 Payment Required all 403 Forbidden all 404 Not Found all 405 Method Not Allowed all 406 Not Acceptable all 407 Proxy Authentication Required all 408 Request Timeout all 410 Gone all 411 Length Required all 412 Precondition Failed DESCRIBE, SETUP 413 Request Entity Too Large all 414 Request-URI Too Long all 415 Unsupported Media Type all 451 Parameter Not Understood SETUP 452 Illegal Conference Identifier SETUP 453 Not Enough Bandwidth SETUP 454 Session Not Found all 455 Method Not Valid In This State all 456 Header Field Not Valid all 457 Invalid Range PLAY 458 Parameter Is Read-Only SET_PARAMETER 459 Aggregate Operation Not Allowed all 460 Only Aggregate Operation Allowed all 461 Unsupported Transport all 462 Destination Unreachable all -------------------------------------------------------- 500 Internal Server Error all 501 Not Implemented all 502 Bad Gateway all 503 Service Unavailable all 504 Gateway Timeout all H. Schulzrinne, A. Rao, R. Lanphier [Page 23] Internet Draft RTSP May 28, 1999 505 RTSP Version Not Supported all 551 Option not support all Table 1: Status codes and their usage with RTSP methods Response-header field names can be extended reliably only in combination with a change in the protocol version. However, new or experimental header fields MAY be given the semantics of response-header fields if all parties in the communication recognize them to be response-header fields. Unrecognized header fields are treated as entity-header fields. 8 Entity Request and Response messages MAY transfer an entity if not otherwise restricted by the request method or response status code. An entity con- sists of entity-header fields and an entity-body, although some responses will only include the entity-headers. In this section, both sender and recipient refer to either the client or the server, depending on who sends and who receives the entity. 8.1 Entity Header Fields Entity-header fields define optional metainformation about the entity- body or, if no body is present, about the resource identified by the request. entity-header = Allow ; Section 12.5 | Content-Base ; Section 12.12 | Content-Encoding ; Section 12.13 | Content-Language ; Section 12.14 | Content-Length ; Section 12.15 | Content-Location ; Section 12.16 | Content-Type ; Section 12.17 | Expires ; Section 12.20 | Last-Modified ; Section 12.25 | extension-header extension-header = message-header The extension-header mechanism allows additional entity-header fields to be defined without changing the protocol, but these fields cannot be assumed to be recognizable by the recipient. Unrecognized header fields SHOULD be ignored by the recipient and forwarded by proxies. H. Schulzrinne, A. Rao, R. Lanphier [Page 24] Internet Draft RTSP May 28, 1999 8.2 Entity Body See [H7.2] 9 Connections RTSP requests can be transmitted in several different ways: + persistent transport connections used for several request- response transactions; + one connection per request/response transaction; + connectionless mode. The type of transport connection is defined by the RTSP URI (Section 3.2). For the scheme "rtsp", a persistent connection is assumed, while the scheme "rtspu" calls for RTSP requests to be sent without setting up a connection. Unlike HTTP, RTSP allows the media server to send requests to the media client. However, this is only supported for persistent connections, as the media server otherwise has no reliable way of reaching the client. Also, this is the only way that requests from media server to client are likely to traverse firewalls. 9.1 Pipelining A client that supports persistent connections or connectionless mode MAY "pipeline" its requests (i.e., send multiple requests without waiting for each response). A server MUST send its responses to those requests in the same order that the requests were received. 9.2 Reliability and Acknowledgements Requests are acknowledged by the receiver unless they are sent to a mul- ticast group. If there is no acknowledgement, the sender may resend the same message after a timeout of one round-trip time (RTT). The round- trip time is estimated as in TCP (RFC 1123) [18], with an initial round- trip value of 500 ms. An implementation MAY cache the last RTT measure- ment as the initial value for future connections. If a reliable transport protocol is used to carry RTSP, requests MUST NOT be retransmitted; the RTSP application MUST instead rely on the underlying transport to provide reliability. H. Schulzrinne, A. Rao, R. Lanphier [Page 25] Internet Draft RTSP May 28, 1999 If both the underlying reliable transport such as TCP and the RTSP application retransmit requests, it is possible that each packet loss results in two retransmissions. The receiver can- not typically take advantage of the application-layer retrans- mission since the transport stack will not deliver the appli- cation-layer retransmission before the first attempt has reached the receiver. If the packet loss is caused by conges- tion, multiple retransmissions at different layers will exac- erbate the congestion. If RTSP is used over a small-RTT LAN, standard procedures for optimizing initial TCP round trip estimates, such as those used in T/TCP (RFC 1644) [22], can be beneficial. The Timestamp header (Section 12.39) is used to avoid the retransmis- sion ambiguity problem [23] and obviates the need for Karn's algorithm. Each request carries a sequence number in the CSeq header (Section 12.18), which is incremented by one for each distinct request transmit- ted. If a request is repeated because of lack of acknowledgement, the request MUST carry the original sequence number (i.e., the sequence num- ber is not incremented). Systems implementing RTSP MUST support carrying RTSP over TCP and MAY support UDP. The default port for the RTSP server is 554 for both UDP and TCP. A number of RTSP packets destined for the same control end point may be packed into a single lower-layer PDU or encapsulated into a TCP stream. RTSP data MAY be interleaved with RTP and RTCP packets. Unlike HTTP, an RTSP message MUST contain a Content-Length header field whenever that message contains a payload. Otherwise, an RTSP packet is terminated with an empty line immediately following the last message header. 10 Method Definitions The method token indicates the method to be performed on the resource identified by the Request-URI case-sensitive. New methods may be defined in the future. Method names may not start with a $ character (decimal 24) and must be a token. Methods are summarized in Table 2. 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 26] Internet Draft RTSP May 28, 1999 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 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). 10.2 DESCRIBE The DESCRIBE method retrieves the description of a presentation or media object identified by the request URL from a server. It may use the Accept header to specify the description formats that the client H. Schulzrinne, A. Rao, R. Lanphier [Page 27] Internet Draft RTSP May 28, 1999 understands. The server responds with a description of the requested resource. The DESCRIBE reply-response pair constitutes the media ini- tialization 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 pre- sentation 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 descrip- tion for the same media via RTSP. Additionally, servers SHOULD NOT use the DESCRIBE response as a means of media indirection. By forcing a DESCRIBE response to contain all media initial- ization for the set of streams that it describes, and discour- aging use of DESCRIBE for media indirection, we avoid looping problems that might result from other approaches. H. Schulzrinne, A. Rao, R. Lanphier [Page 28] Internet Draft RTSP May 28, 1999 Media initialization is a requirement for any RTSP-based system, but the RTSP specification does not dictate that this must be done via the DESCRIBE method. There are three ways that an RTSP client may receive initialization information: + via RTSP's DESCRIBE method; + via some other protocol (HTTP, email attachment, etc.); + via the command line or standard input (thus working as a browser helper application launched with an SDP file or other media ini- tialization format). It is 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 stan- dard 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 descrip- tion in real-time. If a new media stream is added to a presentation (e.g., during a live presentation), the whole presentation description should be sent again, rather than just the additional components, so that components can be deleted. Example: C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0 CSeq: 312 Date: 23 Jan 1997 15:35:06 GMT Session: 47112344 Content-Type: application/sdp Content-Length: 332 v=0 o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4 s=SDP Seminar i=A Seminar on the session description protocol u=http://www.cs.ucl.ac.uk/staff/M.Handley/sdp.03.ps H. Schulzrinne, A. Rao, R. Lanphier [Page 29] Internet Draft RTSP May 28, 1999 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: 312 10.4 SETUP The SETUP request for a URI specifies the transport mechanism to be used for the streamed media. A client can issue a SETUP request for a stream that is already playing to change transport parameters, which a server MAY allow. If it does not allow this, it MUST respond with error 455 (Method Not Valid In This State). For the benefit of any intervening firewalls, a client must indicate the transport parameters even if it has no influence over these parameters, for example, where the server advertises a fixed multicast address. Since SETUP includes all transport initialization information, firewalls and other intermediate network devices (which need this information) are spared the more arduous task of parsing the DESCRIBE response, which has been reserved for media ini- tialization. The Transport header specifies the transport parameters acceptable to the client for data transmission; the response will contain the trans- port 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 30] Internet Draft RTSP May 28, 1999 The server generates session identifiers in response to SETUP requests. If a SETUP request to a server includes a session identifier, the server MUST bundle this setup request into the 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 31] Internet Draft RTSP May 28, 1999 action and can be used by the client to test server liveness. The Range header may also contain a time parameter. This parameter specifies a time in UTC at which the playback should start. If the mes- sage is received after the specified time, playback is started immedi- ately. 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 align- ment 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 32] Internet Draft RTSP May 28, 1999 A media server only supporting playback MUST support the npt format and MAY support the clock and smpte formats. 10.6 PAUSE The PAUSE request causes the stream delivery to be interrupted (halted) temporarily. If the request URL names a stream, only playback and recording of that stream is halted. For example, for audio, this is equivalent to muting. If the request URL names a presentation or group of streams, delivery of all currently active streams within the presen- tation or group is halted. After resuming playback or recording, syn- chronization 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 encoun- tering 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 33] Internet Draft RTSP May 28, 1999 For example, if the server has play requests for ranges 10 to 15 and 20 to 29 pending and then receives a pause request for NPT 21, it would start playing the second range and stop at NPT 21. If the pause request is for NPT 12 and the server is playing at NPT 13 serving the first play request, the server stops immediately. If the pause request is for NPT 16, the server stops after completing the first play request and dis- cards 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, free- ing the resources associated with it. If the URI is the presentation URI for this presentation, any RTSP session identifier associated with the session is no longer valid. Unless all transport parameters are defined by the session description, a SETUP request has to be issued before the session can be played again. Example: C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 892 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 892 10.8 GET_PARAMETER The GET_PARAMETER request retrieves the value of a parameter of a pre- sentation 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"). H. Schulzrinne, A. Rao, R. Lanphier [Page 34] Internet Draft RTSP May 28, 1999 Example: S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 431 Content-Type: text/parameters Session: 12345678 Content-Length: 15 packets_received jitter 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 sev- eral 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 param- eter 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 35] Internet Draft RTSP May 28, 1999 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 Parameter Not Understood CSeq: 421 Content-length: 10 Content-type: text/parameters barparam The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation. 10.10 REDIRECT A redirect request informs the client that it must connect to another server location. It contains the mandatory header Location, which indi- cates 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 36] Internet Draft RTSP May 28, 1999 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, com- mence 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 Loca- tion header. A media server supporting recording of live presentations MUST support the clock range format; the smpte format does not make sense. In this example, the media server was previously invited to the confer- ence indicated. C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0 CSeq: 954 Session: 12345678 Conference: 128.16.64.19/32492374 10.12 Embedded (Interleaved) Binary Data Certain firewall designs and other circumstances may force a server to interleave RTSP methods and stream data. This interleaving should gener- ally 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 decimal), followed by a one-byte channel identifier, followed by the length of the encapsulated binary data as a binary, two-byte integer in network byte order. The stream data follows immediately afterwards, without a CRLF, but including the upper-layer protocol headers. Each $ block contains exactly one upper-layer protocol data unit, e.g., one RTP packet. H. Schulzrinne, A. Rao, R. Lanphier [Page 37] Internet Draft RTSP May 28, 1999 The channel identifier is defined in the Transport header with the interleaved parameter(Section 12.40). 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 12.40). RTCP is needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a con- venient way to tunnel RTP/RTCP packets through the TCP control connection when required by the network configuration and transfer them onto UDP when possible. C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 CSeq: 2 Transport: RTP/AVP/TCP;interleaved=0-1 S->C: RTSP/1.0 200 OK CSeq: 2 Date: 05 Jun 1997 18:57:18 GMT Transport: RTP/AVP/TCP;interleaved=0-1 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 38] Internet Draft RTSP May 28, 1999 Where applicable, HTTP status [H10] codes are reused. Status codes that have the same meaning are not repeated here. See Table 1 for a listing of which status codes may be returned by which requests. 11.1 Success 2xx 11.1.1 250 Low on Storage Space The server returns this warning after receiving a RECORD request that it may not be able to fulfill completely due to insufficient storage space. If possible, the server should use the Range header to indicate what time period it may still be able to record. Since other processes on the server may be consuming storage space simultaneously, a client should take this only as an estimate. 11.2 Redirection 3xx See [H10.3]. Within RTSP, redirection may be used for load balancing or redirecting stream requests to a server topologically closer to the client. Mecha- nisms to determine topological proximity are beyond the scope of this specification. 11.3 Client Error 4xx 11.3.1 405 Method Not Allowed The method specified in the request is not allowed for the resource identified by the request URI. The response MUST include an Allow header containing a list of valid methods for the requested resource. This sta- tus 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 39] Internet Draft RTSP May 28, 1999 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 eas- ier. 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. 11.3.8 457 Invalid Range The Range value given is out of bounds, e.g., beyond the end of the presentation. 11.3.9 458 Parameter Is Read-Only The parameter to be set by SET_PARAMETER can be read but not modified. 11.3.10 459 Aggregate Operation Not Allowed The requested method may not be applied on the URL in question since it is an aggregate (presentation) URL. The method may be applied on a stream URL. 11.3.11 460 Only Aggregate Operation Allowed The requested method may not be applied on the URL in question since it is not an aggregate (presentation) URL. The method may be applied on the presentation URL. 11.3.12 461 Unsupported Transport The Transport field did not contain a supported transport specifica- tion. 11.3.13 462 Destination Unreachable H. Schulzrinne, A. Rao, R. Lanphier [Page 40] Internet Draft RTSP May 28, 1999 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. 11.4 Server Error 5xx 11.4.1 551 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. 12 Header Field Definitions HTTP/1.1 [2] or other, non-standard header fields not listed here cur- rently have no well-defined meaning and SHOULD be ignored by the recipi- ent. 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. 12.1 Accept The Accept request-header field can be used to specify certain presen- tation description content types which are acceptable for the response. The "level" parameter for presentation descriptions is prop- erly defined as part of the MIME type registration, not here. See [H14.1] for syntax. Example of use: Accept: application/rtsl, application/sdp;level=2 H. Schulzrinne, A. Rao, R. Lanphier [Page 41] Internet Draft RTSP May 28, 1999 Header type support methods ------------------------------------------------------------- Accept R opt. entity Accept-Encoding R opt. entity Accept-Language R opt. all Accept-Ranges R opt. all Allow e 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 CSeq g req. all Date g opt. all Expires e opt. DESCRIBE, ANNOUNCE From R opt. all If-Match R opt. SETUP If-Modified-Since R opt. DESCRIBE, SETUP Last-Modified e opt. entity Location r opt. 201, 30x Proxy-Authenticate r req. 407 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 g opt. PLAY, RECORD Session g req. all but OPTIONS Server r opt. all Speed g opt. PLAY Transport g req. SETUP Unsupported r req. all User-Agent R opt. all Vary r opt. all Via g opt. all H. Schulzrinne, A. Rao, R. Lanphier [Page 42] Internet Draft RTSP May 28, 1999 WWW-Authenticate r opt. all Table 3: Overview of RTSP header fields 12.2 Accept-Encoding See [H14.3] 12.3 Accept-Language See [H14.4]. Note that the language specified applies to the presenta- tion description and any reason phrases, not the media content. 12.4 Accept-Ranges 12.5 Allow The Allow entity-header field lists the methods supported by the resource identified by the request-URI. The purpose of this field is to strictly inform the recipient of valid methods associated with the resource. An Allow header field must be present in a 405 (Method Not Allowed) response. Example of use: Allow: SETUP, PLAY, RECORD, SET_PARAMETER 12.6 Authorization See [H14.8] 12.7 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. Bandwidth--- "Bandwidth" ":" 1*DIGIT Example: Bandwidth: 4000 H. Schulzrinne, A. Rao, R. Lanphier [Page 43] Internet Draft RTSP May 28, 1999 12.8 Blocksize The Blocksize 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 mul- tiple of the minimum, media-specific block size, or override it with the media-specific size if necessary. The block size MUST be a positive dec- imal number, measured in octets. The server only returns an error (416) if the value is syntactically invalid. Blocksize--- "Blocksize" ":" 1*DIGIT 12.9 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 applica- tion, 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" H. Schulzrinne, A. Rao, R. Lanphier [Page 44] Internet Draft RTSP May 28, 1999 | "must-revalidate" | "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 any- where. 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 opera- tional 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 authenti- cation 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 net- work 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 con- straints of the request, or respond with a 504 (Gateway Time- out) status. However, if a group of caches is being operated as a unified system with good internal connectivity, such a request MAY be forwarded within that group of caches. max-stale: Indicates that the client is willing to accept a media stream that has exceeded its expiration time. If max-stale is H. Schulzrinne, A. Rao, R. Lanphier [Page 45] Internet Draft RTSP May 28, 1999 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.) 12.10 Conference The Conference 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 A response code of 452 (Conference Not Found) is returned if the confer- ence-id is not valid. 12.11 Connection See [H14.10] 12.12 Content-Base See [H14.11] H. Schulzrinne, A. Rao, R. Lanphier [Page 46] Internet Draft RTSP May 28, 1999 12.13 Content-Encoding See [H14.12] 12.14 Content-Language See [H14.13] 12.15 Content-Length The Content-Length general-header field contains the length of the con- tent 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]. 12.16 Content-Location See [H14.15] 12.17 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. 12.18 CSeq The CSeq general-header field specifies the sequence number for an RTSP request-response pair. This field MUST be present in all requests and responses. For every RTSP request containing the given sequence number, the corresponding response will have the same number. Any retransmitted request must contain the same sequence number as the original (i.e. the sequence number is not incremented for retransmissions of the same request). CSeq--- "Cseq" ":" 1*DIGIT 12.19 Date See [H14.19]. 12.20 Expires H. Schulzrinne, A. Rao, R. Lanphier [Page 47] Internet Draft RTSP May 28, 1999 The Expires entity-header field gives a date and time after which the description or media-stream should be considered stale. The interpreta- tion depends on the method: DESCRIBE response: The Expires header indicates a date and time after which the description should be considered stale. A stale cache entry may not normally be returned by a cache (either a proxy cache or an user agent cache) unless it is first validated with the origin server (or with an intermediate cache that has a fresh copy of the entity). See section 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, espe- cially 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 approx- imately 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 12.9). 12.21 From H. Schulzrinne, A. Rao, R. Lanphier [Page 48] Internet Draft RTSP May 28, 1999 See [H14.22]. 12.22 Host The Host HTTP request header field is not needed for RTSP. It should be silently ignored if sent. 12.23 If-Match See [H14.25]. The If-Match request-header field is especially useful for ensuring the integrity of the presentation description, in both the case where it is fetched via means external to RTSP (such as HTTP), or in the case where the server implementation is guaranteeing the integrity of the descrip- tion 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. 12.24 If-Modified-Since The If-Modified-Since request-header field is used with the DESCRIBE and SETUP methods to make them conditional. If the requested variant has not been modified since the time specified in this field, a description will not be returned from the server (DESCRIBE) or a stream will not be set up (SETUP). Instead, a 304 (Not Modified) response will be returned without any message-body. If-Modified-Since--- "If-Modified-Since" ":" HTTP-date An example of the field is: If-Modified-Since: Sat, 29 Oct 1994 19:43:31 GMT 12.25 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 49] Internet Draft RTSP May 28, 1999 12.26 Location See [H14.30]. 12.27 Proxy-Authenticate See [H14.33]. 12.28 Proxy-Require The Proxy-Require request-header field is used to indicate proxy-sensi- tive features that MUST be supported by the proxy. Any Proxy-Require header features that are not supported by the proxy MUST be negatively acknowledged by the proxy to the client if not supported. Servers should treat this field identically to the Require field. See Section 12.33 for more details on the mechanics of this message and a usage example. 12.29 Public See [H14.35]. 12.30 Range The Range 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 50] Internet Draft RTSP May 28, 1999 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. 12.31 Referer See [H14.37]. The URL refers to that of the presentation description, typically retrieved via HTTP. 12.32 Retry-After See [H14.38]. 12.33 Require The Require request-header field 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 51] Internet Draft RTSP May 28, 1999 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 commu- nicated via the RTSP exchange, since that relationship is an immutable property of "funky-feature" and thus should not be transmitted with every exchange. Proxies and other intermediary devices 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 12.28). 12.34 RTP-Info The RTP-Info response-header field is used to set RTP-specific parame- ters 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 con- trol, a particular stream may not actually generate a packet H. Schulzrinne, A. Rao, R. Lanphier [Page 52] Internet Draft RTSP May 28, 1999 for the Range time value returned or implied. Thus, there is no guarantee that the packet with the sequence number indi- cated by seq actually has the timestamp indicated by rtp- time.) 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 times- tamps to NPT. Furthermore, in order to ensure that this information is available at the necessary time (immedi- ately at startup or after a seek), and that it is deliv- ered reliably, this mapping is placed in the RTSP control channel. In order to compensate for drift for long, uninterrupted pre- sentations, 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. Syntax: RTP-Info --- "RTP-Info" ":" 1#rtsp-info-spec rtsp-info-spec--- stream-url 1*parameter stream-url --- quoted-url | unquoted-url unquoted-url --- "url" "=" safe-url | ";" "mode" = <"> 1#Method <"> quoted-url --- "url" "=" <"> needquote-url <"> safe-url --- url needquote-url --- url url --- ( absoluteURI | relativeURI ) parameter --- ";" "seq" "=" 1*DIGIT | ";" "rtptime" "=" 1*DIGIT Additional constraint: safe-url MUST NOT contain the semicolon (";") or comma (",") characters. The quoted-url form SHOULD only be used when a URL does not meet the safe-url constraint, in order to ensure compati- bility with implementations conformant to RFC 2326 . absoluteURI and relativeURI are defined in RFC 2396 . Example: RTP-Info: url=rtsp://foo.com/bar.avi/streamid=0;seq=45102, url=rtsp://foo.com/bar.avi/streamid=1;seq=30211 H. Schulzrinne, A. Rao, R. Lanphier [Page 53] Internet Draft RTSP May 28, 1999 12.35 Scale A scale value of 1 indicates normal play or record at the normal forward viewing rate. If not 1, the value corresponds to the rate with respect to normal viewing rate. For example, a ratio of 2 indicates twice the normal viewing rate ("fast forward") and a ratio of 0.5 indicates half the normal viewing rate. In other words, a ratio of 2 has normal play time increase at twice the wallclock rate. For every second of elapsed (wallclock) time, 2 seconds of content will be delivered. A negative value indicates reverse direction. Unless requested otherwise by the Speed parameter, the data rate SHOULD not be changed. Implementation of scale changes depends on the server and media type. For video, a server may, for example, deliver only key frames or selected key frames. For audio, it may time-scale the audio while preserving pitch or, less desirably, deliver fragments of audio. The server should try to approximate the viewing rate, but may restrict the range of scale values that it supports. The response MUST contain the actual scale value chosen by the server. If the request contains a Range parameter, the new scale value will take effect at that time. Scale--- "Scale" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ] Example of playing in reverse at 3.5 times normal rate: Scale: -3.5 12.36 Speed The Speed request-header field requests the server to deliver data to the client at a particular speed, contingent on the server's ability and desire to serve the media stream at the given speed. Implementation by the server is OPTIONAL. The default is the bit rate of the stream. The parameter value is expressed as a decimal ratio, e.g., a value of 2.0 indicates that data is to be delivered twice as fast as normal. A speed of zero is invalid. If the request contains a Range parameter, the new speed value will take effect at that time. H. Schulzrinne, A. Rao, R. Lanphier [Page 54] Internet Draft RTSP May 28, 1999 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 presenta- tion 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. 12.37 Server See [H14.39] 12.38 Session The Session request-header 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) and MUST be returned in the SETUP response. Once a client receives a Session identifier, it MUST return it for any request related to that session. Session--- "Session" ":" session-id [ ";" "timeout" "=" delta-seconds ] The timeout parameter is only allowed in a response header. The server uses it to indicate to the client how long the server is prepared to wait between RTSP commands before closing the session due to lack of activity (see Section A). The timeout is measured in seconds, with a default of 60 seconds (1 minute). Note that a session identifier identifies an RTSP session across trans- port 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 55] Internet Draft RTSP May 28, 1999 The session identifier is needed to distinguish several deliv- ery requests for the same URL coming from the same client. The response 454 (Session Not Found) is returned if the session identi- fier is invalid. 12.39 Timestamp The Timestamp general-header field 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) ] 12.40 Transport The Transport request-header field indicates which transport protocol is to be used and configures its parameters such as destination address, compression, multicast time-to-live and destination port for a single stream. It sets those values not already determined by a presentation description. Transports are comma separated, listed in order of preference. Parame- ters may be added to each transport, separated by a semicolon. The Transport header field 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 field in the response to indicate the values actually chosen. 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 56] Internet Draft RTSP May 28, 1999 / 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: This parameter is a mutually exclusive indi- cation of whether unicast or multicast delivery will be attempted. One of the two values MUST be specified. Clients that are capable of handling both unicast and multicast trans- mission 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 destination address with the destina- tion 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. source: If the source address for the stream is different than can be derived from the RTSP endpoint address (the server in play- back or the client in recording), the source address MAY be specified. This information may also be available through SDP. How- ever, since this is more a feature of transport than media initialization, the authoritative source for this information should be in the SETUP response. 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 pro- vided, the default is PLAY. H. Schulzrinne, A. Rao, R. Lanphier [Page 57] Internet Draft RTSP May 28, 1999 append: If the mode parameter includes RECORD, the append parame- ter indicates that the media data should append to the exist- ing 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 parame- ter 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 multi- cast session. It is specified as a range, e.g., port=3456-3457 client_port: This parameter provides the unicast RTP/RTCP port pair on the client where media data and control information is to be sent. It is specified as a range, e.g., port=3456-3457 server_port: This parameter provides the unicast RTP/RTCP port pair on the server where media data and control information is to be sent. It is specified as a range, e.g., port=3456-3457 ssrc: The ssrc parameter indicates the RTP SSRC [24] 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, and is expressed as an eight digit hexideci- mal value. Transport ---- "Transport" ":" H. Schulzrinne, A. Rao, R. Lanphier [Page 58] Internet Draft RTSP May 28, 1999 1#transport-spec transport-spec = transport-protocol/profile[/lower-transport] *parameter transport-protocol = "RTP" profile = "AVP" lower-transport = "TCP" | "UDP" parameter = ( "unicast" | "multicast" ) | ";" "source" [ "=" address ] | ";" "destination" [ "=" address ] | ";" "interleaved" "=" channel [ "-" channel ] | ";" "append" | ";" "ttl" "=" ttl | ";" "layers" "=" 1*DIGIT | ";" "port" "=" port [ "-" port ] | ";" "client_port" "=" port [ "-" port ] | ";" "server_port" "=" port [ "-" port ] | ";" "source" "=" address | ";" "ssrc" "=" ssrc | ";" "mode" = <"> 1#Method <"> ttl = 1*3(DIGIT) port = 1*5(DIGIT) ssrc = 8*8(HEX) channel = 1*3(DIGIT) address = host mode = <"> *Method <"> | Method Example: Transport: RTP/AVP;multicast;ttl=127;mode="PLAY", RTP/AVP;unicast;client_port=3456-3457;mode="PLAY" The Transport header field is restricted to describing a sin- gle 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. 12.41 Unsupported The Unsupported response-header field lists the features not supported by the server. In the case where the feature was specified via the Proxy-Require field (Section 12.33), if there is a proxy on the path between the client and the server, the proxy MUST insert a response mes- sage with a status code of 551 (Option Not Supported). H. Schulzrinne, A. Rao, R. Lanphier [Page 59] Internet Draft RTSP May 28, 1999 See Section 12.33 for a usage example. Unsupported--- "Unsupported" ":" 1#option-tag 12.42 User-Agent See [H14.42] 12.43 Vary See [H14.43] 12.44 Via See [H14.44]. 12.45 WWW-Authenticate 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_PARAM- ETER do not return any data, caching is not really an issue for these requests.) However, it is desirable for the continuous media data, typi- cally 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 descrip- tion. It can determine whether the copy is up-to-date by issuing a SETUP or DESCRIBE request, respectively, and comparing the Last-Modi- fied header with that of the cached copy. If the copy is not up-to-date, it modifies the SETUP transport parameters as appropriate and forwards the request to the origin server. Subsequent control commands such as PLAY or PAUSE then pass the proxy unmodified. The proxy delivers the continuous media data to the client, while possibly making a local copy for later reuse. The exact behavior allowed to the cache is given by the cache-response directives described in Section 12.9. 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 60] Internet Draft RTSP May 28, 1999 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. Typi- cally, a cache eliminates all transport-references (that is, multicast information) from the presentation description, since these are indepen- dent 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 lan- guage or copyright restrictions. It may also give an indication about the timeline of the movie. In this example, the client is only interested in the last part of the movie. C->W: GET /twister.sdp HTTP/1.1 Host: www.example.com Accept: application/sdp W->C: HTTP/1.0 200 OK Content-Type: application/sdp 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 61] Internet Draft RTSP May 28, 1999 m=video 0 RTP/AVP 31 a=control:rtsp://video.example.com/twister/video C->A: SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0 CSeq: 1 Transport: RTP/AVP/UDP;unicast;client_port=3056-3057 A->C: RTSP/1.0 200 OK CSeq: 1 Session: 12345678 Transport: RTP/AVP/UDP;unicast;client_port=3056-3057; server_port=5000-5001 C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 CSeq: 1 Transport: RTP/AVP/UDP;unicast;client_port=3058-3059 V->C: RTSP/1.0 200 OK CSeq: 1 Session: 23456789 Transport: RTP/AVP/UDP;unicast;client_port=3058-3059; server_port=5002-5003 C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 CSeq: 2 Session: 23456789 Range: smpte=0:10:00- V->C: RTSP/1.0 200 OK CSeq: 2 Session: 23456789 Range: smpte=0:10:00-0:20:00 RTP-Info: url=rtsp://video.example.com/twister/video; seq=12312232;rtptime=78712811 C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0 CSeq: 2 Session: 12345678 Range: smpte=0:10:00- A->C: RTSP/1.0 200 OK CSeq: 2 Session: 12345678 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 62] Internet Draft RTSP May 28, 1999 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. Con- tainer 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. 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 63] Internet Draft RTSP May 28, 1999 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 M->C: RTSP/1.0 200 OK CSeq: 4 Session: 12345678 H. Schulzrinne, A. Rao, R. Lanphier [Page 64] Internet Draft RTSP May 28, 1999 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: C->S DESCRIBE rtsp://foo.com/test.wav RTSP/1.0 H. Schulzrinne, A. Rao, R. Lanphier [Page 65] Internet Draft RTSP May 28, 1999 Accept: application/x-rtsp-mh, application/sdp 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: H. Schulzrinne, A. Rao, R. Lanphier [Page 66] Internet Draft RTSP May 28, 1999 C->S PLAY rtsp://foo.com/test.wav/streamid=0 RTSP/1.0 CSeq: 3 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 descrip- tion, 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 C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0 CSeq: 1 M->C: RTSP/1.0 200 OK CSeq: 1 H. Schulzrinne, A. Rao, R. Lanphier [Page 67] Internet Draft RTSP May 28, 1999 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 c=IN IP4 224.2.0.1/16 a=control:rtsp://live.example.com/concert/audio C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 CSeq: 2 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 68] Internet Draft RTSP May 28, 1999 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: 199702170042.SAA08642@obiwan.arl.wustl.edu M->C: RTSP/1.0 200 OK CSeq: 2 Transport: RTP/AVP;multicast;destination=225.219.201.15; port=7000-7001;ttl=127 Session: 91389234234 Conference: 199702170042.SAA08642@obiwan.arl.wustl.edu 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 69] Internet Draft RTSP May 28, 1999 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 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 70] Internet Draft RTSP May 28, 1999 The RTSP syntax is described in an augmented Backus-Naur form (BNF) as used in RFC 2068 [2]. 15.1 Base Syntax OCTET --- CHAR --- UPALPHA --- LOALPHA --- ALPHA --- UPALPHA | LOALPHA DIGIT --- CTL --- CR --- LF --- SP --- HT --- <"> --- CRLF --- CR LF LWS --- [CRLF] 1*( SP | HT ) TEXT --- tspecials --- "(" | ")" | "<" | ">" | "@" | "," | ";" | ":" | " \*[3trans].nr 3crow 18 \!.3rvpt "\$1" \&\h'|\n[3cl0]u' \!.3rvpt "\$1" \*[3trans].nr 3brule 1 \!.3rvpt "\$1" \!.3rmk "\$1" " | <"> | "/" | "[" | "]" | "?" | "=" | "{" | "}" | SP | HT token --- 1* quoted-string --- ( <"> *(qdtext) <"> ) qdtext --- > quoted-pair --- " \*[3trans].nr 3crow 26 \!.3rvpt "\$1" \&\h'|\n[3cl0]u' \!.3rvpt "\$1" H. Schulzrinne, A. Rao, R. Lanphier [Page 71]