HTTP/1.1 200 OK Date: Tue, 09 Apr 2002 05:12:09 GMT Server: Apache/1.3.20 (Unix) Last-Modified: Thu, 27 Mar 1997 16:23:00 GMT ETag: "2f58b2-1f14d-333a9ee4" Accept-Ranges: bytes Content-Length: 127309 Connection: close Content-Type: text/plain Internet Engineering Task Force MMUSIC WG Internet Draft H. Schulzrinne, A. Rao, R. Lanphier ietf-mmusic-rtsp-02.txt Columbia U./Netscape/Progressive Networks March 27, 1997 Expires: September 26, 1997 Real Time Streaming Protocol (RTSP) STATUS OF THIS MEMO This document is an Internet-Draft. 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 learn the current status of any Internet-Draft, please check the ``1id-abstracts.txt'' listing contained in the Internet-Drafts Shadow Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe), munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or ftp.isi.edu (US West Coast). Distribution of this document is unlimited. ABSTRACT The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time properties. RTSP provides an extensible framework to enable controlled, on-demand delivery of real-time data, such as audio and video. Sources of data can include both live data feeds and stored clips. This protocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and delivery mechanisms based upon RTP (RFC 1889). 1 Introduction 1.1 Purpose The Real-Time Streaming Protocol (RTSP) establishes and controls H. Schulzrinne, A. Rao, R. Lanphier [Page 1] Internet Draft RTSP March 27, 1997 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 9.11). In other words, RTSP 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 session, an RTSP client may open and close many reliable transport connections 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, so that extension mechanisms to HTTP can in most cases also be added to RTSP. However, RTSP differs in a number of important aspects from HTTP: o RTSP introduces a number of new methods and has a different protocol identifier. o An RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. (RTSP servers and clients MAY use the HTTP state maintenance mechanism [2].) o Both an RTSP server and client can issue requests. o Data is carried out-of-band, by a different protocol. (There is an exception to this.) o RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, consistent with current HTML internationalization efforts [3]. o The Request-URI always contains the absolute URI. Because of backward compatibility with a historical blunder, HTTP/1.1 carries only the absolute path in the request H. Schulzrinne, A. Rao, R. Lanphier [Page 2] Internet Draft RTSP March 27, 1997 This makes virtual hosting easier. However, this is incompatible with HTTP/1.1, which may be a bad idea. The protocol supports the following operations: Retrieval of media from media server: The client can request a presentation 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 distributed teaching applications. Several parties in the conference 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. 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 xxxx [4]. 1.3 Terminology Some of the terminology has been adopted from HTTP/1.1 [5]. Terms not listed here are defined as in HTTP/1.1. 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. Continuous media: Data where there is a timing relationship between H. Schulzrinne, A. Rao, R. Lanphier [Page 3] Internet Draft RTSP March 27, 1997 source and sink, that is, the sink must reproduce the timing relationshop that existed at the source. The most common examples of continuous media are audio and motion video. Continuous media can be realtime (interactive) , where there is a "tight" timing relationship between source and sink, or streaming (playback) , where the relationship is less strict. Participant: Participants are members of conferences. A participant may be a machine, e.g., a media record or playback server. Media server: The network entity 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 parameter: Parameter specific to a media type that may be changed while the stream is being played or prior to it. (Media) stream: A single media instance, e.g., an audio stream or a video stream as well as a single whiteboard or shared application group. When using RTP, a stream consists of all RTP and RTCP packets created by a source within an RTP session. This is equivalent to the definition of a DSM-CC stream. Message: The basic unit of RTSP communication, consisting of a structured sequence of octets matching the syntax defined in Section 14 and transmitted via a connection or a connectionless protocol. Presentation: A set of one or more streams which the server allows the client to manipulate together. A presentation has a single time axis for all streams belonging to it. Presentations are defined by presentation descriptions (see below). A presentation description contains RTSP URIs that define which streams can be controlled individually and an RTSP URI to control the whole presentation. A movie or live concert consisting of one or more audio and video streams is be an example of a presentation. Presentation description: A presentation description contains information about one or more media streams within a presentation, such as the set of encodings, network addresses and information about the content. Other IETF protocols such as SDP [6] use the term "session" for a live presentation. 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 4] Internet Draft RTSP March 27, 1997 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 consist 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. 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 transport level (TLS [7]) or within the protocol itself. All HTTP authentication mechanisms such as basic [5] and digest authentication [8] are directly applicable. Transport-independent: RTSP may use either an unreliable datagram protocol (UDP) [9], a reliable datagram protocol (RDP, not widely used [10]) or a reliable stream protocol such as TCP [11] as it implements application-level reliability. Multi-server capable: Each media stream within a presentation can reside on a different server. The client automatically establishes several concurrent control sessions with the different media servers. Media synchronization is performed at the transport level. Control of recording devices: The protocol can control both recording and playback devices, as well as devices that can alternate between the two modes ("VCR"). Separation of stream control and conference initiation: Stream control is divorced from inviting a media server to a conference. The only requirement is that the conference initiation protocol either provides or can be used to create a unique conference identifier. In particular, SIP [12] or H.323 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 5] Internet Draft RTSP March 27, 1997 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 presentation 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 [13]) firewalls. A firewall may need to understand the SETUP method to open a "hole" for the UDP media stream. HTTP-friendly: Where sensible, RTSP re-uses HTTP concepts, so that the existing infrastructure can be re-used. This infrastructure includes JEPI (the Joint Electronic Payment Initiative) for electronic payments and PICS (Platform for Internet Content Selection) for associating labels with content. However, RTSP does not just add methods to HTTP, since the controlling continuous media requires server state in most cases. Appropriate server control: If a client can start a stream, it must be able to stop a stream. Servers should not start streaming to clients in such a way that clients cannot stop the stream. Transport negotiation: The client can negotiate the transport method prior to actually needing to process a continuous media stream. 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 possible. The protocol would not address how several clients negotiate 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 6] Internet Draft RTSP March 27, 1997 servers by necessity will support different sets of requests. For example: o A server may only be capable of playback, not recording and thus has no need to support the RECORD request. o A server may not be capable of seeking (absolute positioning), say, if it is to support live events only. o Some servers may not support setting stream parameters and thus not support GET_PARAMETER and SET_PARAMETER. A server SHOULD implement all header fields described in Section 11. 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, where the methods described in [H19.6] are not likely to be supported across all servers. RTSP can be extended in three ways, listed in order of the magnitude of changes supported: o 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.) o New methods can be added. If the recipient of the message does not understand the request, it responds with error code 501 (Not implemented) and the sender can then attempt an earlier, less functional version. o 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 presentation is made up of are defined by a presentation description file, the format of which is outside the scope of this specification. The presentation description file may be obtained by the client using HTTP or other means such as email and may not necessarily be stored on the media server. For the purposes of this specification, a presentation description is assumed to describe one or more presentations, each of which maintains a common time axis. For simplicity of exposition and H. Schulzrinne, A. Rao, R. Lanphier [Page 7] Internet Draft RTSP March 27, 1997 without loss of generality, it is assumed that the presentation description contains exactly one such presentation. A presentation may contain several media streams. The presentation description file contains a description of the media streams making up the presentation, including their encodings, language, and other parameters that enable the client to choose the most appropriate combination of media. In this presentation description, each media stream that is individually controllable by 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 capable of. Besides the media parameters, the network destination address and port need to be determined. Several modes of operation can be distinguished: Unicast: The media is transmitted to the source of the RTSP request, with the port number chosen by the client. Alternatively, the media is transmitted on the same reliable stream as RTSP. Multicast, server chooses address: The media server picks the multicast address and port. This is the typical case for a live or near-media-on-demand transmission. Multicast, client chooses address: If the server is to participate in an existing multicast conference, the multicast address, port and encryption key are given by the conference description, established by means outside the scope of this specification. 1.7 RTSP States RTSP controls a stream which may be sent via a separate protocol, independent of the control channel. For example, RTSP control may occur on a TCP connection while the data flows via UDP. Thus, data delivery continues even if no RTSP requests are received by the media server. Also, during its lifetime, a single media stream may be controlled by RTSP requests issued sequentially on different TCP connections. Therefore, the server needs to maintain "session state" to be able to correlate RTSP requests with a stream. The state transitions are described in Section 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 8] Internet Draft RTSP March 27, 1997 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. 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. Having the presentation description be returned by the web server makes it possible to have the web server take care of authentication and billing, by handing out a presentation description whose media identifier includes an encrypted version of the requestor's IP address and a timestamp, with a shared secret between web and media server. 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, in that 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 9] Internet Draft RTSP March 27, 1997 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). All the mechanisms specified in this document are described in both prose and an augmented Backus-Naur form (BNF) similar to that used in RFC 2068 [H2.1]. It is described in detail in [14]. In this draft, we use indented and smaller-type paragraphs to provide background and motivation. Some of these paragraphs are marked with HS, AR and RL, designating opinions and comments by the individual authors which may not be shared by the co-authors and require resolution. 3 Protocol Parameters 3.1 RTSP Version applies, with HTTP replaced by RTSP. 3.2 RTSP URL The "rtsp" and "rtspu" schemes are used to refer to network resources via the RTSP protocol. This section defines the scheme-specific syntax and semantics for RTSP URLs. rtsp_URL = ( "rtsp:" | "rtspu:" ) "//" 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 protocol (within the Internet, TCP), while the scheme rtspu identifies an unreliable protocol (within the Internet, UDP). H. Schulzrinne, A. Rao, R. Lanphier [Page 10] Internet Draft RTSP March 27, 1997 If the port is empty or not given, port 554 is assumed. The semantics are that the identified resource can be controlled be 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 [15]). A presentation or a stream is identified by an textual media identifier, using the character set and escape conventions [H3.2] of URLs [16]. Requests described in Section 9 can refer to either the whole presentation or an individual stream within the presentation. Note that some methods can only be applied to streams, not presentations and vice versa. A specific instance of a presentation or stream, e.g., one of several concurrent transmissions of the same content, an RTSP session , is indicated by the Session header field (Section 11.26) where needed. 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 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 individual 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 11] Internet Draft RTSP March 27, 1997 unique. For H.323, the conferenceID value is to be used. conference-id = 1*OCTET ; LWS must be URL-escaped Conference identifiers are used to allow to allow RTSP sessions to obtain parameters from multimedia conferences the media server is participating in. These conferences are created by protocols outside the scope of this specification, e.g., H.323 [17] 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. If the conference participant inviting the media server would only supply a conference identifier which is unique for that inviting party, the media server could add an internal identifier for that party, e.g., its Internet address. However, this would prevent that the conference participant and the initiator of the RTSP commands are two different entities. 3.4 SMPTE Relative Timestamps A SMPTE relative time-stamp 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 , with the origin at the start of the clip. For NTSC, the frame rate is 29.97 frames per second. This is handled by dropping the first frame index of every minute, except every tenth minute. If the frame value is zero, it may be omitted. smpte-range = "smpte" "=" smpte-time "-" [ smpte-time ] smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT [ "." 1*2DIGIT ] Examples: smpte=10:12:33.40- smpte=10:7:33- smpte=10:7:0-10:7:33 H. Schulzrinne, A. Rao, R. Lanphier [Page 12] Internet Draft RTSP March 27, 1997 3.5 Normal Play Time Normal play time (NPT) indicates the stream absolute position relative to the beginning of the presentation, measured in seconds and microseconds. The beginning of a presentation corresponds to 0 seconds and 0 microseconds. Negative values are not defined. The microsecond field is always less than 1,000,000. 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." [18] npt-range = "npt" "=" npt-time "-" [ npt-time ] npt-time = 1*DIGIT [ ":" *DIGIT ] Examples: npt=123:45-125 3.6 Absolute Time Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Fractions of a second may be indicated. 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 Example 4 RTSP Message H. Schulzrinne, A. Rao, R. Lanphier [Page 13] Internet Draft RTSP March 27, 1997 RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2044). 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. 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): 1. Any response message which MUST NOT include a message-body (such as the 1xx, 204, and 304 responses) is always H. Schulzrinne, A. Rao, R. Lanphier [Page 14] Internet Draft RTSP March 27, 1997 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 11.12) 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 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 chunked 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 explicitly. 5 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. Request = Request-line CRLF *request-header CRLF [ message-body ] Request-Line = Method SP Request-URI SP RTSP-Version SP seq-no CRLF Method = "DESCRIBE" ; Section | "GET_PARAMETER" ; Section | "OPTIONS" ; Section | "PAUSE" ; Section | "PLAY" ; Section | "RECORD" ; Section H. Schulzrinne, A. Rao, R. Lanphier [Page 15] Internet Draft RTSP March 27, 1997 | "REDIRECT" ; Section | "SETUP" ; Section | "SET_PARAMETER" ; Section | "TEARDOWN" ; Section | extension-method extension-method = token Request-URI = "*" | absolute_URI RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT seq-no = 1*DIGIT Note that in contrast to HTTP/1.1, RTSP requests always contain the absolute URL (that is, including the scheme, host and port) rather than just the absolute path. 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 6 Response [H6] applies except that HTTP-Version is replaced by RTSP-Version define some HTTP codes. The valid response codes and the methods they can be used with are defined in the table 1. After receiving and interpreting a request message, the recipient responds with an RTSP response message. Response = Status-Line ; Section *( general-header ; Section | response-header ; Section | entity-header ) ; Section CRLF [ message-body ] ; Section H. Schulzrinne, A. Rao, R. Lanphier [Page 16] Internet Draft RTSP March 27, 1997 6.1 Status-Line The first line of a Response message is the Status-Line , consisting of the protocol version followed by a numeric status code, the sequence number of the corresponding request 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. Note that the addition of a Status-Line = RTSP-Version SP Status-Code SP seq-no SP Reason-Phrase CRLF 6.1.1 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 section10. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata and the Reason-Phrase is intended for the human user. The client is not required to examine or display the Reason- Phrase The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. There are 5 values for the first digit: o 1xx: Informational - Request received, continuing process o 2xx: Success - The action was successfully received, understood, and accepted o 3xx: Redirection - Further action must be taken in order to complete the request o 4xx: Client Error - The request contains bad syntax or cannot be fulfilled o 5xx: Server Error - The server failed to fulfill an apparently valid request The individual values of the numeric status codes defined for RTSP/1.0, and an example set of corresponding Reason-Phrase 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 status codes and adds RTSP-specific status codes in the starting at 450 to avoid conflicts with newly H. Schulzrinne, A. Rao, R. Lanphier [Page 17] Internet Draft RTSP March 27, 1997 defined HTTP status codes. Status-Code = "100" ; Continue | "200" ; OK | "201" ; Created | "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 | "409" ; Conflict | "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 | "45x" ; Session Not Found | "45x" ; Method Not Valid in This State | "45x" ; Header Field Not Valid for Resource | "45x" ; Invalid Range | "45x" ; Parameter Is Read-Only | "500" ; Internal Server Error | "501" ; Not Implemented | "502" ; Bad Gateway | "503" ; Service Unavailable | "504" ; Gateway Time-out | "505" ; HTTP Version not supported | extension-code extension-code = 3DIGIT Reason-Phrase = * H. Schulzrinne, A. Rao, R. Lanphier [Page 18] Internet Draft RTSP March 27, 1997 RTSP status codes are extensible. RTSP applications are not required to understand the meaning of all registered status codes, though such understanding is obviously desirable. However, applications MUST understand the class of any status code, as indicated by the first digit, and treat any unrecognized response as being equivalent to the x00 status code of that class, with the exception that an unrecognized response MUST NOT be cached. For example, if an unrecognized status code of 431 is received by the client, it can safely assume that there was something wrong with its request and treat the response as if it had received a 400 status code. In such cases, user agents SHOULD present to the user the entity returned with the response, since that entity is likely to include human- readable information which will explain the unusual status. 6.1.2 Response Header Fields The response-header fields allow the request recipient to pass additional information about the response which cannot be placed in the Status-Line server and about further access to the resource identified by the Request-URI response-header = Location ; Section | Proxy-Authenticate ; Section | Public ; Section | Retry-After ; Section | Server ; Section | Vary ; Section | WWW-Authenticate ; Section 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. 7 Entity Request and Response messages MAY transfer an entity if not otherwise restricted by the request method or response status code. An entity consists of entity-header fields and an entity-body, although some responses will only include the entity-headers. In this section, both sender and recipient refer to either the client H. Schulzrinne, A. Rao, R. Lanphier [Page 19] Internet Draft RTSP March 27, 1997 Code reason _______________________________________________________________ 100 Continue all _______________________________________________________________ 200 OK all 201 Created 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 409 Conflict 410 Gone all 411 Length Required SETUP 412 Precondition Failed 413 Request Entity Too Large SETUP 414 Request-URI Too Long all 415 Unsupported Media Type SETUP 45x Only Valid for Stream SETUP 45x Invalid parameter SETUP 45x Not Enough Bandwidth SETUP 45x Illegal Conference Identifier SETUP 45x Illegal Session Identifier PLAY, RECORD, TEARDOWN 45x Parameter Is Read-Only SET_PARAMETER 45x Header Field Not Valid all _______________________________________________________________ 500 Internal Server Error all 501 Not Implemented all 502 Bad Gateway all 503 Service Unavailable all 504 Gateway Timeout all 505 RTSP Version Not Supported all Table 1: Status codes and their usage with RTSP methods H. Schulzrinne, A. Rao, R. Lanphier [Page 20] Internet Draft RTSP March 27, 1997 or the server, depending on who sends and who receives the entity. 7.1 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 14.7 | Content-Encoding ; Section 14.12 | Content-Language ; Section 14.13 | Content-Length ; Section 14.14 | Content-Type ; Section 14.18 | Expires ; Section 14.21 | Last-Modified ; Section 14.29 | 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. 7.2 Entity Body See [H7.2] 8 Connections RTSP requests can be transmitted in several different ways: o persistent transport connections used for several request- response transactions; o one connection per request/response transaction; o 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 send without setting up a connection. Unlike HTTP, RTSP allows the media server to send requests to the H. Schulzrinne, A. Rao, R. Lanphier [Page 21] Internet Draft RTSP March 27, 1997 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. 8.1 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. 8.2 Reliability and Acknowledgements Requests are acknowledged by the receiver unless they are sent to a multicast 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 TBD), with an initial round-trip value of 500 ms. An implementation MAY cache the last RTT measurement as the initial value for future connections. If a reliable transport protocol is used to carry RTSP, the timeout value MAY be set to an arbitrarily large value. This can greatly increase responsiveness for proxies operating in local-area networks with small RTTs. The mechanism is defined such that the client implementation does not have be aware of whether a reliable or unreliable transport protocol is being used. It is probably a bad idea to have two reliability mechanisms on top of each other, although the RTSP RTT estimate is likely to be larger than the TCP estimate. Each request carries a sequence number, which is incremented by one for each request transmitted. If a request is repeated because of lack of acknowledgement, the sequence number is incremented. This avoids ambiguities when computing round-trip time estimates. [TBD: An initial sequence number negotiation needs to be added for UDP; otherwise, a new stream connection may see a request be acknowledged by a delayed response from an earlier "connection". This handshake can be avoided with a sequence number containing a timestamp of sufficiently high resolution.] The reliability mechanism described here does not protect against reordering. This may cause problems in some instances. For example, a TEARDOWN followed by a PLAY has quite a different effect than the reverse. Similarly, if a PLAY request arrives before all parameters H. Schulzrinne, A. Rao, R. Lanphier [Page 22] Internet Draft RTSP March 27, 1997 are set due to reordering, the media server would have to issue an error indication. Since sequence numbers for retransmissions are incremented (to allow easy RTT estimation), the receiver cannot just ignore out-of-order packets. [TBD: This problem could be fixed by including both a sequence number that stays the same for retransmissions and a timestamp for RTT estimation.] 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 method header MUST contain a Content-Length whenever that method contains a payload. Otherwise, an RTSP packet is terminated with an empty line immediately following the method header. 9 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 method direction object requirement ________________________________________________________ DESCRIBE C -> S, S -> C P,S recommended GET_PARAMETER C -> S, S -> C P,S optional OPTIONS C -> S P,S required PAUSE C -> S P,S recommended PLAY C -> S P,S required RECORD C -> S P,S optional REDIRECT S -> C P,S optional SETUP C -> S S required SET_PARAMETER C -> S, S -> C P,S optional TEARDOWN C -> S P,S required Table 2: Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on Notes on Table 2: PAUSE is recommend, 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 23] Internet Draft RTSP March 27, 1997 SHOULD not try this method again for this server. 9.1 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 non-standard request. It does not influence server state. In addition, if the optional Require header is present, option tags within the header indicate features needed by the requestor that are not required at the version level of the protocol. Example 1: C->S: OPTIONS * RTSP/1.0 1 Require: implicit-play, record-feature Transport-Require: switch-to-udp-control, gzipped-messages Note that these are fictional features (though we may want to make them real one day). Example 2 (using RFC2069-style authentication only as an example): S->C: OPTIONS * RTSP/1.0 1 Authenticate: Digest realm="testrealm@host.com", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", opaque="5ccc069c403ebaf9f0171e9517f40e41" S->C: RTSP/1.0 200 1 OK Date: 23 Jan 1997 15:35:06 GMT Nack-Transport-Require: switch-to-udp-control Note that these are fictional features (though we may want to make them real one day). Example 2 (using RFC2069-style authentication only as an example): C->S: RTSP/1.0 401 1 Unauthorized Authorization: Digest username="Mufasa", realm="testrealm@host.com", nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093", uri="/dir/index.html", H. Schulzrinne, A. Rao, R. Lanphier [Page 24] Internet Draft RTSP March 27, 1997 response="e966c932a9242554e42c8ee200cec7f6", opaque="5ccc069c403ebaf9f0171e9517f40e41" 9.2 DESCRIBE The DESCRIBE method retrieves the description of a presentation or media object identified by the request URL from a server. It may use the Accept header to specify the description formats that the client understands. The server responds with a description of the requested resource. Alternatively, the server may "push" a new description to the client, for example, if a new stream has become available. 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: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 312 Accept: application/sdp, application/rtsl, application/mheg S->C: RTSP/1.0 200 312 OK 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 or S->C: RTSP/1.0 200 312 OK Date: 23 Jan 1997 15:35:06 GMT H. Schulzrinne, A. Rao, R. Lanphier [Page 25] Internet Draft RTSP March 27, 1997 Content-Type: application/rtsl Content-Length: 2782 <2782 octets of data containing stream description> Server to client example: S->C: DESCRIBE /twister RTSP/1.0 902 Session: 1234 Content-Type: application/rtsl new RTSL presentation description 9.3 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. 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. This avoids having firewall to parse numerous different presentation description formats, for information which is irrelevant. If the optional Require header is present, option tags within the header indicate features needed by the requestor that are not required at the version level of the protocol. The Transport-Require header is used to indicate proxy-sensitive features that MUST be stripped by the proxy to the server if not supported. Furthermore, any Transport-Require header features that are not supported by the proxy MUST be negatively acknowledged by the proxy to the client if not supported. HS: In my opinion, the Require header should be replaced by PEP since PEP is standards-track, has more functionality and somebody already did the work. The Transport header specifies the transport parameters acceptable to the client for data transmission; the response will contain the transport parameters selected by the server. H. Schulzrinne, A. Rao, R. Lanphier [Page 26] Internet Draft RTSP March 27, 1997 C->S: SETUP foo/bar/baz.rm RTSP/1.0 302 Transport: rtp/udp;port=458 S->C: RTSP/1.0 200 302 OK Date: 23 Jan 1997 15:35:06 GMT Transport: cush/udp;port=458 9.4 PLAY The PLAY method tells the server to start sending data via the mechanism specified in SETUP. A client MUST NOT issue a PLAY request until any outstanding SETUP requests have been acknowledged as successful. 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 commands in the example below arrive, the server will play first second 10 through 15 and then, immediately following, seconds 20 to 25 and finally seconds 30 through the end. C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 835 Range: npt=10-15 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 836 Range: npt=20-25 C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 837 Range: npt=30- See the description of the PAUSE request for further examples. A PLAY request without a Range header is legal. It starts playing a stream from the beginning unless the stream has been paused. If a stream has been paused via PAUSE, stream delivery resumes at the pause point. If a stream is playing, such a PLAY request causes no further action and can be used by the client to test server liveness. H. Schulzrinne, A. Rao, R. Lanphier [Page 27] Internet Draft RTSP March 27, 1997 The Range header may also contain a time parameter. This parameter specifies a time in UTC at which the playback should start. If the message is received after the specified time, playback is started immediately. The time parameter may be used to aid in synchronisation of streams obtained from different sources. For a on-demand stream, the server replies back with the actual range that will be played back. This may differ from the requested range if alignment of the requested range to valid frame boundaries is required for the media source. If no range is specified in the request, the current position is returned in the reply. The unit of the range in the reply is the same as that in the request. After playing the desired range, the presentation is automatically paused, as if a PAUSE request had been issued. The following example plays the whole presentation starting at SMPTE time code 0:10:20 until the end of the clip. The playback is to start at 15:36 on 23 Jan 1997. C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 833 Range: smpte=0:10:20-;time=19970123T153600Z S->C: RTSP/1.0 200 833 OK 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/1.0 835 Range: clock=19961108T142300Z-19961108T143520Z S->C: RTSP/1.0 200 833 OK Date: 23 Jan 1997 15:35:06 GMT A media server only supporting playback MUST support the smpte format and MAY support the clock format. 9.5 PAUSE H. Schulzrinne, A. Rao, R. Lanphier [Page 28] Internet Draft RTSP March 27, 1997 The PAUSE request causes the stream delivery to be interrupted (halted) temporarily. If the request URL names a stream, only playback and recording of that stream is halted. For example, for audio, this is equivalent to muting. If the request URL names a presentation or group of streams, delivery of all currently active streams within the presentation or group is halted. After resuming playback or recording, synchronization of the tracks MUST be maintained. Any server resources are kept. The PAUSE request may contain a Range header specifying when the stream or presentation is to be halted. The header must contain exactly one value rather than a time range. The normal play time for the stream is set to that value. The pause request becomes effective the first time the server is encountering the time point specified. If this header is missing, stream delivery is interrupted immediately on receipt of the message. 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, it stops immediately. If the pause request is for NPT 16, it stops after completing the first play request and discards the second play request. As another example, if a server has received requests to play ranges 10 to 15 and then 13 to 20, that is, overlapping ranges, the PAUSE request for NPT=14 would take effect while playing 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. Example: C->S: PAUSE /fizzle/foo RTSP/1.0 834 S->C: RTSP/1.0 200 834 OK Date: 23 Jan 1997 15:35:06 GMT H. Schulzrinne, A. Rao, R. Lanphier [Page 29] Internet Draft RTSP March 27, 1997 9.6 TEARDOWN Stop the stream delivery for the given URI, freeing the resources associated with it. If the URI is the root node 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 /fizzle/foo RTSP/1.0 892 S->C: RTSP/1.0 200 892 OK 9.7 GET_PARAMETER The requests retrieves the value of a parameter of a presentation or stream specified in the URI. Multiple parameters can be requested in the message body using the content type text/rtsp-parameters Note that parameters include server and client statistics. IANA registers parameter names for statistics and other purposes. GET_PARAMETER with no entity body may be used to test client or server liveness ("ping"). Example: S->C: GET_PARAMETER /fizzle/foo RTSP/1.0 431 Content-Type: text/rtsp-parameters Session: 1234 Content-Length: 15 packets_received jitter C->S: RTSP/1.0 200 431 OK Content-Length: 46 Content-Type: text/rtsp-parameters packets_received: 10 jitter: 0.3838 H. Schulzrinne, A. Rao, R. Lanphier [Page 30] Internet Draft RTSP March 27, 1997 9.8 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. A server MUST allow a parameter to be set repeatedly to the same value, but it MAY disallow changing parameter values. Note: transport parameters for the media stream MUST only be set with the SETUP command. Restricting setting transport parameters to SETUP is for the benefit of firewalls. The parameters are split in a fine-grained fashion so that there can be more meaningful error indications. However, it may make sense to allow the setting of several parameters if an atomic setting is desirable. Imagine device control where the client does not want the camera to pan unless it can also tilt to the right angle at the same time. A SET_PARAMETER request without parameters can be used as a way to detect client or server liveness. Example: C->S: SET_PARAMETER /fizzle/foo RTSP/1.0 421 Content-type: text/rtsp-parameters fooparam: foostuff barparam: barstuff S->C: RTSP/1.0 450 421 Invalid Parameter Content-Length: 6 barparam 9.9 REDIRECT A redirect request informs the client that it must connect to another server location. It contains the mandatory header Location, which indicates that the client should issue a DESCRIBE for that URL. It H. Schulzrinne, A. Rao, R. Lanphier [Page 31] Internet Draft RTSP March 27, 1997 may contain the parameter Range, which indicates when the redirection takes effect. This example request redirects traffic for this URI to the new server at the given play time: S->C: REDIRECT /fizzle/foo RTSP/1.0 732 Location: rtsp://bigserver.com:8001 Range: clock=19960213T143205Z- 9.10 RECORD This method initiates recording a range of media data according to the presentation description. The timestamp reflects start and end time (UTC). If no time range is given, use the start or end time provided in the presentation description. If the session has already started, commence recording immediately. The Conference header is mandatory. The server decides whether to store the recorded data under the request-URI or another URI. If the server does not use the request- URI, the response SHOULD be 201 (Created) and contain an entity which describes the status of the request and refers to the new resource, and a Location header. A media server supporting recording of live presentations MUST support the clock range format; the smpte format does not make sense. In this example, the media server was previously invited to the conference indicated. C->S: RECORD /meeting/audio.en RTSP/1.0 954 Session: 1234 Conference: 128.16.64.19/32492374 9.11 Embedded Binary Data Binary packets such as RTP data are encapsulated by an ASCII dollar sign (24 decimal), followed by a one-byte session identifier, followed by the length of the encapsulated binary data as a binary, two-byte integer in network byte order. The binary data follows immediately afterwards, without a CRLF. H. Schulzrinne, A. Rao, R. Lanphier [Page 32] Internet Draft RTSP March 27, 1997 10 Status Code Definitions Where applicable, HTTP status [H10] codes are re-used. 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 request. 10.1 Redirection 3xx See [H10.3]. Within RTSP, redirection may be used for load balancing or redirecting stream requests to a server topologically closer to the client. Mechanisms to determine topological proximity are beyond the scope of this specification. 10.2 Client Error 4xx 10.2.1 451 Parameter Not Understood The recipient of the request does not support one or more parameters contained in the request. 10.2.2 452 Conference Not Found The conference indicated by a Conference header field is unknown to the media server. 10.2.3 453 Not Enough Bandwidth The request was refused since there was insufficient bandwidth. This may, for example, be the result of a resource reservation failure. 10.2.4 45x Session Not Found The RTSP session identifier is invalid or has timed out. 10.2.5 45x Method Not Valid in This State The client or server cannot process this request in its current state. 10.2.6 45x 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. 10.2.7 45x Invalid Range H. Schulzrinne, A. Rao, R. Lanphier [Page 33] Internet Draft RTSP March 27, 1997 The Range value given is out of bounds, e.g., beyond the end of the presentation. 10.2.8 45x Parameter Is Read-Only The parameter to be set by SET_PARAMETER can only be read, but not modified. 11 Header Field Definitions HTTP/1.1 or other, non-standard header fields not listed here currently have no well-defined meaning and SHOULD be ignored by the recipient. Tables 3 summarizes the header fields used by RTSP. Type "R" designates request headers, type "r" response headers. 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 'r' will be send in every request of this type; merely, that client (for response headers) and server (for request headers) MUST implement them. 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. If the field content does not apply to the particular resource, the server MUST return status 45x (Header Field Not Valid for Resource). 11.1 Accept The Accept request-header field can be used to specify certain presentation description content types which are acceptable for the response. The "level" parameter for presentation descriptions is properly defined as part of the MIME type registration, not here. See [H14.1] for syntax. Example of use: Accept: application/rtsl, application/sdp;level=2 H. Schulzrinne, A. Rao, R. Lanphier [Page 34] Internet Draft RTSP March 27, 1997 Header type support methods _________________________________________________________________ Accept R opt. entity Accept-Encoding R opt. entity Accept-Language R opt. all Authorization R opt. all Bandwidth R opt. SETUP Blocksize R opt. all but OPTIONS, TEARDOWN Cache-Control Rr opt. SETUP Conference R opt. SETUP Connection Rr req. all Content-Encoding R req. SET_PARAMETER Content-Encoding r req. DESCRIBE Content-Length R req. SET_PARAMETER Content-Length r req. entity Content-Type R req. SET_PARAMETER Content-Type r req. entity Date Rr opt. all Expires r opt. DESCRIBE If-Modified-Since R opt. DESCRIBE, SETUP Last-Modified r opt. entity 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 Scale Rr opt. PLAY, RECORD Session Rr req. all but SETUP, OPTIONS Server r opt. all Speed Rr opt. PLAY Transport Rr req. SETUP Transport-Require R xeq. all User-Agent R opt. all Via Rr opt. all WWW-Authenticate r opt. all Table 3: Overview of RTSP header fields 11.2 Accept-Encoding See [H14.3] 11.3 Accept-Language See [H14.4]. Note that the language specified applies to the presentation description and any reason phrases, not the media H. Schulzrinne, A. Rao, R. Lanphier [Page 35] Internet Draft RTSP March 27, 1997 content. 11.4 Allow The Allow response header field lists the methods supported by the resource identified by the request-URI. The purpose of this field is to strictly inform the recipient of valid methods associated with the resource. An Allow header field must be present in a 405 (Method not allowed) response. Example of use: Allow: SETUP, PLAY, RECORD, SET_PARAMETER 11.5 Authorization See [H14.8] 11.6 Bandwidth The Bandwidth request header field describes the estimated bandwidth available to the client, expressed as a positive integer and measured in bits per second. Bandwidth = "Bandwidth" ":" 1*DIGIT Example: Bandwidth: 4000 11.7 Blocksize This request header field is sent from the client to the media server asking the server for a particular media packet size. This packet size does not include lower-layer headers such as IP, UDP, or RTP. The server is free to use a blocksize which is lower than the one requested. The server MAY truncate this packet size to the closest multiple of the minimum media-specific block size or overrides it with the media specific size if necessary. The block size is a strictly positive decimal number and measured in octets. The server only returns an error (416) if the value is syntactically invalid. H. Schulzrinne, A. Rao, R. Lanphier [Page 36] Internet Draft RTSP March 27, 1997 11.8 Cache-Control The Cache-Control general header field is used to specify directives that MUST be obeyed by all caching mechanisms along the request/response chain. Cache directives must be passed through by a proxy or gateway application, regardless of their significance to that application, since the directives may be applicable to all recipients along the request/response chain. It is not possible to specify a cache- directive for a specific cache. Cache-Control should only be specified in a SETUP request and its response. Note: Cache-Control does not govern the caching of responses as for HTTP, but rather of the stream identified by the SETUP request. Responses to RTSP requests are not cacheable. [HS: Should there be an exception for DESCRIBE?] Cache-Control = "Cache-Control" ":" 1#cache-directive cache-directive = cache-request-directive | cache-response-directive cache-request-directive = "no-cache" | "max-stale" | "min-fresh" | "only-if-cached" | cache-extension cache-response-directive = "public" | "private" | "no-cache" | "no-transform" | "must-revalidate" | "proxy-revalidate" | "max-age" "=" delta-seconds | cache-extension cache-extension = token [ "=" ( token | quoted-string ) ] no-cache: Indicates that the media stream MUST NOT be cached anywhere. This allows an origin server to prevent caching even H. Schulzrinne, A. Rao, R. Lanphier [Page 37] Internet Draft RTSP March 27, 1997 by caches that have been configured to return stale responses to client requests. public: Indicates that the media stream is cachable 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 certain stream. A proxy might, for example, convert between video formats to save cache space or to reduce the amount of traffic on a slow link. Serious operational problems may occur, however, when these transformations have been applied to streams intended for certain kinds of applications. For example, applications for medical imaging, scientific data analysis and those using end-to-end authentication, all depend on receiving a stream that is bit for bit identical to the original entity-body. Therefore, if a response includes the no-transform directive, an intermediate cache or proxy MUST NOT change the encoding of the stream. Unlike HTTP, RTSP does not provide for partial transformation at this point, e.g., allowing translation into a different language. only-if-cached: In some cases, such as times of extremely poor network connectivity, a client may want a cache to return only those media streams that it currently has stored, and not to receive these from the origin server. To do this, the client may include the only-if-cached directive in a request. If it receives this directive, a cache SHOULD either respond using a cached media stream that is consistent with the other constraints of the request, or respond with a 504 (Gateway Timeout) status. However, if a group of caches is being operated as a unified system with good internal connectivity, such a request MAY be forwarded within that group of caches. max-stale: Indicates that the client is willing to accept a media stream that has exceeded its expiration time. If max-stale is assigned a value, then the client is willing to accept a response that has exceeded its expiration time by no more than the specified number of seconds. If no value is assigned to max-stale, then the client is willing to accept a stale response of any age. min-fresh: Indicates that the client is willing to accept a media stream whose freshness lifetime is no less than its current age plus the specified time in seconds. That is, the client wants a H. Schulzrinne, A. Rao, R. Lanphier [Page 38] Internet Draft RTSP March 27, 1997 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. (I.e., 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.) 11.9 Conference This request header field establishes a logical connection between a conference, established using non-RTSP means, 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 11.10 Connection See [H14.10]. 11.11 Content-Encoding See [H14.12] 11.12 Content-Length This field contains the length of the content of the method (i.e. after the double CRLF following the last header). Unlike HTTP, it MUST be included in all messages that carry content beyond the header portion of the message. It is interpreted according to [H14.14]. 11.13 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. H. Schulzrinne, A. Rao, R. Lanphier [Page 39] Internet Draft RTSP March 27, 1997 11.14 Date See [H14.19]. 11.15 Expires The Expires entity-header field gives the date/time after which the media-stream should be considered stale. A stale cache entry may not normally be returned by a cache (either a proxy cache or an user agent cache) unless it is first validated with the origin server (or with an intermediate cache that has a fresh copy of the entity). See section 13.2 for further discussion of the expiration model. The presence of an Expires field does not imply that the original resource will change or cease to exist at, before, or after that time. The format is an absolute date and time as defined by HTTP-date in [H3.3]; it MUST be in RFC1123-date format: Expires = "Expires" ":" HTTP-date An example of its use is Expires: Thu, 01 Dec 1994 16:00:00 GMT RTSP/1.0 clients and caches MUST treat other invalid date formats, especially including the value "0", as in the past (i.e., "already expired"). To mark a response as "already expired," an origin server should use an Expires date that is equal to the Date header value. To mark a response as "never expires," an origin server should use an Expires date approximately one year from the time the response is sent. RTSP/1.0 servers should not send Expires dates more than one year in the future. The presence of an Expires header field with a date value of some time in the future on a media stream that otherwise would by default be non-cacheable indicates that the media stream is cachable, unless indicated otherwise by a Cache-Control header field (Section 11.8. H. Schulzrinne, A. Rao, R. Lanphier [Page 40] Internet Draft RTSP March 27, 1997 11.16 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 setup ( 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 11.17 Last-modified The Last-Modified entity-header field indicates the date and time at which the origin server believes the variant was last modified. See [H14.29]. If the request URI refers to an aggregate, the field indicates the last modification time across all leave nodes of that aggregate. 11.18 Location See [H14.30]. 11.19 Nack-Transport-Require Negative acknowledgement of features not supported by the server. If there is a proxy on the path between the client and the server, the proxy MUST insert a message reply with an error message 506 (Feature not supported). HS: Same caveat as for Require applies. 11.20 Range This request header field specifies a range of time. The range can be specified in a number of units. This specification defines the smpte (see Section 3.4) and clock (see Section 3.6) range units. Within H. Schulzrinne, A. Rao, R. Lanphier [Page 41] Internet Draft RTSP March 27, 1997 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. 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 header. It allows to select a clip from the media object, to play from a given point to the end and from the current location to a given point. 11.21 Require The Require header is used by clients to query the server about features that it may or may not support. The server MUST respond to this header by negatively acknowledging those features which are NOT supported in the Unsupported header. HS: Naming of features -- yet another name space. I believe this header field to be redundant. PEP should be used instead. For example C->S: SETUP /foo/bar/baz.rm RTSP/1.0 302 Require: funky-feature Funky-Parameter: funkystuff S->C: RTSP/1.0 200 506 Option not supported Unsupported: funky-feature C->S: SETUP /foo/bar/baz.rm RTSP/1.0 303 S->C: RTSP/1.0 200 303 OK H. Schulzrinne, A. Rao, R. Lanphier [Page 42] Internet Draft RTSP March 27, 1997 This is to make sure that the client-server interaction will proceed optimally when all options are understood by both sides, and only slow down if options aren't 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 doesn't understand. 11.22 Retry-After See [H14.38]. 11.23 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 43] Internet Draft RTSP March 27, 1997 11.24 Speed This request header fields parameter requests the server to deliver data to the client at a particular speed, contingent on the server's ability and desire to serve the media stream at the given speed. Implementation by the server is OPTIONAL. The default is the bit rate of the stream. The parameter value is expressed as a decimal ratio, e.g., a value of 2.0 indicates that data is to be delivered twice as fast as normal. A speed of zero is invalid. A negative value indicates that the stream is to be played back in reverse direction. HS: With 'Scale', the negative value is redundant and should probably be removed since it only leads to possible conflicts when Scale is positive and Speed negative. If the request contains a Range parameter, the new speed value will take effect at that time. Speed = "Speed" ":" [ "-" ] 1*DIGIT [ "." *DIGIT ] Example: Speed: 2.5 11.25 Server See [H14.39] 11.26 Session This request and response header field identifies an RTSP session, started by the media server in a SETUP response and concluded by TEARDOWN on the presentation URL. The session identifier is chosen by the media server and has the same syntax as a conference identifier. Once a client receives a Session identifier, it MUST return it for any request related to that session. HS: This may be redundant with the standards-track HTTP state maintenance mechanism [2]. The equivalent way of H. Schulzrinne, A. Rao, R. Lanphier [Page 44] Internet Draft RTSP March 27, 1997 doing this would be for the server to send Set-Cookie: Session="123"; Version=1; Path = "/twister" and for the client to return later Cookie: Session = "123"; $Version=1; $Path = "/twister" response to the TEARDOWN message, the server would simply send Set-Cookie: Session="123"; Version=1; Max-Age=0 to get rid of the cookie on the client side. Cookies also have a time-out, so that a server may limit the lifetime of a session at will. Unlike a web browser, a client would not store these states on disk. To avoid privacy issues, we should prohibit the Host parameter. 11.27 Transport This request header indicates which transport protocol is to be used and configures its parameters such as multicast, compression, multicast time-to-live and destination port for a single stream. It sets those values not already determined by a presentation description. In some cases, the presentation description contains all necessary information. In those cases, a Transport header field (and the SETUP request containing it) are not needed. in whatever protocol is being used by the control stream. Currently, the next-layer protocols RTP is defined. Parameters may be added to each protocol, separated by a semicolon. For RTP, the boolean parameter compressed is defined, indicating compressed RTP according to RFC XXXX. For multicast UDP, the integer parameter ttl defines the time-to-live value to be used. The client may specify the multicast address with the multicast parameter. A server SHOULD authenticate the client before allowing the client to direct a media stream to a multicast address not chosen by the server to avoid becoming the unwitting perpetrator of a denial-of-service attack. For UDP and TCP, the parameter port defines the port data is to be sent to. The SSRC parameter indicates the RTP SSRC 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. The Transport header MAY also be used to change certain transport parameters. A server MAY refuse to change parameters of an existing stream. The server MAY return a Transport response header in the response to indicate the values actually chosen. A Transport request header field may contain a list of transport H. Schulzrinne, A. Rao, R. Lanphier [Page 45] Internet Draft RTSP March 27, 1997 options acceptable to the client. In that case, the server MUST return a single option which was actually chosen. The Transport header field makes sense only for an individual media stream, not a presentation. Transport = "Transport" ":" 1#transport-protocol/upper-layer *parameter transport-protocol = "UDP" | "TCP" upper-layer = "RTP" parameters = ";" "multicast" [ "=" mca ] | ";" "compressed" | ";" "interleaved" | ";" "ttl" "=" ttl | ";" "port" "=" port | ";" "ssrc" "=" ssrc ttl = 1*3(DIGIT) port = 1*5(DIGIT) ssrc = 8*8(HEX) mca = host Example: Transport: udp/rtp;compressed;ttl=127;port=3456 11.28 Transport-Require The Transport-Require header is used to indicate proxy-sensitive features that MUST be stripped by the proxy to the server if not supported. Furthermore, any Transport-Require header features that are not supported by the proxy MUST be negatively acknowledged by the proxy to the client if not supported. See Section 11.21 for more details on the mechanics of this message and a usage example. HS: Same caveat as for Require applies. 11.29 Unsupported See Section 11.21 for a usage example. HS: same caveat as for Require applies. H. Schulzrinne, A. Rao, R. Lanphier [Page 46] Internet Draft RTSP March 27, 1997 11.30 User-Agent See [H14.42] 11.31 Via See [H14.44]. 11.32 WWW-Authenticate See [H14.46]. 12 Caching In HTTP, response-request pairs are cached. RTSP differs significantly in that respect. Responses are not cachable, with the exception of the stream description returned by DESCRIBE. (Since the responses for anything but DESCRIBE and GET_PARAMETER do not return any data, caching is not really an issue for these requests.) However, it is desirable for the continuous media data, typically delivered out-of-band with respect to RTSP, to be cached. On receiving a SETUP or PLAY request, the proxy would ascertain as to whether it has an up-to-date copy of the continuous media content. If not, it would modify the SETUP transport parameters as appropriate and forward the request to the origin server. Subsequent control commands such as PLAY or PAUSE would pass the proxy unmodified. The proxy would then pass the continuous media data to the client, while possibly making a local copy for later re-use. The exact behavior allowed to the cache is given by the cache-response directives described in Section 11.8. A cache MUST answer any DESCRIBE requests if it is currently serving the stream to the requestor, as it is possible that low-level details of the stream description may have changed on the origin-server. Note that an RTSP cache, unlike the HTTP cache, is of the "cut- through" variety. Rather than retrieving the whole resource from the origin server, the cache simply copies the streaming data as it passes by on its way to the client, thus, it does not introduce additional latency. To the client, an RTSP proxy cache would appear like a regular media server, to the media origin server like a client. Just like an HTTP cache has to store the content type, content language, etc. for the objects it caches, a media cache has to store the presentation description. Typically, a cache would eliminate all transport- references (that is, multicast information) from the presentation description, since these are independent of the data delivery from H. Schulzrinne, A. Rao, R. Lanphier [Page 47] Internet Draft RTSP March 27, 1997 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. 13 Examples The following examples reference stream description formats that are not finalized, such as RTSL and SDP. Please do not use these examples as a reference for those formats. 13.1 Media on Demand (Unicast) Client C requests a movie from media servers A ( audio.example.com ) and V ( video.example.com ). The media description is stored on a web server W. The media description contains descriptions of the presentation and all its streams, including the codecs that are available, dynamic RTP payload types, the protocol stack and content information such as language or copyright restrictions. It may also give an indication about the time line of the movie. In our example, the client is only interested in the last part of the movie. The server requires authentication for this movie. The audio track can be dynamically switched between between two sets of encodings. The URL with scheme rtpsu indicates the media servers want to use UDP for exchanging RTSP messages. C->W: DESCRIBE /twister HTTP/1.1 Host: www.example.com Accept: application/rtsl; application/sdp W->C: 200 OK Content-Type: application/rtsl H. Schulzrinne, A. Rao, R. Lanphier [Page 48] Internet Draft RTSP March 27, 1997 C->A: SETUP rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 1 Transport: rtp/udp;compression;port=3056 A->C: RTSP/1.0 200 1 OK Session: 1234 C->V: SETUP rtsp://video.example.com/twister/video RTSP/1.0 1 Transport: rtp/udp;compression;port=3058 V->C: RTSP/1.0 200 1 OK Session: 1235 C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0 2 Session: 1235 Range: smpte=0:10:00- V->C: RTSP/1.0 200 2 OK C->A: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 2 Session: 1234 Range: smpte=0:10:00- A->C: 200 2 OK C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 3 Session: 1234 A->C: 200 3 OK C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0 3 Session: 1235 V->C: 200 3 OK Even though the audio and video track are on two different servers, may start at slightly different times and may drift with respect to each other, the client can synchronize the two using standard RTP methods, in particular the time scale contained in the RTCP sender reports. 13.2 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 H. Schulzrinne, A. Rao, R. Lanphier [Page 49] Internet Draft RTSP March 27, 1997 description, while the media server M maintains the full description. During the RTSP session, a new subtitling stream is added. C->W: GET /concert HTTP/1.1 Host: www.example.com W->C: HTTP/1.1 200 OK Content-Type: application/rtsl C->M: DESCRIBE rtsp://live.example.com/concert/audio RTSP/1.0 1 M->C: RTSP/1.0 200 1 OK Content-Type: application/rtsl C->M: SETUP rtsp://live.example.com/concert/audio RTSP/1.0 2 Transport: multicast=224.2.0.1; port=3456; ttl=16 C->M: PLAY rtsp://live.example.com/concert/audio RTSP/1.0 3 Range: smpte 1:12:0 M->C: RTSP/1.0 405 3 No positioning possible M->C: DESCRIBE concert RTSP/1.0 Content-Type: application/rtsl C->M: PLAY rtsp://live.example.com/concert/lyrics RTSP/1.0 The attempt to position the stream fails since this is a live presentation. 13.3 Playing media into an existing session H. Schulzrinne, A. Rao, R. Lanphier [Page 50] Internet Draft RTSP March 27, 1997 A conference participant C wants to have the media server M play back a demo tape into an existing conference. When retrieving the presentation description, 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: GET /demo HTTP/1.1 Host: www.example.com Accept: application/rtsl, application/sdp M->C: HTTP/1.1 200 1 OK Content-type: application/rtsl C->M: SETUP rtsp://server.example.com/demo/548/sound RTSP/1.0 2 Conference: 218kadjk 13.4 Recording The conference participant C asks the media server M to record a meeting. If the presentation description contains any alternatives, the server records them all. C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 89 Content-Type: application/sdp v=0 s=Mbone Audio i=Discussion of Mbone Engineering Issues M->C: 415 89 Unsupported Media Type Accept: application/rtsl C->M: DESCRIBE rtsp://server.example.com/meeting RTSP/1.0 90 Content-Type: application/rtsl M->C: 200 90 OK C->M: RECORD rtsp://server.example.com/meeting RTSP/1.0 91 H. Schulzrinne, A. Rao, R. Lanphier [Page 51] Internet Draft RTSP March 27, 1997 Range: clock 19961110T1925-19961110T2015 14 Syntax The RTSP syntax is described in an augmented Backus-Naur form (BNF) as used in RFC 2068 (HTTP/1.1). 14.1 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 = "(" | ")" | "<" | ">" | "@" | "," | ";" | ":" | " | "/" | "[" | "]" | "?" | "=" | "{" | "}" | SP | HT token = 1* quoted-string = ( <"> *(qdtext) <"> ) qdtext = > quoted-pair = " message-header = field-name ":" [ field-value ] CRLF field-name = token field-value = *( field-content | LWS ) field-content = 15 Security Considerations H. Schulzrinne, A. Rao, R. Lanphier [Page 52] Internet Draft RTSP March 27, 1997 The protocol offers the opportunity for a remote-control denial-of- service attack. The attacker, using a forged source IP address, can ask for a stream to be played back to that forged IP address. Since there is no relation between a transport layer connection and an RTSP session, it is possible for a malicious client to issue requests with random session identifiers which would affect unsuspecting clients. This does not require spoofing of network packet addresses. The server SHOULD use a large random session identifier to make this attack more difficult. Both problems can be be prevented by appropriate authentication. In addition, the security considerations outlined in [H15] apply. A RTSP Protocol State Machines The RTSP client and server state machines describe the behavior of the protocol from RTSP session initialization through RTSP session termination. [TBD: should we allow for the trivial case of a server that only implements the PLAY message, with no control.] State is defined on a per object basis. An object is uniquely identified by the stream URL and the RTSP session identifier. (A server may choose to generate dynamic presentation descriptions where the URL is unique for a particular RTSP session and thus may not need an explicit RTSP session identifier in the request header.) Any request/reply using URLs denoting an RTSP session comprised of multiple streams will have an effect on the individual states of all the substreams. For example, if the stream /movie contains two substreams /movie/audio and /movie/video, then the following command: PLAY /movie RTSP/1.0 559 Session: 12345 will have an effect on the states of movie/audio and movie/video. This example does not imply a standard way to represent substreams in URLs or a relation to the filesystem. See Section 3.2. The requests OPTIONS, DESCRIBE, GET_PARAMETER, SET_PARAMETER do H. Schulzrinne, A. Rao, R. Lanphier [Page 53] Internet Draft RTSP March 27, 1997 not have any effect on client or server state and are therefore not listed in the state tables. Client and servers MUST disregard messages with a sequence number less than the last one. If no message has been received, the first received message's sequence number will be the starting point. A.1 Client State Machine The client can assume the following states: Init: SETUP has been sent, waiting for reply. Ready: SETUP reply received OR after playing, PAUSE reply received. Playing: PLAY reply received Recording: RECORD reply received The client changes state on receipt of replies to requests. If no explicit SETUP is required for the object (for example, it is available via a multicast group), state begins at READY. In this case, there are only two states, READY and PLAYING. The "next state" column indicates the state assumed after receiving a success response (2xx). If a request yields a status code greater or equal to 300, the client state becomes Init, with the exception of status codes 401 (Unauthorized) and 402 (Payment Required), where the state remains unchanged and the request should be re-issued with the appropriate authentication or payment information. Messages not listed for each state MUST NOT be issued by the client in that state, with the exception of messages not affecting state, as listed above. Receiving a REDIRECT from the server is equivalent to receiving a 3xx redirect status from the server. HS: Depends on allowing PLAY without SETUP. After 4xx or 5xx error, do we go back to Init? state message next state _______________________________________________________ Init SETUP Ready TEARDOWN Init Ready PLAY Playing RECORD Recording TEARDOWN Init Playing PAUSE Ready TEARDOWN Init H. Schulzrinne, A. Rao, R. Lanphier [Page 54] Internet Draft RTSP March 27, 1997 PLAY Playing RECORD Recording SETUP Playing (changed transport) Recording PAUSE Init TEARDOWN Init PLAY Playing RECORD Recording SETUP Recording (changed transport) A.2 Server State Machine The server can assume the following states: Init: The initial state, no valid SETUP received. Ready: Last SETUP received was successful, reply sent or after playing, last PAUSE received was successful, reply sent. Playing: Last PLAY received was successful, reply sent. Data is being sent. Recording: The server is recording media data. The server changes state on receiving requests. If the server is in state Playing or Recording and in unicast mode, it MAY revert to Init and tear down the RTSP session if it has not received "wellness" information, such as RTCP reports, from the client for a defined interval, with a default of one minute. If the server is in state Ready, it MAY revert to Init if it does not receive an RTSP request for an interval of more than one minute. The REDIRECT message, when sent, is effective immediately. If a similar change of location occurs at a certain time in the future, this is assumed to be indicated by the presentation description. SETUP is valid in states Init and Ready only. An error message should be returned in other cases. If no explicit SETUP is required for the object, state starts at READY, there are only two states READY and PLAYING. state message next state ___________________________________ Init SETUP Ready TEARDOWN Init Ready PLAY Playing SETUP Ready TEARDOWN Ready H. Schulzrinne, A. Rao, R. Lanphier [Page 55] Internet Draft RTSP March 27, 1997 Playing PLAY Playing PAUSE Ready TEARDOWN Ready RECORD Recording SETUP Playing Recording RECORD Recording PAUSE Ready TEARDOWN Ready PLAY Playing SETUP Recording B Open Issues o Define text/rtsp-parameter MIME type. o HS believes that RTSP should only control individual media objects rather than aggregates. This avoids disconnects between presentation descriptions and streams and avoids having to deal separately with single-host and multi-host case. Cost: several PLAY/PAUSE/RECORD in one packet, one for each stream. o Allow changing of transport for a stream that's playing? May not be a great idea since the same can be accomplished by tear down and re-setup. o Allow fragment (#) identifiers for controlling substreams in Quicktime, AVI and ASF files? o How does the server get back to the client unless a persistent connection is used? Probably cannot, in general. o Cache and proxy behavior? o Session: or Set-Cookie: ? o When do relative RTSP URLs make sense? o Nack-require, etc. are dubious. This is getting awfully close to the HTTP extension mechanisms [19] in complexity, but is different. o Use HTTP absolute path + Host field or do the right thing and carry full URL, including host in request? C Changes Since the February 1997 version, the following changes were made: H. Schulzrinne, A. Rao, R. Lanphier [Page 56] Internet Draft RTSP March 27, 1997 o Various editorial changes and clarifications. o Removed references to SDF and replaced by RTSL. o Added Scale general header. o Clarify behavior of PLAY. o Rename GET to DESCRIBE. o Removed SESSION since it is just DESCRIBE in the other direction. o Rename CLOSE to TEARDOWN, in symmetry with SETUP. o Terminology adjusted to "presentation" and "stream". o Redundant syntax BNF in appendix removed since it just duplicates HTTP spec. o Beginnings of cache control. Changes are marked by changebars in the margins of the PostScript version. D Author Addresses Henning Schulzrinne Dept. of Computer Science Columbia University 1214 Amsterdam Avenue New York, NY 10027 USA electronic mail: schulzrinne@cs.columbia.edu Anup Rao Netscape Communications Corp. USA electronic mail: anup@netscape.com Robert Lanphier Progressive Networks 1111 Third Avenue Suite 2900 Seattle, WA 98101 USA electronic mail: robla@prognet.com E Acknowledgements H. Schulzrinne, A. Rao, R. Lanphier [Page 57] Internet Draft RTSP March 27, 1997 This draft is based on the functionality of the RTSP draft. It also borrows format and descriptions from HTTP/1.1. This document has benefited greatly from the comments of all those participating in the MMUSIC-WG. In addition to those already mentioned, the following individuals have contributed to this specification: Rahul Agarwal Eduardo F. Llach Bruce Butterfield Rob McCool Martin Dunsmuir Sujal Patel Eric Fleischman Mark Handley Igor Plotnikov Peter Haight Pinaki Shah Brad Hefta-Gaub Jeff Smith John K. Ho Alexander Sokolsky Ruth Lang Dale Stammen Stephanie Leif John Francis Stracke F Bibliography [1] H. Schulzrinne, "RTP profile for audio and video conferences with minimal control," RFC 1890, Internet Engineering Task Force, Jan. 1996. [2] D. Kristol and L. Montulli, "HTTP state management mechanism," RFC 2109, Internet Engineering Task Force, Feb. 1997. [3] F. Yergeau, G. Nicol, G. Adams, and M. Duerst, "Internationalization of the hypertext markup language," RFC 2070, Internet Engineering Task Force, Jan. 1997. [4] S. Bradner, "Key words for use in RFCs to indicate requirement levels," Internet Draft, Internet Engineering Task Force, Jan. 1997. Work in progress. [5] R. Fielding, J. Gettys, J. Mogul, H. Frystyk, and T. Berners-Lee, "Hypertext transfer protocol -- HTTP/1.1," RFC 2068, Internet Engineering Task Force, Jan. 1997. [6] M. Handley, "SDP: Session description protocol," Internet Draft, Internet Engineering Task Force, Nov. 1996. Work in progress. [7] A. Freier, P. Karlton, and P. Kocher, "The TLS protocol," Internet Draft, Internet Engineering Task Force, Dec. 1996. Work in progress. H. Schulzrinne, A. Rao, R. Lanphier [Page 58] Internet Draft RTSP March 27, 1997 [8] J. Franks, P. Hallam-Baker, J. Hostetler, P. A. Luotonen, and E. L. Stewart, "An extension to HTTP: digest access authentication," RFC 2069, Internet Engineering Task Force, Jan. 1997. [9] J. Postel, "User datagram protocol," STD 6, RFC 768, Internet Engineering Task Force, Aug. 1980. [10] R. Hinden and C. Partridge, "Version 2 of the reliable data protocol (RDP)," RFC 1151, Internet Engineering Task Force, Apr. 1990. [11] J. Postel, "Transmission control protocol," STD 7, RFC 793, Internet Engineering Task Force, Sept. 1981. [12] M. Handley, H. Schulzrinne, and E. Schooler, "SIP: Session initiation protocol," Internet Draft, Internet Engineering Task Force, Dec. 1996. Work in progress. [13] P. McMahon, "GSS-API authentication method for SOCKS version 5," RFC 1961, Internet Engineering Task Force, June 1996. [14] D. Crocker, "Augmented BNF for syntax specifications: ABNF," Internet Draft, Internet Engineering Task Force, Oct. 1996. Work in progress. [15] R. Elz, "A compact representation of IPv6 addresses," RFC 1924, Internet Engineering Task Force, Apr. 1996. [16] T. Berners-Lee, L. Masinter, and M. McCahill, "Uniform resource locators (URL)," RFC 1738, Internet Engineering Task Force, Dec. 1994. [17] International Telecommunication Union, "Visual telephone systems and equipment for local area networks which provide a non-guaranteed quality of service," Recommendation H.323, Telecommunication Standardization Sector of ITU, Geneva, Switzerland, May 1996. [18] ISO/IEC, "Information technology -- generic coding of moving pictures and associated audio informaiton -- part 6: extension for digital storage media and control," Draft International Standard ISO 13818-6, International Organization for Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland, Nov. 1995. [19] D. Connolly, "PEP: an extension mechanism for http," Internet Draft, Internet Engineering Task Force, Jan. 1997. Work in progress. H. Schulzrinne, A. Rao, R. Lanphier [Page 59] Internet Draft RTSP March 27, 1997 Table of Contents 1 Introduction ........................................ 1 1.1 Purpose ............................................. 1 1.2 Requirements ........................................ 3 1.3 Terminology ......................................... 3 1.4 Protocol Properties ................................. 5 1.5 Extending RTSP ...................................... 6 1.6 Overall Operation ................................... 7 1.7 RTSP States ......................................... 8 1.8 Relationship with Other Protocols ................... 9 2 Notational Conventions .............................. 10 3 Protocol Parameters ................................. 10 3.1 H3.1 ................................................ 10 3.2 RTSP URL ............................................ 10 3.3 Conference Identifiers .............................. 11 3.4 SMPTE Relative Timestamps ........................... 12 3.5 Normal Play Time .................................... 13 3.6 Absolute Time ....................................... 13 4 RTSP Message ........................................ 13 4.1 Message Types ....................................... 14 4.2 Message Headers ..................................... 14 4.3 Message Body ........................................ 14 4.4 Message Length ...................................... 14 5 Request ............................................. 15 6 Response ............................................ 16 6.1 Status-Line ......................................... 17 6.1.1 Status Code and Reason Phrase ....................... 17 6.1.2 Response Header Fields .............................. 19 7 Entity .............................................. 19 7.1 Entity Header Fields ................................ 21 7.2 Entity Body ......................................... 21 8 Connections ......................................... 21 8.1 Pipelining .......................................... 22 8.2 Reliability and Acknowledgements .................... 22 9 Method Definitions .................................. 23 9.1 OPTIONS ............................................. 24 9.2 DESCRIBE ........................................... 25 9.3 SETUP .............................................. 26 9.4 PLAY ............................................... 27 9.5 PAUSE .............................................. 28 9.6 TEARDOWN ........................................... 30 9.7 GET_PARAMETER ...................................... 30 9.8 SET_PARAMETER ...................................... 31 9.9 REDIRECT ........................................... 31 9.10 RECORD ............................................. 32 H. Schulzrinne, A. Rao, R. Lanphier [Page 60] Internet Draft RTSP March 27, 1997 9.11 Embedded Binary Data ................................ 32 10 Status Code Definitions ............................. 33 10.1 Redirection 3xx ..................................... 33 10.2 Client Error 4xx .................................... 33 10.2.1 451 Parameter Not Understood ........................ 33 10.2.2 452 Conference Not Found ............................ 33 10.2.3 453 Not Enough Bandwidth ............................ 33 10.2.4 45x Session Not Found ............................... 33 10.2.5 45x Method Not Valid in This State .................. 33 10.2.6 45x Header Field Not Valid for Resource ............. 33 10.2.7 45x Invalid Range ................................... 33 10.2.8 45x Parameter Is Read-Only .......................... 34 11 Header Field Definitions ............................ 34 11.1 Accept .............................................. 34 11.2 Accept-Encoding ..................................... 35 11.3 Accept-Language ..................................... 35 11.4 Allow ............................................... 36 11.5 Authorization ....................................... 36 11.6 Bandwidth ........................................... 36 11.7 Blocksize ........................................... 36 11.8 Cache-Control ....................................... 37 11.9 Conference .......................................... 39 11.10 Connection .......................................... 39 11.11 Content-Encoding .................................... 39 11.12 Content-Length ...................................... 39 11.13 Content-Type ........................................ 39 11.14 Date ................................................ 40 11.15 Expires ............................................. 40 11.16 If-Modified-Since ................................... 41 11.17 Last-modified ....................................... 41 11.18 Location ............................................ 41 11.19 Nack-Transport-Require .............................. 41 11.20 Range ............................................... 41 11.21 Require ............................................. 42 11.22 Retry-After ......................................... 43 11.23 Scale ............................................... 43 11.24 Speed ............................................... 44 11.25 Server .............................................. 44 11.26 Session ............................................. 44 11.27 Transport ........................................... 45 11.28 Transport-Require ................................... 46 11.29 Unsupported ......................................... 46 11.30 User-Agent .......................................... 47 11.31 Via ................................................. 47 11.32 WWW-Authenticate .................................... 47 12 Caching ............................................. 47 13 Examples ............................................ 48 13.1 Media on Demand (Unicast) ........................... 48 H. Schulzrinne, A. Rao, R. Lanphier [Page 61] Internet Draft RTSP March 27, 1997 13.2 Live Media Presentation Using Multicast ............. 49 13.3 Playing media into an existing session .............. 50 13.4 Recording ........................................... 51 14 Syntax .............................................. 52 14.1 Base Syntax ......................................... 52 15 Security Considerations ............................. 52 A RTSP Protocol State Machines ........................ 53 A.1 Client State Machine ................................ 54 A.2 Server State Machine ................................ 55 B Open Issues ......................................... 56 C Changes ............................................. 56 D Author Addresses .................................... 57 E Acknowledgements .................................... 57 F Bibliography ........................................ 58 H. Schulzrinne, A. Rao, R. Lanphier [Page 62]