Internet Engineering Task Force MMUSIC WG Internet Draft H. Schulzrinne Columbia U. A. Rao Cisco R. Lanphier RealNetworks M. Westerlund Ericsson draft-ietf-mmusic-rfc2326bis-02.txt November 01, 2002 Expires: April, 2003 Real Time Streaming Protocol (RTSP) STATUS OF THIS MEMO This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet- Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference mate- rial or to cite them other than as "work in progress". The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt To view the list Internet-Draft Shadow Directories, see http://www.ietf.org/shadow.html. Abstract This memorandum is a revision of RFC 2326, which is currently a Pro- posed Standard. The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the delivery of data with real-time proper- ties. RTSP provides an extensible framework to enable controlled, on- demand delivery of real-time data, such as audio and video. Sources H. Schulzrinne et. al. [Page 1] Internet Draft RTSP November 01, 2002 of data can include both live data feeds and stored clips. This pro- tocol is intended to control multiple data delivery sessions, provide a means for choosing delivery channels such as UDP, multicast UDP and TCP, and provide a means for choosing delivery mechanisms based upon RTP (RFC 1889). H. Schulzrinne et. al. [Page 2] Internet Draft RTSP November 01, 2002 1 Introduction 1.1 The Update of the Specification | This is the draft to an update of the RTSP which currently is a pro- | posed standard defined in [21]. During the years since RTSP was pub- | lished many flaws has been found. This draft tries to address these. | The work is not yet completed to get all known issues resolved. | The goal is to progress RTSP to draft standard. If that is possible | without first publishing it as a proposed standard is not yet deter- | mined, as it depends on the changes necessary to make the protocol | work. | See the list of changes in chapter F to see what has been addressed. | The currently open issues are listed in chapter E | There is currently a list of reported bugs available at "http://rtsp- | spec.sourceforge.net". This list should be taken into account when | reading this specification. A lot of these bugs are addressed but not | yet all. Please comment on unresolved ones to give your view. | Another way of giving input on this work is to send e-mail to the | MMUSIC WG's mailing list mmusic@ietf.org and the authors. | Take special notice of the following: | + The example section 14 has not yet been revised as the changes | to protocol has not been completed. | + The BNF chapter 16 has neither been compiled completely. | 1.2 Purpose The Real-Time Streaming Protocol (RTSP) establishes and controls either a single or several time-synchronized streams of continuous media such as audio and video. It does not typically deliver the con- tinuous streams itself, although interleaving of the continuous media stream with the control stream is possible (see Section 10.13). 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 presenta- tion description. H. Schulzrinne et. al. [Page 3] Internet Draft RTSP November 01, 2002 There is no necessity for a notion of an RTSP connection; instead, a server maintains a session labeled by an identifier. An RTSP session is in normally not 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 proto- col 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 contin- uous media. The protocol is intentionally similar in syntax and operation to HTTP/1.1 [26] so that extension mechanisms to HTTP can in most cases also be added to RTSP. However, RTSP differs in a number of important aspects from HTTP: + RTSP introduces a number of new methods and has a different pro- tocol identifier. + An RTSP server needs to maintain state by default in almost all cases, as opposed to the stateless nature of HTTP. + Both an RTSP server and client can issue requests. + Data is usually carried out-of-band by a different protocol. Session descriptions is one possible exception. + RTSP is defined to use ISO 10646 (UTF-8) rather than ISO 8859-1, consistent with current HTML internationalization efforts [3]. + The Request-URI always contains the absolute URI. Because of backward compatibility with a historical blunder, HTTP/1.1 [26] carries only the absolute path in the request and puts the host name in a separate header field. This makes "virtual hosting" easier, where a single host with one IP address hosts several document trees. The protocol supports the following operations: Retrieval of media from media server: The client can request a pre- sentation description via HTTP or some other method. If the presentation is being multicast, the presentation description contains the multicast addresses and ports to be used for the continuous media. If the presentation is to be sent only to the client via unicast, the client provides the destination H. Schulzrinne et. al. [Page 4] Internet Draft RTSP November 01, 2002 for security reasons. Invitation of a media server to a conference: A media server can be "invited" to join an existing conference, either to play back media into the presentation or to record all or a subset of the media in a presentation. This mode is useful for dis- tributed teaching applications. Several parties in the confer- ence may take turns "pushing the remote control buttons". Addition of media to an existing presentation: Particularly for live presentations, it is useful if the server can tell the client about additional media becoming available. RTSP requests may be handled by proxies, tunnels and caches as in HTTP/1.1 [26]. 1.3 Requirements The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this document are to be interpreted as described in RFC 2119 [4]. 1.4 Terminology Some of the terminology has been adopted from HTTP/1.1 [26]. Terms not listed here are defined as in HTTP/1.1. Aggregate control: The control of the multiple streams using a sin- gle timeline by the server. For audio/video feeds, this means that the client may issue a single play or pause message to control both the audio and video feeds. | Aggregate control URI: The URI that represents the whole aggregate. | Normally specified in the session description. Conference: a multiparty, multimedia presentation, where "multi" implies greater than or equal to one. Client: The client requests media service from the media server. | Connection: A transport layer virtual circuit established between two programs for the purpose of communication. Container file: A file which may contain multiple media streams which often comprise a presentation when played together. RTSP servers may offer aggregate control on these files, though the concept of a container file is not embedded in the protocol. H. Schulzrinne et. al. [Page 5] Internet Draft RTSP November 01, 2002 Continuous media: Data where there is a timing relationship between source and sink; that is, the sink must reproduce the timing relationship that existed at the source. The most common exam- ples of continuous media are audio and motion video. Continu- ous media can be real-time (interactive), where there is a "tight" timing relationship between source and sink, or streaming (playback), where the relationship is less strict. Entity: The information transferred as the payload of a request or response. An entity consists of metainformation in the form of entity-header fields and content in the form of an entity- body, as described in Section 8. Feature tag: A tag representing a certain set of functionality, i.e. a feature. Media initialization: Datatype/codec specific initialization. This includes such things as clockrates, color tables, etc. Any transport-independent information which is required by a client for playback of a media stream occurs in the media ini- tialization phase of stream setup. Media parameter: Parameter specific to a media type that may be changed before or during stream playback. Media server: The server providing playback or recording services for one or more media streams. Different media streams within a presentation may originate from different media servers. A media server may reside on the same or a different host as the web server the presentation is invoked from. Media server indirection: Redirection of a media client to a dif- ferent media server. (Media) stream: A single media instance, e.g., an audio stream or a video stream as well as a single whiteboard or shared applica- tion group. When using RTP, a stream consists of all RTP and RTCP packets created by a source within an RTP session. This is equivalent to the definition of a DSM-CC stream([5]). Message: The basic unit of RTSP communication, consisting of a structured sequence of octets matching the syntax defined in Section 16 and transmitted via a connection or a connection- less protocol. Non-Aggregated Control: Control of a single media stream. Only possible in RTSP sessions with a single media. H. Schulzrinne et. al. [Page 6] Internet Draft RTSP November 01, 2002 Participant: Member of a conference. A participant may be a machine, e.g., a media record or playback server. Presentation: A set of one or more streams presented to the client as a complete media feed, using a presentation description as defined below. In most cases in the RTSP context, this implies aggregate control of those streams, but does not have to. Presentation description: A presentation description contains information about one or more media streams within a presenta- tion, such as the set of encodings, network addresses and information about the content. Other IETF protocols such as SDP (RFC 2327 [24]) use the term "session" for a live presen- tation. The presentation description may take several differ- ent formats, including but not limited to the session descrip- tion format SDP. Response: An RTSP response. If an HTTP response is meant, that is indicated explicitly. Request: An RTSP request. If an HTTP request is meant, that is indicated explicitly. RTSP session: A state established on a RTSP server by a client with | an SETUP request. The RTSP session exist until it either time- | outs or is explicitly removed by a TEARDOWN request. The ses- | sion contains state about which media resources that can be | played or recorded, and their transport. Transport initialization: The negotiation of transport information (e.g., port numbers, transport protocols) between the client and the server. 1.5 Protocol Properties RTSP has the following properties: Extendable: New methods and parameters can be easily added to RTSP. Easy to parse: RTSP can be parsed by standard HTTP or MIME parsers. Secure: RTSP re-uses web security mechanisms, either at the trans- port level (TLS, RFC 2246 [27]) or within the protocol itself. All HTTP authentication mechanisms such as basic (RFC 2616 [26]) and digest authentication (RFC 2069 [6]) are directly applicable. H. Schulzrinne et. al. [Page 7] Internet Draft RTSP November 01, 2002 Transport-independent: RTSP may use either an unreliable datagram protocol (UDP) (RFC 768 [7]), a reliable datagram protocol (RDP, RFC 1151, not widely used [8]) or a reliable stream pro- tocol such as TCP (RFC 793 [9]) as it implements application- level reliability. Multi-server capable: Each media stream within a presentation can reside on a different server. The client automatically estab- lishes several concurrent control sessions with the different media servers. Media synchronization is performed at the transport level. Control of recording devices: The protocol can control both record- ing and playback devices, as well as devices that can alter- nate between the two modes ("VCR"). Separation of stream control and conference initiation: Stream con- trol is divorced from inviting a media server to a conference. In particular, SIP [10] or H.323 [28] may be used to invite a server to a conference. Suitable for professional applications: RTSP supports frame-level accuracy through SMPTE time stamps to allow remote digital editing. Presentation description neutral: The protocol does not impose a particular presentation description or metafile format and can convey the type of format to be used. However, the presenta- tion description must contain at least one RTSP URI. Proxy and firewall friendly: The protocol should be readily handled by both application and transport-layer (SOCKS [11]) fire- walls. A firewall may need to understand the SETUP method to open a "hole" for the UDP media stream. HTTP-friendly: Where sensible, RTSP reuses HTTP concepts, so that the existing infrastructure can be reused. This infrastructure includes PICS (Platform for Internet Content Selection [12,13]) for associating labels with content. However, RTSP does not just add methods to HTTP since the controlling con- tinuous media requires server state in most cases. Appropriate server control: If a client can start a stream, it must be able to stop a stream. Servers should not start streaming to clients in such a way that clients cannot stop the stream. Transport negotiation: The client can negotiate the transport method prior to actually needing to process a continuous media H. Schulzrinne et. al. [Page 8] Internet Draft RTSP November 01, 2002 stream. Capability negotiation: If basic features are disabled, there must be some clean mechanism for the client to determine which methods are not going to be implemented. This allows clients to present the appropriate user interface. For example, if seeking is not allowed, the user interface must be able to disallow moving a sliding position indicator. An earlier requirement in RTSP was multi-client capability. However, it was determined that a better approach was to make sure that the protocol is easily extensible to the multi- client scenario. Stream identifiers can be used by several control streams, so that "passing the remote" would be possi- ble. The protocol would not address how several clients nego- tiate access; this is left to either a "social protocol" or some other floor control mechanism. 1.6 Extending RTSP Since not all media servers have the same functionality, media servers by necessity will support different sets of requests. For example: + A server may only be capable of playback thus has no need to sup- port the RECORD request. + A server may not be capable of seeking (absolute positioning) if it is to support live events only. + Some servers may not support setting stream parameters and thus not support GET_PARAMETER and SET_PARAMETER. A server SHOULD implement all header fields described in Section 12. It is up to the creators of presentation descriptions not to ask the impossible of a server. This situation is similar in HTTP/1.1 [26], where the methods described in [H19.5] are not likely to be supported across all servers. RTSP can be extended in three ways, listed here in order of the mag- nitude of changes supported: + Existing methods can be extended with new parameters, as long as these parameters can be safely ignored by the recipient. (This is equivalent to adding new parameters to an HTML tag.) If the client needs negative acknowledgement when a method extension is H. Schulzrinne et. al. [Page 9] Internet Draft RTSP November 01, 2002 not supported, a tag corresponding to the extension may be added in the Require: field (see Section 12.32). + New methods can be added. If the recipient of the message does not understand the request, it responds with error code 501 (Not Implemented) and the sender should not attempt to use this method again. A client may also use the OPTIONS method to inquire about methods supported by the server. The server SHOULD list the meth- ods it supports using the Public response header. + A new version of the protocol can be defined, allowing almost all aspects (except the position of the protocol version number) to change. 1.7 Overall Operation Each presentation and media stream may be identified by an RTSP URL. The overall presentation and the properties of the media the presen- tation 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 main- tains a common time axis. For simplicity of exposition and 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, lan- guage, 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 iden- tified 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 exam- ple, 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 distin- guished: H. Schulzrinne et. al. [Page 10] Internet Draft RTSP November 01, 2002 Unicast: The media is transmitted to the source of the RTSP request, with the port number chosen by the client. Alterna- tively, the media is transmitted on the same reliable stream as RTSP. Multicast, server chooses address: The media server picks the mul- ticast address and port. This is the typical case for a live or near-media-on-demand transmission. Multicast, client chooses address: If the server is to participate in an existing multicast conference, the multicast address, port and encryption key are given by the conference descrip- tion, established by means outside the scope of this specifi- cation. 1.8 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 con- trolled by RTSP requests issued sequentially on different TCP connec- tions. Therefore, the server needs to maintain "session state" to be able to correlate RTSP requests with a stream. The state transitions are described in Appendix A. Many methods in RTSP do not contribute to state. However, the follow- ing play a central role in defining the allocation and usage of stream resources on the server: SETUP, PLAY, RECORD, PAUSE, and TEAR- DOWN. SETUP: Causes the server to allocate resources for a stream and create an RTSP session. PLAY and RECORD: Starts data transmission on a stream allocated via SETUP. PAUSE: Temporarily halts a stream without freeing server resources. TEARDOWN: Frees resources associated with the stream. The RTSP session ceases to exist on the server. RTSP methods that contribute to state use the Session header field (Section 12.37) to identify the RTSP session whose state is being manipulated. The server generates session identifiers in response to SETUP requests (Section 10.4). H. Schulzrinne et. al. [Page 11] Internet Draft RTSP November 01, 2002 1.9 Relationship with Other Protocols RTSP has some overlap in functionality with HTTP. It also may inter- act with HTTP in that the initial contact with streaming content is often to be made through a web page. The current protocol specifica- tion aims to allow different hand-off points between a web server and the media server implementing RTSP. For example, the presentation description can be retrieved using HTTP or RTSP, which reduces roundtrips in web-browser-based scenarios, yet also allows for stan- dalone RTSP servers and clients which do not rely on HTTP at all. However, RTSP differs fundamentally from HTTP in that most data delivery takes place out-of-band in a different protocol. HTTP is an asymmetric protocol where the client issues requests and the server responds. In RTSP, both the media client and media server can issue requests. RTSP requests are also not stateless; they may set parame- ters and continue to control a media stream long after the request has been acknowledged. Re-using HTTP functionality has advantages in at least two areas, namely security and proxies. The requirements are very similar, so having the ability to adopt HTTP work on caches, proxies and authentication is valuable. While most real-time media will use RTP as a transport protocol, RTSP is not tied to RTP. RTSP assumes the existence of a presentation description format that can express both static and temporal properties of a presentation containing several media streams. 2 Notational Conventions Since many of the definitions and syntax are identical to HTTP/1.1, this specification only points to the section where they are defined rather than copying it. For brevity, [HX.Y] is to be taken to refer to Section X.Y of the current HTTP/1.1 specification (RFC 2616 [26]). All the mechanisms specified in this document are described in both prose and an augmented Backus-Naur form (BNF) similar to that used in [H2.1]. It is described in detail in RFC 2234 [14], with the differ- ence that this RTSP specification maintains the "#" notation for comma-separated lists from [H2.1]. In this draft, we use indented and smaller-type paragraphs to provide background and motivation. This is intended to give readers who were not involved with the formulation of the specification an H. Schulzrinne et. al. [Page 12] Internet Draft RTSP November 01, 2002 understanding of why things are the way that they are in RTSP. b 3 Protocol Parameters 3.1 RTSP Version HTTP Specification Section [H3.1] applies, with HTTP replaced by RTSP. This specification defines version 1.0 of 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 syn- tax and semantics for RTSP URLs. | rtsp_URL = ( "rtsp:" / "rtspu:" / "rtsps" ) || "//" host [ ":" port ] [ abs_path ] || host = As defined by RFC 2732 [30] || abs_path = As defined by RFC 2396 [22] || port = *DIGIT || 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 pro- tocol (within the Internet, TCP), while the scheme rtspu identifies an unreliable protocol (within the Internet, UDP). The scheme rtsps identifies a reliable transport using TLS [27]. If the port is empty or not given, port 554 is assumed. The seman- tics are that the identified resource can be controlled by RTSP at the server listening for TCP (scheme "rtsp") connections or UDP (scheme "rtspu") packets on that port of host, and the Request-URI for the resource is rtsp_URL. The use of IP addresses in URLs SHOULD be avoided whenever possible (see RFC 1924 [16]). Note: Using qualified domain names in any URL is one requirement for making it possible for RFC 2326 implementations of RTSP to use IPv6. This specification is updated to allow for lit- eral IPv6 addresses in RTSP URLs using the host specification in RFC 2732 [30]. H. Schulzrinne et. al. [Page 13] Internet Draft RTSP November 01, 2002 A presentation or a stream is identified by a textual media identi- fier, using the character set and escape conventions [H3.2] of URLs (RFC 2396 [22]). URLs may refer to a stream or an aggregate of streams, i.e., a presentation. Accordingly, requests described in Section 10 can apply to either the whole presentation or an individ- ual stream within the presentation. Note that some request methods can only be applied to streams, not presentations and vice versa. For example, the RTSP URL: rtsp://media.example.com:554/twister/audiotrack identifies the audio stream within the presentation "twister", which can be controlled via RTSP requests issued over a TCP connection to port 554 of host media.example.com Also, the RTSP URL: rtsp://media.example.com:554/twister identifies the presentation "twister", which may be composed of audio and video streams. This does not imply a standard way to reference streams in URLs. The presentation description defines the hierarchical relationships in the presentation and the URLs for the indi- vidual streams. A presentation description may name a stream "a.mov" and the whole presentation "b.mov". The path components of the RTSP URL are opaque to the client and do not imply any particular file system structure for the server. This decoupling also allows presentation descriptions to be used with non-RTSP media control protocols simply by replacing the scheme in the URL. 3.3 Session Identifiers Session identifiers are opaque strings of arbitrary length. Linear white space must be URL-escaped. A session identifier MUST be chosen randomly and MUST be at least eight octets long to make guessing it more difficult. (See Section 17.) H. Schulzrinne et. al. [Page 14] Internet Draft RTSP November 01, 2002 session-id = 8*( ALPHA / DIGIT / safe ) 3.4 SMPTE Relative Timestamps A SMPTE relative timestamp expresses time relative to the start of the clip. Relative timestamps are expressed as SMPTE time codes for frame-level access accuracy. The time code has the format hours:minutes:seconds:frames.subframes, with the origin at the start of the clip. The default smpte format is"SMPTE 30 drop" format, with frame rate is 29.97 frames per second. Other SMPTE codes MAY be supported (such as "SMPTE 25") through the use of alternative use of "smpte time". For the "frames" field in the time value can assume the values 0 through 29. The difference between 30 and 29.97 frames per second is handled by dropping the first two frame indices (values 00 and 01) of every minute, except every tenth minute. If the frame value is zero, it may be omitted. Subframes are measured in one-hundredth of a frame. smpte-range = smpte-type "=" smpte-range-spec || smpte-range-spec = ( smpte-time "-" [ smpte-time ] ) || / ( "-" smpte-time ) || smpte-type = "smpte" / "smpte-30-drop" / "smpte-25" || ; other timecodes may be added || smpte-time = 1*2DIGIT ":" 1*2DIGIT ":" 1*2DIGIT || [ ":" 1*2DIGIT [ "." 1*2DIGIT ] ] || Examples: smpte=10:12:33:20- smpte=10:07:33- smpte=10:07:00-10:07:33:05.01 smpte-25=10:07:00-10:07:33:05.01 3.5 Normal Play Time Normal play time (NPT) indicates the stream absolute position rela- tive to the beginning of the presentation, not to be confused with the Network Time Protocol (NTP). The timestamp consists of a decimal fraction. The part left of the decimal may be expressed in either seconds or hours, minutes, and seconds. The part right of the decimal point measures fractions of a second. H. Schulzrinne et. al. [Page 15] Internet Draft RTSP November 01, 2002 The beginning of a presentation corresponds to 0.0 seconds. Negative values are not defined. The special constant now is defined as the current instant of a live event. It MAY only be used for live events, and SHALL NOT be used for on-demand content. NPT is defined as in DSM-CC: "Intuitively, NPT is the clock the viewer associates with a program. It is often digitally displayed on a VCR. NPT advances normally when in normal play mode (scale = 1), advances at a faster rate when in fast scan forward (high positive scale ratio), decrements when in scan reverse (high negative scale ratio) and is fixed in pause mode. NPT is (logically) equivalent to SMPTE time codes." [5] npt-range = ["npt" "="] npt-range-spec ; implementations SHOULD use npt= prefix, but SHOULD ; be prepared to interoperate with RFC 2326 ; implementations which don't use it npt-range-spec = ( npt-time "-" [ npt-time ] ) / ( "-" npt-time ) npt-time = "now" / npt-sec / npt-hhmmss npt-sec = 1*DIGIT [ "." *DIGIT ] npt-hhmmss = npt-hh ":" npt-mm ":" npt-ss [ "." *DIGIT ] npt-hh = 1*DIGIT ; any positive number npt-mm = 1*2DIGIT ; 0-59 npt-ss = 1*2DIGIT ; 0-59 Examples: npt=123.45-125 npt=12:05:35.3- npt=now- The syntax conforms to ISO 8601. The npt-sec notation is opti- mized for automatic generation, the ntp-hhmmss notation for consumption by human readers. The "now" constant allows clients to request to receive the live feed rather than the stored or time-delayed version. This is needed since neither absolute time nor zero time are appropriate for this case. 3.6 Absolute Time Absolute time is expressed as ISO 8601 timestamps, using UTC (GMT). Fractions of a second may be indicated. | H. Schulzrinne et. al. [Page 16] Internet Draft RTSP November 01, 2002 utc-range = "clock" "=" utc-range-spec || utc-range-spec = ( utc-time "-" [ utc-time ] ) / ( "-" utc-time ) || utc-time = utc-date "T" utc-time "Z" || utc-date = 8DIGIT ; < YYYYMMDD > || utc-time = 6DIGIT [ "." fraction ] ; < HHMMSS.fraction > || fraction = 1*DIGIT || Example for November 8, 1996 at 14h37 and 20 and a quarter seconds UTC: 19961108T143720.25Z 3.7 Option Tags Option tags are unique identifiers used to designate new options in RTSP. These tags are used in in Require (Section 12.32), Proxy- Require (Section 12.27), and Supported (Section 12.38) header fields. Syntax: option-tag = token The creator of a new RTSP option should either prefix the option with a reverse domain name (e.g., "com.foo.mynewfeature" is an apt name for a feature whose inventor can be reached at "foo.com"), or regis- ter the new option with the Internet Assigned Numbers Authority (IANA), see IANA Section 18. 4 RTSP Message RTSP is a text-based protocol and uses the ISO 10646 character set in UTF-8 encoding (RFC 2279 [18]). Lines are terminated by CRLF, but receivers should be prepared to also interpret CR and LF by them- selves 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. H. Schulzrinne et. al. [Page 17] Internet Draft RTSP November 01, 2002 The 10646 character set avoids tricky character set switching, but is invisible to the application as long as US-ASCII is being used. This is also the encoding used for RTCP. ISO 8859-1 translates directly into Unicode with a high-order octet of zero. ISO 8859-1 characters with the most-significant bit set are represented as 1100001x 10xxxxxx. (See RFC 2279 [18]) RTSP messages can be carried over any lower-layer transport protocol | that is 8-bit clean. RTSP messages are vulnerable to bit errors and | SHOULD NOT be subjected to them. 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 terminated by the first empty line after the header fields, regardless of the entity-header fields present in the message. (Note: An empty line consists of only CRLF.) 2. If a Content-Length header field (section 12.14) 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. Note that RTSP does not (at present) support the HTTP/1.1 "chunked" transfer coding(see [H3.6.1]) and requires the presence of the Con- tent-Length header field. H. Schulzrinne et. al. [Page 18] Internet Draft RTSP November 01, 2002 Given the moderate length of presentation descriptions returned, the server should always be able to determine its length, even if it is generated dynamically, making the chun- ked transfer encoding unnecessary. 5 General Header Fields See [H4.5], except that Pragma, Trailer, Transfer-Encoding, Upgrade, and Warning headers are not defined. RTSP further defines the CSeq, and Timestamp: general-header = Cache-Control ; Section 12.9 / Connection ; Section 12.10 / CSeq ; Section 12.17 / Date ; Section 12.18 / Timestamp ; Section 12.39 / Via ; Section 12.44 6 Request A request message from a client to a server or vice versa includes, within the first line of that message, the method to be applied to the resource, the identifier of the resource, and the protocol ver- sion in use. Request = Request-Line ; Section 6.1 *( general-header ; Section 5 / request-header ; Section 6.2 / entity-header ) ; Section 8.1 CRLF [ message-body ] ; Section 4.3 6.1 Request Line Request-Line = Method SP Request-URI SP RTSP-Version CRLF Method = "DESCRIBE" ; Section 10.2 / "ANNOUNCE" ; Section 10.3 / "GET_PARAMETER" ; Section 10.8 / "OPTIONS" ; Section 10.1 H. Schulzrinne et. al. [Page 19] Internet Draft RTSP November 01, 2002 / "PAUSE" ; Section 10.6 / "PLAY" ; Section 10.5 / "PING" ; Section 10.12 / "RECORD" ; Section 10.11 / "REDIRECT" ; Section 10.10 / "SETUP" ; Section 10.4 / "SET_PARAMETER" ; Section 10.9 / "TEARDOWN" ; Section 10.7 / extension-method extension-method = token Request-URI = "*" / absolute_URI RTSP-Version = "RTSP" "/" 1*DIGIT "." 1*DIGIT 6.2 Request Header Fields request-header = Accept ; Section 12.1 / Accept-Encoding ; Section 12.2 / Accept-Language ; Section 12.3 / Authorization ; Section 12.6 / Bandwidth ; Section 12.7 / Blocksize ; Section 12.8 / From ; Section 12.20 / If-Modified-Since ; Section 12.23 / Proxy-Require ; Section 12.27 / Range ; Section 12.29 / Referer ; Section 12.30 / Require ; Section 12.32 / Scale ; Section 12.34 / Session ; Section 12.37 / Speed ; Section 12.35 / Supported ; Section 12.38 / Transport ; Section 12.40 / User-Agent ; Section 12.42 Note that in contrast to HTTP/1.1 [26], RTSP requests always contain the absolute URL (that is, including the scheme, host and port) rather than just the absolute path. HTTP/1.1 requires servers to understand the absolute URL, but clients are supposed to use the Host request header. This is purely needed for backward-compatibility with HTTP/1.0 H. Schulzrinne et. al. [Page 20] Internet Draft RTSP November 01, 2002 servers, a consideration that does not apply to RTSP. The asterisk "*" in the Request-URI means that the request does not apply to a particular resource, but to the server or proxy itself, and is only allowed when the method used does not necessarily apply to a resource. One example would be: OPTIONS * RTSP/1.0 Which will determine the capabilities of the server or the proxy that first receives the request. If one needs to address the server explic- itly one needs to put in a absolute URL with the servers address. OPTIONS rtsp://example.com RTSP/1.0 7 Response [H6] applies except that HTTP-Version is replaced by RTSP-Version. Also, RTSP defines additional status codes and does not define some HTTP codes. The valid response codes and the methods they can be used with are defined in Table 1. After receiving and interpreting a request message, the recipient responds with an RTSP response message. Response = Status-Line ; Section 7.1 *( general-header ; Section 5 / response-header ; Section 7.1.2 / entity-header ) ; Section 8.1 CRLF [ message-body ] ; Section 4.3 7.1 Status-Line The first line of a Response message is the Status-Line, consisting of the protocol version followed by a numeric status code, and the textual phrase associated with the status code, with each element separated by SP characters. No CR or LF is allowed except in the final CRLF sequence. H. Schulzrinne et. al. [Page 21] Internet Draft RTSP November 01, 2002 Status-Line = RTSP-Version SP Status-Code SP Reason-Phrase CRLF 7.1.1 Status Code and Reason Phrase The Status-Code element is a 3-digit integer result code of the attempt to understand and satisfy the request. These codes are fully defined in Section 11. The Reason-Phrase is intended to give a short textual description of the Status-Code. The Status-Code is intended for use by automata and the Reason-Phrase is intended for the human user. The client is not required to examine or display the Reason- Phrase. The first digit of the Status-Code defines the class of response. The last two digits do not have any categorization role. There are 5 values for the first digit: + 1xx: Informational - Request received, continuing process + 2xx: Success - The action was successfully received, understood, and accepted + 3xx: Redirection - Further action must be taken in order to com- plete the request + 4xx: Client Error - The request contains bad syntax or cannot be fulfilled + 5xx: Server Error - The server failed to fulfill an apparently valid request The individual values of the numeric status codes defined for RTSP/1.0, and an example set of corresponding Reason-Phrase's, are presented below. The reason phrases listed here are only recommended -- they may be replaced by local equivalents without affecting the protocol. Note that RTSP adopts most HTTP/1.1 [26] status codes and adds RTSP-specific status codes starting at x50 to avoid conflicts with newly defined HTTP status codes. Status-Code = "100" ; Continue / "200" ; OK / "201" ; Created / "250" ; Low on Storage Space / "300" ; Multiple Choices / "301" ; Moved Permanently H. Schulzrinne et. al. [Page 22] Internet Draft RTSP November 01, 2002 / "302" ; Moved Temporarily / "303" ; See Other / "304" ; Not Modified / "305" ; Use Proxy / "350" ; Going Away / "351" ; Load Balancing / "400" ; Bad Request / "401" ; Unauthorized / "402" ; Payment Required / "403" ; Forbidden / "404" ; Not Found / "405" ; Method Not Allowed / "406" ; Not Acceptable / "407" ; Proxy Authentication Required / "408" ; Request Time-out / "410" ; Gone / "411" ; Length Required / "412" ; Precondition Failed / "413" ; Request Entity Too Large / "414" ; Request-URI Too Large / "415" ; Unsupported Media Type / "451" ; Parameter Not Understood / "452" ; reserved / "453" ; Not Enough Bandwidth / "454" ; Session Not Found / "455" ; Method Not Valid in This State / "456" ; Header Field Not Valid for Resource / "457" ; Invalid Range / "458" ; Parameter Is Read-Only / "459" ; Aggregate operation not allowed / "460" ; Only aggregate operation allowed / "461" ; Unsupported transport / "462" ; Destination unreachable / "500" ; Internal Server Error / "501" ; Not Implemented / "502" ; Bad Gateway / "503" ; Service Unavailable / "504" ; Gateway Time-out / "505" ; RTSP Version not supported / "551" ; Option not supported / extension-code extension-code = 3DIGIT Reason-Phrase = * H. Schulzrinne et. al. [Page 23] Internet Draft RTSP November 01, 2002 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 unrecog- nized 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. 7.1.2 Response Header Fields The response-header fields allow the request recipient to pass addi- tional information about the response which cannot be placed in the Status-Line. These header fields give information about the server and about further access to the resource identified by the Request- URI. response-header = Accept-Ranges ; Section 12.4 / Location ; Section 12.25 / Proxy-Authenticate ; Section 12.26 / Public ; Section 12.28 / Range ; Section 12.29 / Retry-After ; Section 12.31 / RTP-Info ; Section 12.33 / Scale ; Section 12.34 / Session ; Section 12.37 / Server ; Section 12.36 / Speed ; Section 12.35 / Transport ; Section 12.40 / Unsupported ; Section 12.41 / Vary ; Section 12.43 / WWW-Authenticate ; Section 12.45 Response-header field names can be extended reliably only in combina- tion with a change in the protocol version. However, new or experi- mental 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. H. Schulzrinne et. al. [Page 24] Internet Draft RTSP November 01, 2002 Code reason -------------------------------------------------------- 100 Continue all -------------------------------------------------------- 200 OK all 201 Created RECORD 250 Low on Storage Space RECORD -------------------------------------------------------- 300 Multiple Choices all 301 Moved Permanently all 302 Found all 303 See Other all 305 Use Proxy all 350 Going Away all 351 Load Balancing all -------------------------------------------------------- 400 Bad Request all 401 Unauthorized all 402 Payment Required all 403 Forbidden all 404 Not Found all 405 Method Not Allowed all 406 Not Acceptable all 407 Proxy Authentication Required all 408 Request Timeout all 410 Gone all 411 Length Required all 412 Precondition Failed DESCRIBE, SETUP 413 Request Entity Too Large all 414 Request-URI Too Long all 415 Unsupported Media Type all 451 Parameter Not Understood SET_PARAMETER 452 reserved n/a 453 Not Enough Bandwidth SETUP 454 Session Not Found all 455 Method Not Valid In This State all 456 Header Field Not Valid all 457 Invalid Range PLAY, PAUSE 458 Parameter Is Read-Only SET_PARAMETER 459 Aggregate Operation Not Allowed all 460 Only Aggregate Operation Allowed all 461 Unsupported Transport all 462 Destination Unreachable all -------------------------------------------------------- 500 Internal Server Error all 501 Not Implemented all 502 Bad Gateway all H. Schulzrinne et. al. [Page 25] Internet Draft RTSP November 01, 2002 503 Service Unavailable all 504 Gateway Timeout all 505 RTSP Version Not Supported all 551 Option not support all Table 1: Status codes and their usage with RTSP methods 8 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 or the server, depending on who sends and who receives the entity. 8.1 Entity Header Fields Entity-header fields define optional meta-information about the entity-body or, if no body is present, about the resource identified by the request. entity-header = Allow ; Section 12.5 / Content-Base ; Section 12.11 / Content-Encoding ; Section 12.12 / Content-Language ; Section 12.13 / Content-Length ; Section 12.14 / Content-Location ; Section 12.15 / Content-Type ; Section 12.16 / Expires ; Section 12.19 / Last-Modified ; Section 12.24 / 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. 8.2 Entity Body See [H7.2] with the addition that a RTSP message with an entity body | MUST include a Content-Type header. H. Schulzrinne et. al. [Page 26] Internet Draft RTSP November 01, 2002 9 Connections RTSP requests can be transmitted in several different ways: + persistent transport connections used for several request- response transactions; + one connection per request/response transaction; + connectionless mode. The type of transport connection is defined by the RTSP URI (Section 3.2). For the scheme "rtsp", a connection is assumed, while the scheme "rtspu" calls for RTSP requests to be sent without setting up a connection. Unlike HTTP, RTSP allows the media server to send requests to the media client. However, this is only supported for persistent connec- tions, as the media server otherwise has no reliable way of reaching the client. Also, this is the only way that requests from media server to client are likely to traverse firewalls. 9.1 Pipelining A client that supports persistent connections or connectionless mode MAY "pipeline" its requests (i.e., send multiple requests without waiting for each response). A server MUST send its responses to those requests in the same order that the requests were received. 9.2 Reliability and Acknowledgements Requests are acknowledged by the receiver unless they are sent to a | 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 1123) [15], 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, requests MUST | NOT be retransmitted; the RTSP application MUST instead rely on the | underlying transport to provide reliability. | If both the underlying reliable transport such as TCP and the | RTSP application retransmit requests, it is possible that each | packet loss results in two retransmissions. The receiver can- | not typically take advantage of the application-layer retrans- | mission since the transport stack will not deliver the | H. Schulzrinne et. al. [Page 27] Internet Draft RTSP November 01, 2002 application-layer retransmission before the first attempt has | reached the receiver. If the packet loss is caused by conges- | tion, multiple retransmissions at different layers will exac- | erbate the congestion. | If RTSP is used over a small-RTT LAN, standard procedures for opti- | mizing initial TCP round trip estimates, such as those used in T/TCP | (RFC 1644) [19], can be beneficial. | The Timestamp header (Section 12.39) is used to avoid the retransmis- | sion ambiguity problem [20] and obviates the need for Karn's algo- | rithm. | Each request carries a sequence number in the CSeq header (Section | 12.17), which MUST be incremented by one for each distinct request | transmitted. If a request is repeated because of lack of acknowledge- | ment, the request MUST carry the original sequence number (i.e., the | sequence number is not incremented). | Systems implementing RTSP MUST support carrying RTSP over TCP and MAY | support UDP. The default port for the RTSP server is 554 for both UDP | and TCP. | A number of RTSP packets destined for the same control end point may | be packed into a single lower-layer PDU or encapsulated into a TCP | stream. RTSP data MAY be interleaved with RTP and RTCP packets. | Unlike HTTP, an RTSP message MUST contain a Content-Length header | field whenever that message contains a payload. Otherwise, an RTSP | packet is terminated with an empty line immediately following the | last message header. | 9.3 The usage of connections | TCP can be used for both persistent connections and for one message | exchange per connection, as presented above. This section gives fur- | ther rules and recommendations on how to handle these connections so | maximum interoperability and flexibility can be achieved. | A server SHALL handle both persistent connections and one | request/response transaction per connection. A persistent connection | MAY be used for all transactions between the server and client, | including messages to multiple RTSP sessions. However the persistent | connection MAY also be closed after a few message exchanges, e.g. the | initial setup and play command in a session. Later when the client | wishes to send a new request, e.g. pause, to the session a new con- | nection is opened. This connection may either be for a single message | exchange or can be kept open for several messages, i.e. persistent. | H. Schulzrinne et. al. [Page 28] Internet Draft RTSP November 01, 2002 The client MAY close the connection at any time when no outstanding | request/response transactions exist. The server SHOULD NOT close the | connection unless at least one RTSP session timeout period has passed | without data traffic. A server MUST NOT close a connection directly | after responding to a TEARDOWN request for the whole session. | The client SHOULD NOT have more than one connection to the server at | any given point. If a client or proxy handles multiple RTSP sessions | on the same server, it is RECOMMENDED to use only a single connec- | tion. | Older services which was implemented according to RFC 2326 sometimes | requires the client to use persistent connection. The client closing | the connection may result in that the server removes the session. To | achieve interoperability with old servers any client is strongly REC- | OMMENDED to use persistent connections. To make it practically possi- | ble for a client to the rules outlined in this chapter a feature tag | is defined. | con.non-persistent || If a service requires the use of persistent connection a option tag | is specified for usage in Require and Proxy-Require. | con.persistent || A server implemented according to this specification MUST respond | that it supports the feature tag above. A client MAY send a request | including the Supported header in a request to determine support of | non-persistent connections. A server supporting non-persistent con- | nections MUST return the "con.non-persistent" feature tag in its | response. If the client receives the feature tag in the response, it | can be certain that the server handles non-persistent connections. | 9.4 Use of Transport Layer Security | 9.5 Use of IPv6 | This specification has been updated so that it supports IPv6. How- | ever this support was not present in RFC 2326 therefore some interop- | erability issues exist. A RFC 2326 implementation can support IPv6 as | long as no explicit IPv6 addresses are used within RTSP messages. | This require that any RTSP URL pointing at a IPv6 host must use fully | qualified domain name and not a IPv6 address. Further the Transport | H. Schulzrinne et. al. [Page 29] Internet Draft RTSP November 01, 2002 header must not use the parameters source and destination. | Implementations according to this specification MUST understand IPv6 | addresses in URLs, and headers. By this requirement the option-tag | "play.basic" and "record.basic" can be used to determine that a | server or client is capable of handling IPv6 within RTSP. | 10 Method Definitions The method token indicates the method to be performed on the resource | identified by the Request-URI case-sensitive. New methods may be | defined in the future. Method names may not start with a $ character | (decimal 24) and must be a token as defined by the ABNF. Methods are | summarized in Table 2. | method direction object Server req. Client req. ---------------------------------------------------------------- DESCRIBE C->S P,S recommended recommended ANNOUNCE C->S, S->C P,S optional optional GET_PARAMETER C->S, S->C P,S optional optional OPTIONS C->S, S->C P,S R=Req, Sd=Opt Sd=Req, R=Opt PAUSE C->S P,S recommended recommended PING C->S, S->C P,S recommended optional PLAY C->S P,S required required RECORD C->S P,S optional optional REDIRECT S->C P,S optional optional SETUP C->S S required required SET_PARAMETER C->S, S->C P,S optional optional TEARDOWN C->S P,S required required Table 2: Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Legend: R=Responde to, Sd=Send, Opt: Optional, Req: Required, Rec: Recommended Notes on Table 2: PAUSE is recommended, but not required in that a fully functional server can be built that does not support this method, for example, for live feeds. If a server does not support a particular method, it MUST return 501 (Not Implemented) and a client SHOULD not try this method again for this server. 10.1 OPTIONS H. Schulzrinne et. al. [Page 30] Internet Draft RTSP November 01, 2002 The behavior is equivalent to that described in [H9.2]. An OPTIONS | request may be issued at any time, e.g., if the client is about to | try a nonstandard request. It does not influence the session state. | The Public header MUST be included in responses to indicate which | methods that are supported by the server. To specify which methods | that are possible to use for the specified resource, the Allow MAY be | used. By including in the OPTIONS request a Supported header, the | requester can determine which options the other part supports. | The request URI determines which scope the options request has. By | giving the URI of a certain media the capabilities regarding this | media will be responded. By using the "*" URI the request regards the | server without any media relevance. Example: C->S: OPTIONS * RTSP/1.0 CSeq: 1 User-Agent: PhonyClient 1.2 Require: implicit-play Proxy-Require: gzipped-messages Supported: play-basic S->C: RTSP/1.0 200 OK CSeq: 1 Public: DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE Supported: play-basic, gzipped-messages, implicit-play Server: PhonyServer 1.0 Note that the option tags in Require and Proxy-Require are necessar- ily fictional features (one would hope that we would not purposefully overlook a truly useful feature just so that we could have a strong example in this section). 10.2 DESCRIBE The DESCRIBE method retrieves the description of a presentation or media object identified by the request URL from a server. It may use the Accept header to specify the description formats that the client understands. The server responds with a description of the requested resource. The DESCRIBE reply-response pair constitutes the media ini- tialization phase of RTSP. Example: H. Schulzrinne et. al. [Page 31] Internet Draft RTSP November 01, 2002 C->S: DESCRIBE rtsp://server.example.com/fizzle/foo RTSP/1.0 CSeq: 312 User-Agent: PhonyClient 1.2 Accept: application/sdp, application/rtsl, application/mheg S->C: RTSP/1.0 200 OK CSeq: 312 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 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=application 32416 UDP WB a=orient:portrait The DESCRIBE response MUST contain all media initialization informa- tion for the resource(s) that it describes. If a media client obtains a presentation description from a source other than DESCRIBE and that description contains a complete set of media initialization parame- ters, the client SHOULD use those parameters and not then request a description for the same media via RTSP. Additionally, servers SHOULD NOT use the DESCRIBE response as a means of media indirection. By forcing a DESCRIBE response to contain all media initial- ization for the set of streams that it describes, and discour- aging use of DESCRIBE for media indirection, we avoid looping problems that might result from other approaches. Media initialization is a requirement for any RTSP-based system, but the RTSP specification does not dictate that this must be done via the DESCRIBE method. There are three ways that an RTSP client may receive initialization information: H. Schulzrinne et. al. [Page 32] Internet Draft RTSP November 01, 2002 + via RTSP's DESCRIBE method; + via some other protocol (HTTP, email attachment, etc.); + via the command line or standard input (thus working as a browser helper application launched with an SDP file or other media ini- tialization format). It is RECOMMENDED that minimal servers support the DESCRIBE method, and highly recommended that minimal clients support the ability to act as a "helper application" that accepts a media initialization file from standard input, command line, and/or other means that are appropriate to the operating environment of the client. 10.3 ANNOUNCE The ANNOUNCE method serves two purposes: When sent from client to server, ANNOUNCE posts the description of a presentation or media object identified by the request URL to a server. When sent from server to client, ANNOUNCE updates the ses- sion description in real-time. If a new media stream is added to a presentation (e.g., during a live presentation), the whole presentation description should be sent again, rather than just the additional components, so that components can be deleted. Example: C->S: ANNOUNCE rtsp://server.example.com/fizzle/foo RTSP/1.0 CSeq: 312 Date: 23 Jan 1997 15:35:06 GMT Session: 47112344 Content-Type: application/sdp Content-Length: 332 v=0 o=mhandley 2890844526 2890845468 IN IP4 126.16.64.4 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 H. Schulzrinne et. al. [Page 33] Internet Draft RTSP November 01, 2002 m=video 2232 RTP/AVP 31 S->C: RTSP/1.0 200 OK CSeq: 312 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 10.4 SETUP The SETUP request for a URI specifies the transport mechanism to be used for the streamed media. A client can issue a SETUP request for a stream that is already set up or playing in the session to change transport parameters, which a server MAY allow. If it does not allow this, it MUST respond with error 455 (Method Not Valid In This State). A server MAY allow a client to do SETUP while in playing state to add | additional media streams. If not supported the server shall responde | with error 455 (Method Not Allowed In This State). If supported the | added media shall then start to play in sync with the already playing | media. To be able to sync the media with the already playing streams | the SETUP response MUST include a RTP-Info header with the timestamp | value, and a Range header with the corresponding normal play time. To | indicate support for this optional support the options-tag: | "setup.playing" is defined. 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 multi- cast address. Since SETUP includes all transport initialization information, firewalls and other intermediate network devices (which need this information) are spared the more arduous task of parsing the DESCRIBE response, which has been reserved for media ini- tialization. The Transport header specifies the transport parameters acceptable to the client for data transmission; the response will contain the transport parameters selected by the server. C->S: SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0 H. Schulzrinne et. al. [Page 34] Internet Draft RTSP November 01, 2002 CSeq: 302 Transport: RTP/AVP;unicast;client_port=4588-4589 S->C: RTSP/1.0 200 OK CSeq: 302 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Session: 47112344 Transport: RTP/AVP;unicast; client_port=4588-4589;server_port=6256-6257 The server generates session identifiers in response to SETUP requests. If a SETUP request to a server includes a session identi- fier, the server MUST bundle this setup request into the existing session (aggregated session) or return error 459 (Aggregate Operation Not Allowed) (see Section 11.4.11). To control an aggregated session an aggregated control URI MUST be | used. The aggregated control URI MUST be different from any of the | media control URIs included in the aggregate. The aggregated URI | SHOULD be specified by session description, as no general rule exist | to derive it from the included media's. | A session will exist until it is torn down by a TEARDOWN request or | times out. The server MAY remove a session that have had no liveness | signs from the client in the specified timeout time. The default | timeout time is 60 seconds, the server MAY set this to another value, | by in the SETUP response include a timeout value in the session | header. For further discussion see chapter 12.37. Signs of client | liveness are: | + RTCP sender or receiver reports from the client in any of the RTP | sessions part of the RTSP session. | + Any RTSP request which includes a Session header with the ses- | sion's ID. | 10.5 PLAY The PLAY method tells the server to start sending data via the mecha- nism specified in SETUP. A client MUST NOT issue a PLAY request until any outstanding SETUP requests have been acknowledged as successful. In an aggregated session the PLAY request MUST contain an aggregated | control URL. A server SHALL responde with error 460 (Only Aggregate | H. Schulzrinne et. al. [Page 35] Internet Draft RTSP November 01, 2002 Operation Allowed) if the client PLAY request URI is for one of the | media. The media in an aggregate SHALL be played in sync. If a client | want individual control of the media it must use separate RTSP ses- | sions for each media. | The PLAY request positions the normal play time to the beginning of | the range specified by the Range header and delivers stream data | until the end of the range is reached. To allow for precise composi- | tion multiple ranges MAY be specified. The range values are valid if | all given ranges are part of any media. If a given range value points | outside of the media, the response SHALL be the 457 (Invalid Range) | error code. | The below example will first play seconds 10 through 15, then, imme- | diately following, seconds 20 to 25, and finally seconds 30 through | the end. | C->S: PLAY rtsp://audio.example.com/audio RTSP/1.0 | CSeq: 835 | Session: 12345678 | Range: npt=10-15, npt=20-25, 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. The Range header may also contain a time parameter. This parameter | specifies a time in UTC at which the playback should start. If the | message is received after the specified time, playback is started | immediately. The time parameter may be used to aid in synchronization | of streams obtained from different sources. Note: The usage of time | has two problems. First, at the time requested the RTSP state machine | may not accept the request. The client will not get any notification | of the failure. Secondly, the server has difficulties to produce the | synchronization information for the RTP-Info header ahead of the | actually play-out. Due to these reasons it is RECOMMENDED that a | client not issues more than one timed request and no request without | timing , until it is performed. The server SHALL in responses to | timed PLAY request give in the RTP-Info header, the sequence number | of the next RTP packet that will be send for that media, the RTP | timestamp value corresponding to the activation time of the request. | H. Schulzrinne et. al. [Page 36] Internet Draft RTSP November 01, 2002 Unless the session is in paused state and not plays a single media | packet the RTP sequence number will be in error. The RTP timestamp | should be correct unless another timestamp rate has been used in | between the issuing of the request and activation. | For a on-demand stream, the server MUST reply 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 start position SHALL still be returned in the reply. The | unit of the range in the reply is the same as that in the request. If | the medias part of an aggregate has different lengths the PLAY | request and any Range SHALL be performed as long it is valid for the | longest media. Media will be sent whenever it is available for the | given play-out point. | After playing the desired range, the presentation is NOT automati- | cally paused, media deliver simple stops. A PAUSE request MUST be | issued before another PLAY request can issued. Note: This is one | change resulting in a non-operability with RFC 2326 implementations. | A client not issuing a PAUSE request before a new PLAY will be stuck | in PLAYING state. A client desiring to play the media from the begin- | ning MUST send a PLAY request with a Range header pointing at the | beginning, e.g. npt=0-. | 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. Note: The RTP-Info headers has been broken into several lines to fit the page. C->S: PLAY rtsp://audio.example.com/twister.en RTSP/1.0 CSeq: 833 Session: 12345678 Range: smpte=0:10:20-;time=19970123T153600Z S->C: RTSP/1.0 200 OK CSeq: 833 Date: 23 Jan 1997 15:35:06 GMT Server: PhonyServer 1.0 Range: smpte=0:10:22-;time=19970123T153600Z RTP-Info:url=rtsp://example.com/twister.en; seq=14783;rtptime=2345962545 H. Schulzrinne et. al. [Page 37] Internet Draft RTSP November 01, 2002 For playing back a recording of a live presentation, it may be desir- able to use clock units: C->S: PLAY rtsp://audio.example.com/meeting.en RTSP/1.0 CSeq: 835 Session: 12345678 Range: clock=19961108T142300Z-19961108T143520Z S->C: RTSP/1.0 200 OK CSeq: 835 Date: 23 Jan 1997 15:35:06 GMT Server:PhonyServer 1.0 Range: clock=19961108T142300Z-19961108T143520Z RTP-Info:url=rtsp://example.com/meeting.en; seq=53745;rtptime=484589019 A media server only supporting playback MUST support the npt format and MAY support the clock and smpte formats. All range specifiers in this specification allow for ranges with | unspecified begin times (e.g. "npt=-30"). When used in a PLAY | request, the server treats this as a request to start/resume playback | from the current pause point, ending at the end time specified in the | Range header. If the pause point is located later than the given end | value, a 457 (Invalid Range) response SHALL be given. | The queued play functionality described in RFC 2326 [21] is removed | and multiple ranges can be used to achieve a similar performance. If | a server receives a PLAY request while in the PLAY state, the server | SHALL responde using the error code 455 (Method Not Valid In This | State). This will signal the client that queued play are not sup- | ported. | The use of PLAY for keep-alive signaling, i.e. PLAY request without a | range header, has also been decapitated. Instead a client can use, | PING, SET_PARAMETER or OPTIONS for keep alive. A server receiving a | PLAY keep alive SHALL respond with the 455 error code. | When playing live media, indicated by the Transport headers mode | parameter the session are in a live state. This live state will put | some restrictions on the action available for a client. A PLAY | request without a Range header will start media deliver at the cur- | rent point in the live presentation, i.e. now. Any seeking in the | media will be impossible. The only allowed usage of the Range header | H. Schulzrinne et. al. [Page 38] Internet Draft RTSP November 01, 2002 is npt=now-, and certain clock units. The usage of npt=now- is unnec- | essary as it has the exact same meaning as a request without Range | header. The clock format can be used to specify start and stop times | for media delivery in a live session. | 10.6 PAUSE The PAUSE request causes the stream delivery to be interrupted | (halted) temporarily. A PAUSE request MUST be done with the aggre- | gated control URI for aggregated sessions, resulting in all media | being halted, or the media URI for non-aggregated sessions. Any | attempt to do muting of a single media with an PAUSE request in an | aggregated session SHALL be responded with error 460 (Only Aggregate | Operation Allowed). After resuming playback or recording, synchro- | nization of the tracks MUST be maintained. Any server resources are | kept, though servers MAY close the session and free resources after | being paused for the duration specified with the timeout parameter of | the Session header in the SETUP message. Example: C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 834 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 834 Date: 23 Jan 1997 15:35:06 GMT Range: npt=45.76 The PAUSE request may contain a Range header specifying when the | stream or presentation is to be halted. We refer to this point as the | "pause point". The header MUST contain a single value, expressed as | the beginning value an open range. For example, the following clip | will be played from 10 seconds through 21 seconds of the clip's nor- | mal play time, under the assumption that the PAUSE request reaches | the server within 11 seconds of the PLAY request. Note that some | lines has been broken in an non-correct way to fit the page: | C->S: PLAY rtsp://example.com/fizzle/foo RTSP/1.0 | CSeq: 834 | Session: 12345678 | Range: npt=10-30 | H. Schulzrinne et. al. [Page 39] Internet Draft RTSP November 01, 2002 S->C: RTSP/1.0 200 OK | CSeq: 834 | Date: 23 Jan 1997 15:35:06 GMT | Server: PhonyServer 1.0 | Range: npt=10-30 | RTP-Info:url=rtsp://example.com/fizzle/audiotrack; | seq=5712;rtptime=934207921, | url=rtsp://example.com/fizzle/videotrack; | seq=57654;rtptime=2792482193 | Session: 12345678 | C->S: PAUSE rtsp://example.com/fizzle/foo RTSP/1.0 | CSeq: 835 | Session: 12345678 | Range: npt=21- | S->C: RTSP/1.0 200 OK | CSeq: 835 | Date: 23 Jan 1997 15:35:09 GMT | Server: PhonyServer 1.0 | Range: npt=21- | Session: 12345678 | The pause request becomes effective the first time the server is | encountering the time point specified in any of the multiple ranges. | If the Range header specifies a time outside any range from the PLAY | request, the error 457 (Invalid Range) SHALL be returned. If a media | unit (such as an audio or video frame) starts presentation at exactly | the pause point, it is not played or recorded. If the Range header is | missing, stream delivery is interrupted immediately on receipt of the | message and the pause point is set to the current normal play time. | However, the pause point in the media stream MUST be maintained. A | subsequent PLAY request without Range header resumes from the pause | point and play until media end. | The actual pause point after any PAUSE request SHALL be returned to | the client by adding a Range header with what remains unplayed of the | PLAY request's ranges, i.e. including all the remaining ranges part | of multiple range specification. If one desires to resume playing a | ranged request, one simple included the Range header from the PAUSE | response. | For example, if the server have a play request 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 | H. Schulzrinne et. al. [Page 40] Internet Draft RTSP November 01, 2002 first play request, the server stops immediately. If the pause | request is for NPT 16, the server returns a 457 error message. To | prevent that the second range is played and the server stops after | completing the first range, a PAUSE request for 20 must be issued. | As another example, if a server has received requests to play ranges | 10 to 15 and then 13 to 20 (that is, overlapping ranges), the PAUSE | request for NPT=14 would take effect while the server plays the first | range, with the second range effectively being ignored, assuming the | PAUSE request arrives before the server has started playing the sec- | ond, overlapping range. Regardless of when the PAUSE request arrives, | it sets the pause point to 14. | If the server has already sent data beyond the time specified in the | the PAUSE request Range header, a PLAY without range would still | resume at that point in time, specified by the pause's range header, | as it is assumed that the client has discarded data after that point. | This ensures continuous pause/play cycling without gaps. 10.7 TEARDOWN The TEARDOWN request stops the stream delivery for the given URI, | freeing the resources associated with it. If the URI is the aggre- | gated control URI for this presentation, any RTSP session identifier | associated with the session is no longer valid. The use of "*" as URI | in TEARDOWN will also result in that the session is removed indepen- | dent of the number of medias that was part of it. If the URI in the | request was for a media within an aggregated session that media is | removed from the aggregate. However the session and any other media | stream yet not torn down remains, and any valid request, e.g. PLAY or | SETUP, can be issued. As an optional feature a server MAY keep the | session in case the last remaining media is torn down with a TEARDOWN | request with an URI equal to the media URI. To Indicate what has been | performed, a server that after any TEARDOWN request, still has a | valid session MUST in the response return a session header. | A server MAY choose to allow TEARDOWN of individual media while in | PLAY state. When this is not allowed the response SHALL be 455 | (Method Not Valid In This State). If a server implements TEARDOWN and | SETUP in PLAY state it MUST signal this using the "setup.playing" | option tag. | Example: C->S: TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 892 H. Schulzrinne et. al. [Page 41] Internet Draft RTSP November 01, 2002 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 892 Server: PhonyServer 1.0 10.8 GET_PARAMETER The GET_PARAMETER request retrieves the value of a parameter of a | presentation or stream specified in the URI. If the Session header is | present in a request, the value of a parameter MUST be retrieved in | the sessions context. The content of the reply and response is left | to the implementation. GET_PARAMETER with no entity body may be used | to test client or server liveness ("ping"). Example: S->C: GET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 431 Content-Type: text/parameters Session: 12345678 Content-Length: 15 packets_received jitter C->S: RTSP/1.0 200 OK CSeq: 431 Content-Length: 46 Content-Type: text/parameters packets_received: 10 jitter: 0.3838 The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with the intention that the reply content and response content will be defined after further experimentation. 10.9 SET_PARAMETER This method requests to set the value of a parameter for a presenta- tion or stream specified by the URI. H. Schulzrinne et. al. [Page 42] Internet Draft RTSP November 01, 2002 A request is RECOMMENDED to only contain a single parameter to allow | the client to determine why a particular request failed. If the | request contains several parameters, the server MUST only act on the | request if all of the parameters can be set successfully. A server | MUST allow a parameter to be set repeatedly to the same value, but it | MAY disallow changing parameter values. If the receiver of the | request does not understand or can locate a parameter error 451 | (Parameter Not Understood) SHALL be used. In the case a parameter is | not allowed to change the error code 458 (Parameter Is Read-Only). | The response body SHOULD contain only the parameters that has errors. | Otherwise no body SHALL be returned. Note: transport parameters for the media stream MUST only be set with the SETUP command. Restricting setting transport parameters to SETUP is for the benefit of firewalls. The parameters are split in a fine-grained fashion so that there can be more meaningful error indications. However, it may make sense to allow the setting of several parameters if an atomic setting is desirable. Imagine device control where the client does not want the camera to pan unless it can also tilt to the right angle at the same time. Example: C->S: SET_PARAMETER rtsp://example.com/fizzle/foo RTSP/1.0 CSeq: 421 Content-length: 20 Content-type: text/parameters barparam: barstuff S->C: RTSP/1.0 451 Parameter Not Understood CSeq: 421 Content-length: 10 Content-type: text/parameters barparam The "text/parameters" section is only an example type for parameter. This method is intentionally loosely defined with H. Schulzrinne et. al. [Page 43] Internet Draft RTSP November 01, 2002 the intention that the reply content and response content will be defined after further experimentation. 10.10 REDIRECT A redirect request informs the client that it MUST connect to another | server location. REDIRECT SHALL only be sent to the client who cur- | rently has a session at the server. The REDIRECT request MAY contain | the header Location, which indicates that the client should issue | requests for that URL. If the Location URL only contains a host | address the client shall connect to the given host, while using the | path from the URL on the current server. | The redirect request MAY contain the header Range, which indicates | when the redirection takes effect. If the Range contains a time= | value that is the wall clock time that the redirection MUST at the | latest take place. When the time= parameter is present the range | value MUST be ignored. However the range entered MUST be syntactical | correct and SHALL point at the beginning of any on-demand content. If | no time parameter is part of the Range header then redirection SHALL | take place when the media playout from the server reaches the given | time. The range value MUST be a single value in the open ended form, | e.g. npt=59-. | If a Session header is included in the REDIRECT request the client | MUST redirect the indicated session. If no Session header is included | the client MUST redirect all sessions that it have on the server | sending the request. | If the client wants to continue to send or receive media for this | resource, the client MUST issue a TEARDOWN request for the current | session. A new session must be established with the designated host. | A client SHOULD issue a new DESCRIBE request with the URL given in | the Location header, unless the URL only contains a host address. In | the cases the Location only contains a host address the client MAY | assume that the media on the server it is redirected to is identical. | Identical media means that all media configuration information from | the old session still is valid except for the host address. In the | case of absolute URLs in the location header the media redirected to | can be either identical, slightly different or totally different. | This is the reason why a new DESCRIBE request SHOULD be issued. | This example request redirects traffic for this session to the new | server at the given absolute time: | S->C: REDIRECT rtsp://example.com/fizzle/foo RTSP/1.0 | CSeq: 732 | H. Schulzrinne et. al. [Page 44] Internet Draft RTSP November 01, 2002 Location: rtsp://bigserver.com:8001 | Range: clock=19960213T143205Z- | Session: uZ3ci0K+Ld-M | 10.11 RECORD This method initiates recording a range of media data according to the presentation description. The timestamp reflects start and end time (UTC). If no time range is given, use the start or end time pro- vided in the presentation description. If the session has already started, commence recording immediately. The server decides whether to store the recorded data under the request-URI or another URI. If the server does not use the request- URI, the response SHOULD be 201 (Created) and contain an entity which describes the status of the request and refers to the new resource, and a Location header. A media server supporting recording of live presentations MUST sup- port the clock range format; the smpte format does not make sense. In this example, the media server was previously invited to the con- ference indicated. C->S: RECORD rtsp://example.com/meeting/audio.en RTSP/1.0 CSeq: 954 Session: 12345678 Conference: 128.16.64.19/32492374 Note: this example needs work, or needs to be removed. More thoughts on how it works together with ANNOUNCE is needed. Also notification on out of disk is needed. The use of aggre- gated and non-aggregated control needs to be clarified. 10.12 PING This method is a bi-directional mechanism for server or client live- | ness checking. It has no side effects. The issuer of the request MUST | include a session header with the session ID of the session that is | being checked for liveness. | H. Schulzrinne et. al. [Page 45] Internet Draft RTSP November 01, 2002 Prior to using this method, an OPTIONS method is RECOMMENDED to be | issued in the direction which the PING method would be used. This | method MUST NOT be used if support is not indicated by the Public | header. Note: That an 501 (Not Implemented) response means that the | keep-alive timer has not been updated. | When a proxy is in use, PING with a * indicates a single-hop liveness | check, whereas PING with a URL including an host address indicates an | end-to-end liveness check. | Example: | C->S: PING * RTSP/1.0 | CSeq: 123 | Session:12345678 | S->C: RTSP/1.0 200 OK | CSeq: 123 | Session:12345678 | 10.13 Embedded (Interleaved) Binary Data Certain firewall designs and other circumstances may force a server | to interleave RTSP methods and stream data. This interleaving should | generally be avoided unless necessary since it complicates client and | server operation and imposes additional overhead. Also head of line | blocking may cause problems. Interleaved binary data SHOULD only be | used if RTSP is carried over TCP. Stream data such as RTP packets is encapsulated by an ASCII dollar sign (24 decimal), followed by a one-byte channel identifier, fol- lowed by the length of the encapsulated binary data as a binary, two- byte integer in network byte order. The stream data follows immedi- ately afterwards, without a CRLF, but including the upper-layer pro- tocol headers. Each $ block contains exactly one upper-layer protocol data unit, e.g., one RTP packet. The channel identifier is defined in the Transport header with the interleaved parameter(Section 12.40). When the transport choice is RTP, RTCP messages are also interleaved by the server over the TCP connection. As a default, RTCP packets are sent on the first available channel higher than the RTP channel. The client MAY explicitly request RTCP packets on another channel. This is done by specifying two channels in the interleaved parameter of the Transport header(Section 12.40). H. Schulzrinne et. al. [Page 46] Internet Draft RTSP November 01, 2002 RTCP is needed for synchronization when two or more streams are interleaved in such a fashion. Also, this provides a con- venient way to tunnel RTP/RTCP packets through the TCP control connection when required by the network configuration and transfer them onto UDP when possible. C->S: SETUP rtsp://foo.com/bar.file RTSP/1.0 CSeq: 2 Transport: RTP/AVP/TCP;unicast;interleaved=0-1 S->C: RTSP/1.0 200 OK CSeq: 2 Date: 05 Jun 1997 18:57:18 GMT Transport: RTP/AVP/TCP;unicast;interleaved=0-1 Session: 12345678 C->S: PLAY rtsp://foo.com/bar.file RTSP/1.0 CSeq: 3 Session: 12345678 S->C: RTSP/1.0 200 OK CSeq: 3 Session: 12345678 Date: 05 Jun 1997 18:59:15 GMT RTP-Info: url=rtsp://foo.com/bar.file; seq=232433;rtptime=972948234 S->C: $000{2 byte length}{"length" bytes data, w/RTP header} S->C: $000{2 byte length}{"length" bytes data, w/RTP header} S->C: $001{2 byte length}{"length" bytes RTCP packet} 11 Status Code Definitions Where applicable, HTTP status [H10] codes are reused. Status codes that have the same meaning are not repeated here. See Table 1 for a listing of which status codes may be returned by which requests. All error messages, 4xx and 5xx MAY return a body containing further information about the error. 11.1 Success 1xx 11.1.1 100 Continue See, [H10.1.1]. H. Schulzrinne et. al. [Page 47] Internet Draft RTSP November 01, 2002 11.2 Success 2xx 11.2.1 250 Low on Storage Space The server returns this warning after receiving a RECORD request that it may not be able to fulfill completely due to insufficient storage space. If possible, the server should use the Range header to indi- cate what time period it may still be able to record. Since other processes on the server may be consuming storage space simultane- ously, a client should take this only as an estimate. 11.3 Redirection 3xx See [H10.3] for definition of status code 300 to 305. However com- | ments are given for some to how they apply to RTSP. Further a couple | of new status codes are defined. | Within RTSP, redirection may be used for load balancing or redirect- | ing stream requests to a server topologically closer to the client. | Mechanisms to determine topological proximity are beyond the scope of | this specification. | 11.3.1 300 Multiple Choices | 11.3.2 301 Moved Permanently | The request resource are moved permanently and resides now at the URI | given by the location header. The user client SHOULD redirect auto- | matically to the given URI. | 11.3.3 302 Found | The requested resource reside temporarily at the URI given by the | Location header. The Location header MUST be included. | 11.3.4 303 See Other | This status code SHALL NOT be used in RTSP. However as it was allowed | to use in RFC 2326 it is possible that such response will be | received. | 11.3.5 304 Not Modified | 11.3.6 305 Use Proxy | See [H10.3.6]. | H. Schulzrinne et. al. [Page 48] Internet Draft RTSP November 01, 2002 11.3.7 350 Going Away | The server the request was directed at will not be available any | more. This can be for a number of reasons, such as maintenance, or | power failure. If there is a alternative server available the Loca- | tion header SHOULD contain a URI to the same resource at that host. | In case that no server is available the Location header MUST NOT be | included. | In the case the client has an established session on the server giv- | ing the 350 response code, it SHALL immediately do TEARDOWN on that | session. It is RECOMMENDED that the server tries to send REDIRECT | request if possible instead of waiting for a client request to | responde to. | 11.3.8 351 Load Balancing | The server the request was issued for is currently uneven loaded and | request that further request is directed to another server. The | Location header MUST be included in the response and contain the URI | of the other server. If the both server has the requested resource in | the same place only the Server part of the URI MAY be given. In all | other cases an absolute URI MUST be given. 11.4 Client Error 4xx 11.4.1 400 Bad Request The request could not be understood by the server due to malformed syntax. The client SHOULD NOT repeat the request without modifica- tions [H10.4.1]. If the request does not have a CSeq header, the server MUST NOT include a CSeq in the response. 11.4.2 405 Method Not Allowed The method specified in the request is not allowed for the resource identified by the request URI. The response MUST include an Allow header containing a list of valid methods for the requested resource. This status code is also to be used if a request attempts to use a method not indicated during SETUP, e.g., if a RECORD request is issued even though the mode parameter in the Transport header only specified PLAY. 11.4.3 451 Parameter Not Understood The recipient of the request does not support one or more parameters | contained in the request.When returning this error message the sender | SHOULD return a entity body containing the offending parameter(s). H. Schulzrinne et. al. [Page 49] Internet Draft RTSP November 01, 2002 11.4.4 452 reserved This error code was removed from RFC 2326 [21] and is obsolete. 11.4.5 453 Not Enough Bandwidth The request was refused because there was insufficient bandwidth. This may, for example, be the result of a resource reservation fail- ure. 11.4.6 454 Session Not Found The RTSP session identifier in the Session header is missing, invalid, or has timed out. 11.4.7 455 Method Not Valid in This State The client or server cannot process this request in its current state. The response SHOULD contain an Allow header to make error recovery easier. 11.4.8 456 Header Field Not Valid for Resource The server could not act on a required request header. For example, | if PLAY contains the Range header field but the stream does not allow | seeking. This error message may also be used for specifying when the | time format in Range is impossible for the resource. In that case the | Accept-Ranges header SHOULD be returned to inform the client of which | format(s) that are allowed. 11.4.9 457 Invalid Range The Range value given is out of bounds, e.g., beyond the end of the presentation. 11.4.10 458 Parameter Is Read-Only The parameter to be set by SET_PARAMETER can be read but not modi- | fied. When returning this error message the sender SHOULD return a | entity body containing the offending parameter(s). 11.4.11 459 Aggregate Operation Not Allowed The requested method may not be applied on the URL in question since it is an aggregate (presentation) URL. The method may be applied on a media URL. H. Schulzrinne et. al. [Page 50] Internet Draft RTSP November 01, 2002 11.4.12 460 Only Aggregate Operation Allowed The requested method may not be applied on the URL in question since | it is not an aggregate control (presentation) URL. The method may be | applied on the aggregate control URL. 11.4.13 461 Unsupported Transport The Transport field did not contain a supported transport specifica- tion. 11.4.14 462 Destination Unreachable The data transmission channel could not be established because the client address could not be reached. This error will most likely be the result of a client attempt to place an invalid Destination param- eter in the Transport field. 11.5 Server Error 5xx 11.5.1 551 Option not supported An option given in the Require or the Proxy-Require fields was not supported. The Unsupported header SHOULD be returned stating the option for which there is no support. 12 Header Field Definitions The general syntax for header fields is covered in Section 4.2 This | section lists the full set of header fields along with notes on syn- tax, meaning, and usage. Throughout this section, we use [HX.Y] to refer to Section X.Y of the current HTTP/1.1 specification RFC 2616 [26]. Examples of each header field are given. Information about header fields in relation to methods and proxy pro- cessing is summarized in Table 4 and Table 5. The "where" column describes the request and response types in which the header field can be used. Values in this column are: R: header field may only appear in requests; r: header field may only appear in responses; 2xx, 4xx, etc.: A numerical value or range indicates response codes with which the header field can be used; H. Schulzrinne et. al. [Page 51] Internet Draft RTSP November 01, 2002 method direction object acronym Body ----------------------------------------------- DESCRIBE C->S P,S DES r ANNOUNCE C->S, S->C P,S ANN R GET_PARAMETER C->S, S->C P,S GPR R,r OPTIONS C->S P,S OPT S->C PAUSE C->S P,S PSE PING C->S, S->C P,S PNG PLAY C->S P,S PLY RECORD C->S P,S REC REDIRECT S->C P,S RDR SETUP C->S S STP SET_PARAMETER C->S, S->C P,S SPR R,r TEARDOWN C->S P,S TRD Table 3: Overview of RTSP methods, their direction, and what objects (P: presentation, S: stream) they operate on. Body notes if a method is allowed to carry body and in which direction, R = Request, r=response. Note: It is allowed for all error messages 4xx and 5xx to have a body c: header field is copied from the request to the response. An empty entry in the "where" column indicates that the header field may be present in all requests and responses. The "proxy" column describes the operations a proxy may perform on a header field: a: A proxy can add or concatenate the header field if not present. m: A proxy can modify an existing header field value. d: A proxy can delete a header field value. r: A proxy must be able to read the header field, and thus this header field cannot be encrypted. The rest of the columns relate to the presence of a header field in a method. The method names when abbreviated, are according to table 3: c: Conditional; requirements on the header field depend on the con- text of the message. H. Schulzrinne et. al. [Page 52] Internet Draft RTSP November 01, 2002 m: The header field is mandatory. m*: The header field SHOULD be sent, but clients/servers need to be prepared to receive messages without that header field. o: The header field is optional. *: The header field is required if the message body is not empty. See sections 12.14, 12.16 and 4.3 for details. -: The header field is not applicable. "Optional" means that a Client/Server MAY include the header field in a request or response, and a Client/Server MAY ignore the header field if present in the request or response (The exception to this rule is the Require header field discussed in 12.32). A "mandatory" header field MUST be present in a request, and MUST be understood by the Client/Server receiving the request. A mandatory response header field MUST be present in the response, and the header field MUST be understood by the Client/Server processing the response. "Not appli- cable" means that the header field MUST NOT be present in a request. If one is placed in a request by mistake, it MUST be ignored by the Client/Server receiving the request. Similarly, a header field labeled "not applicable" for a response means that the Client/Server MUST NOT place the header field in the response, and the Client/Server MUST ignore the header field in the response. A Client/Server SHOULD ignore extension header parameters that are not understood. The From, Location, and RTP-Info header fields contain a URI. If the URI contains a comma, or semicolon, the URI MUST be enclosed in dou- ble quotas ("). Any URI parameters are contained within these quotas. If the URI is not enclosed in double quotas, any semicolon- delimited parameters are header-parameters, not URI parameters. 12.1 Accept | The Accept request-header field can be used to specify certain pre- sentation description content types which are acceptable for the response. The "level" parameter for presentation descriptions is prop- erly defined as part of the MIME type registration, not here. H. Schulzrinne et. al. [Page 53] Internet Draft RTSP November 01, 2002 Header Where Proxy DES OPT SETUP PLAY PAUSE TRD -------------------------------------------------------------- Accept R o - - - - - Accept-Encoding R r o - - - - - Accept-Language R r o - - - - - Accept-Ranges r r - - o - - - Accept-Ranges 456 r - - - o o - Allow r - o - - - - Allow 405 - - - m m - Authorization R o o o o o o Bandwidth R o o o o - - Blocksize R o - o o - - Cache-Control r - - o - - - Connection o o o o o o Content-Base r o - - - - - Content-Base 4xx o o o o o o Content-Encoding R r - - - - - - Content-Encoding r r o - - - - - Content-Encoding 4xx r o o o o o o Content-Language R r - - - - - - Content-Language r r o - - - - - Content-Language 4xx r o o o o o o Content-Length r r * - - - - - Content-Length 4xx r * * * * * * Content-Location r o - - - - - Content-Location 4xx o o o o o o Content-Type r * - - - - - Content-Type 4xx * * * * * * CSeq Rc m m m m m m Date am o o o o o o Expires r r o - - - - - From R r o o o o o o Host o o o o o o If-Match R r - - o - - - If-Modified-Since R r o - o - - - Last-Modified r r o - - - - - Location 3xx o - o - - - Proxy-Authenticate 407 amr m m m m m m Proxy-Require R ar o o o o o o Public r admr - m* - - - - Public 501 admr m* m* m* m* m* m* Range R - - - o o - Range r - - c m* - - Referer R o o o o o o Require R o o o o o o H. Schulzrinne et. al. [Page 54] Internet Draft RTSP November 01, 2002 Retry-After 3xx,503 o o o - - - RTP-Info r - - o m - - Scale - - - o - - Session R - o o m m m Session r - c m m m o Server R - o - - - - Server r o o o o o o Speed - - - o - - Supported R o o o o o o Supported r c c c c c c Timestamp R o o o o o o Timestamp c m m m m m m Transport - - m - - - Unsupported r c c c c c c User-Agent R m* m* m* m* m* m* Vary r c c c c c c Via R amr o o o o o o Via c dr m m m m m m WWW-Authenticate 401 m m m m m m -------------------------------------------------------------- Header Where Proxy DES OPT SETUP PLAY PAUSE TRD Table 4: Overview of RTSP header fields related to methods DESCRIBE, OPTIONS, SETUP, PLAY, PAUSE, and TEARDOWN. See [H14.1] for syntax. Example of use: Accept: application/rtsl q=1.0, application/sdp;level=2 12.2 Accept-Encoding See [H14.3] 12.3 Accept-Language See [H14.4]. Note that the language specified applies to the presen- tation description and any reason phrases, not the media content. 12.4 Accept-Ranges | H. Schulzrinne et. al. [Page 55] Internet Draft RTSP November 01, 2002 Header Where Proxy GPR SPR ANN REC RDR PNG --------------------------------------------------------- Allow 405 - - m m - - Authorization R o o o o o o Bandwidth R - o - - - - Blocksize R - o - - - - Connection o o o o o - Content-Base R o o o - - - Content-Base r o o - - - - Content-Base 4xx o o o o o - Content-Encoding R r o o o - - - Content-Encoding r r o o - - - - Content-Encoding 4xx r o o o o o - Content-Language R r o o o - - - Content-Language r r o o - - - - Content-Language 4xx r o o o o o - Content-Length R r * * * - - - Content-Length r r * * - - - - Content-Length 4xx r * * * * * - Content-Location R o o o - - - Content-Location r o o - - - - Content-Location 4xx o o o o o - Content-Type R * * * - - - Content-Type r * * - - - - Content-Type 4xx * * * * * - CSeq Rc m m m m m m Date am o o o o o o From R r o o o o o o Host o o o o o o Last-Modified R r - - o - - - Last-Modified r r o - - - - - Location R - - - - m - Proxy-Authenticate 407 amr m m m m m m Proxy-Require R ar o o o o o o Public 501 admr m* m* m* m* m* m* Range R - - - - o - Referer R o o o o o - Require R o o o o o o Retry-After 3xx,503 o o - - - - Scale - - - o - - Session R o o m m o m Session r c c m m o m Server R o o o - o o Server r o o o o - o Supported R o o o o o o H. Schulzrinne et. al. [Page 56] Internet Draft RTSP November 01, 2002 Supported r c c c c c c Timestamp R o o o o o o Timestamp c m m m m m m Unsupported r c c c c c c User-Agent R m* m* m* m* - m* User-Agent r - - - - m* - Vary r c c c c - - Via R amr o o o o o o Via c dr m m m m m m WWW-Authenticate 401 m m m m m m --------------------------------------------------------- Header Where Proxy GPR SPR ANN REC RDR PNG Table 5: Overview of RTSP header fields related to methods GET_PARAM- ETER, SET_PARAMETER, ANNOUNCE, RECORD, REDIRECT, and PING. The Accept-Ranges response-header field allows the server to indicate | its acceptance of range requests and possible formats for a resource: | Accept-Ranges = "Accept-Ranges" ":" || acceptable-ranges || acceptable-ranges = 1#range-unit / "none" || range-unit = NPT / SMPTE / UTC / LIVE || This header has the same syntax as [H14.5]. However new range-units | are defined and byte-ranges SHALL NOT be used. Inclusion of any of | the three time formats indicates acceptance by the server for PLAY | and PAUSE requests with this format. Inclusion of the "LIVE" tag | indicates that the resource has LIVE properties. The headers value is | valid for the resource specified by the URI in the request, this | response corresponds to. | A server is RECOMMENDED to use this header in SETUP responses to | indicate to the client which range time formats the media supports. | The header SHOULD also be included in "456" responses which is a | result of use of unsupported range formats. | 12.5 Allow The Allow entity-header field lists the methods supported by the resource identified by the request-URI. The purpose of this field is to strictly inform the recipient of valid methods associated with the H. Schulzrinne et. al. [