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RFC 2326


Real Time Streaming Protocol (RTSP)

Part 4 of 4, p. 72 to 92
Prev RFC Part


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15 Syntax

   The RTSP syntax is described in an augmented Backus-Naur form (BNF)
   as used in RFC 2068 [2].

15.1 Base Syntax

   OCTET              =      <any 8-bit sequence of data>
   CHAR               =      <any US-ASCII character (octets 0 - 127)>
   UPALPHA            =      <any US-ASCII uppercase letter "A".."Z">
   LOALPHA            =      <any US-ASCII lowercase letter "a".."z">
   ALPHA              =      UPALPHA | LOALPHA

   DIGIT              =      <any US-ASCII digit "0".."9">
   CTL                =      <any US-ASCII control character
                              (octets 0 - 31) and DEL (127)>
   CR                 =      <US-ASCII CR, carriage return (13)>
   LF                 =      <US-ASCII LF, linefeed (10)>

   SP                 =      <US-ASCII SP, space (32)>
   HT                 =      <US-ASCII HT, horizontal-tab (9)>
   <">                =      <US-ASCII double-quote mark (34)>
   CRLF               =      CR LF

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   LWS                =      [CRLF] 1*( SP | HT )
   TEXT               =      <any OCTET except CTLs>
   tspecials          =      "(" | ")" | "<" | ">" | "@"
                      |       "," | ";" | ":" | "\" | <">
                      |       "/" | "[" | "]" | "?" | "="
                      |       "{" | "}" | SP | HT

   token              =      1*<any CHAR except CTLs or tspecials>
   quoted-string      =      ( <"> *(qdtext) <"> )
   qdtext             =      <any TEXT except <">>
   quoted-pair        =      "\" CHAR

   message-header     =      field-name ":" [ field-value ] CRLF
   field-name         =      token
   field-value        =      *( field-content | LWS )
   field-content      =      <the OCTETs making up the field-value and
                              consisting of either *TEXT or
                              combinations of token, tspecials, and

   safe               =  "\$" | "-" | "_" | "." | "+"
   extra              =  "!" | "*" | "$'$" | "(" | ")" | ","

   hex                =  DIGIT | "A" | "B" | "C" | "D" | "E" | "F" |
                        "a" | "b" | "c" | "d" | "e" | "f"
   escape             =  "\%" hex hex
   reserved           =  ";" | "/" | "?" | ":" | "@" | "&" | "="

   unreserved         =  alpha | digit | safe | extra
   xchar              =  unreserved | reserved | escape

16 Security Considerations

   Because of the similarity in syntax and usage between RTSP servers
   and HTTP servers, the security considerations outlined in [H15]
   apply.  Specifically, please note the following:

   Authentication Mechanisms:
          RTSP and HTTP share common authentication schemes, and thus
          should follow the same prescriptions with regards to
          authentication. See [H15.1] for client authentication issues,
          and [H15.2] for issues regarding support for multiple
          authentication mechanisms.

   Abuse of Server Log Information:
          RTSP and HTTP servers will presumably have similar logging
          mechanisms, and thus should be equally guarded in protecting
          the contents of those logs, thus protecting the privacy of the

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          users of the servers. See [H15.3] for HTTP server
          recommendations regarding server logs.

   Transfer of Sensitive Information:
          There is no reason to believe that information transferred via
          RTSP may be any less sensitive than that normally transmitted
          via HTTP. Therefore, all of the precautions regarding the
          protection of data privacy and user privacy apply to
          implementors of RTSP clients, servers, and proxies. See
          [H15.4] for further details.

   Attacks Based On File and Path Names:
          Though RTSP URLs are opaque handles that do not necessarily
          have file system semantics, it is anticipated that many
          implementations will translate portions of the request URLs
          directly to file system calls. In such cases, file systems
          SHOULD follow the precautions outlined in [H15.5], such as
          checking for ".." in path components.

   Personal Information:
          RTSP clients are often privy to the same information that HTTP
          clients are (user name, location, etc.) and thus should be
          equally. See [H15.6] for further recommendations.

   Privacy Issues Connected to Accept Headers:
          Since may of the same "Accept" headers exist in RTSP as in
          HTTP, the same caveats outlined in [H15.7] with regards to
          their use should be followed.

   DNS Spoofing:
          Presumably, given the longer connection times typically
          associated to RTSP sessions relative to HTTP sessions, RTSP
          client DNS optimizations should be less prevalent.
          Nonetheless, the recommendations provided in [H15.8] are still
          relevant to any implementation which attempts to rely on a
          DNS-to-IP mapping to hold beyond a single use of the mapping.

   Location Headers and Spoofing:
          If a single server supports multiple organizations that do not
          trust one another, then it must check the values of Location
          and Content-Location headers in responses that are generated
          under control of said organizations to make sure that they do
          not attempt to invalidate resources over which they have no
          authority. ([H15.9])

   In addition to the recommendations in the current HTTP specification
   (RFC 2068 [2], as of this writing), future HTTP specifications may
   provide additional guidance on security issues.

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   The following are added considerations for RTSP implementations.

   Concentrated denial-of-service attack:
          The protocol offers the opportunity for a remote-controlled
          denial-of-service attack. The attacker may initiate traffic
          flows to one or more IP addresses by specifying them as the
          destination in SETUP requests. While the attacker's IP address
          may be known in this case, this is not always useful in
          prevention of more attacks or ascertaining the attackers
          identity. Thus, an RTSP server SHOULD only allow client-
          specified destinations for RTSP-initiated traffic flows if the
          server has verified the client's identity, either against a
          database of known users using RTSP authentication mechanisms
          (preferably digest authentication or stronger), or other
          secure means.

   Session hijacking:
          Since there is no relation between a transport layer
          connection and an RTSP session, it is possible for a malicious
          client to issue requests with random session identifiers which
          would affect unsuspecting clients. The server SHOULD use a
          large, random and non-sequential session identifier to
          minimize the possibility of this kind of attack.

          Servers SHOULD implement both basic and digest [8]
          authentication. In environments requiring tighter security for
          the control messages, the RTSP control stream may be

   Stream issues:
          RTSP only provides for stream control. Stream delivery issues
          are not covered in this section, nor in the rest of this memo.
          RTSP implementations will most likely rely on other protocols
          such as RTP, IP multicast, RSVP and IGMP, and should address
          security considerations brought up in those and other
          applicable specifications.

   Persistently suspicious behavior:
          RTSP servers SHOULD return error code 403 (Forbidden) upon
          receiving a single instance of behavior which is deemed a
          security risk. RTSP servers SHOULD also be aware of attempts
          to probe the server for weaknesses and entry points and MAY
          arbitrarily disconnect and ignore further requests clients
          which are deemed to be in violation of local security policy.

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Appendix A: RTSP Protocol State Machines

   The RTSP client and server state machines describe the behavior of
   the protocol from RTSP session initialization through RTSP session

   State is defined on a per object basis. An object is uniquely
   identified by the stream URL and the RTSP session identifier. Any
   request/reply using aggregate URLs denoting RTSP presentations
   composed of multiple streams will have an effect on the individual
   states of all the streams. For example, if the presentation /movie
   contains two streams, /movie/audio and /movie/video, then the
   following command:

     PLAY rtsp:// RTSP/1.0
     CSeq: 559
     Session: 12345678

   will have an effect on the states of movie/audio and movie/video.

     This example does not imply a standard way to represent streams in
     URLs or a relation to the filesystem. See Section 3.2.

   SET_PARAMETER do not have any effect on client or server state and
   are therefore not listed in the state tables.

A.1 Client State Machine

   The client can assume the following states:

          SETUP has been sent, waiting for reply.

          SETUP reply received or PAUSE reply received while in Playing

          PLAY reply received

          RECORD reply received

   In general, the client changes state on receipt of replies to
   requests. Note that some requests are effective at a future time or
   position (such as a PAUSE), and state also changes accordingly. If no
   explicit SETUP is required for the object (for example, it is

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   available via a multicast group), state begins at Ready. In this
   case, there are only two states, Ready and Playing. The client also
   changes state from Playing/Recording to Ready when the end of the
   requested range is reached.

   The "next state" column indicates the state assumed after receiving a
   success response (2xx). If a request yields a status code of 3xx, the
   state becomes Init, and a status code of 4xx yields no change in
   state. Messages not listed for each state MUST NOT be issued by the
   client in that state, with the exception of messages not affecting
   state, as listed above. Receiving a REDIRECT from the server is
   equivalent to receiving a 3xx redirect status from the server.

   state       message sent     next state after response
   Init        SETUP            Ready
               TEARDOWN         Init
   Ready       PLAY             Playing
               RECORD           Recording
               TEARDOWN         Init
               SETUP            Ready
   Playing     PAUSE            Ready
               TEARDOWN         Init
               PLAY             Playing
               SETUP            Playing (changed transport)
   Recording   PAUSE            Ready
               TEARDOWN         Init
               RECORD           Recording
               SETUP            Recording (changed transport)

A.2 Server State Machine

   The server can assume the following states:

          The initial state, no valid SETUP has been received yet.

          Last SETUP received was successful, reply sent or after
          playing, last PAUSE received was successful, reply sent.

          Last PLAY received was successful, reply sent. Data is being

          The server is recording media data.

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   In general, the server changes state on receiving requests. If the
   server is in state Playing or Recording and in unicast mode, it MAY
   revert to Init and tear down the RTSP session if it has not received
   "wellness" information, such as RTCP reports or RTSP commands, from
   the client for a defined interval, with a default of one minute. The
   server can declare another timeout value in the Session response
   header (Section 12.37). If the server is in state Ready, it MAY
   revert to Init if it does not receive an RTSP request for an interval
   of more than one minute. Note that some requests (such as PAUSE) may
   be effective at a future time or position, and server state changes
   at the appropriate time. The server reverts from state Playing or
   Recording to state Ready at the end of the range requested by the

   The REDIRECT message, when sent, is effective immediately unless it
   has a Range header specifying when the redirect is effective. In such
   a case, server state will also change at the appropriate time.

   If no explicit SETUP is required for the object, the state starts at
   Ready and there are only two states, Ready and Playing.

   The "next state" column indicates the state assumed after sending a
   success response (2xx). If a request results in a status code of 3xx,
   the state becomes Init. A status code of 4xx results in no change.

     state           message received  next state
     Init            SETUP             Ready
                     TEARDOWN          Init
     Ready           PLAY              Playing
                     SETUP             Ready
                     TEARDOWN          Init
                     RECORD            Recording
     Playing         PLAY              Playing
                     PAUSE             Ready
                     TEARDOWN          Init
                     SETUP             Playing
     Recording       RECORD            Recording
                     PAUSE             Ready
                     TEARDOWN          Init
                     SETUP             Recording

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Appendix B: Interaction with RTP

   RTSP allows media clients to control selected, non-contiguous
   sections of media presentations, rendering those streams with an RTP
   media layer[24]. The media layer rendering the RTP stream should not
   be affected by jumps in NPT. Thus, both RTP sequence numbers and RTP
   timestamps MUST be continuous and monotonic across jumps of NPT.

   As an example, assume a clock frequency of 8000 Hz, a packetization
   interval of 100 ms and an initial sequence number and timestamp of
   zero. First we play NPT 10 through 15, then skip ahead and play NPT
   18 through 20. The first segment is presented as RTP packets with
   sequence numbers 0 through 49 and timestamp 0 through 39,200. The
   second segment consists of RTP packets with sequence number 50
   through 69, with timestamps 40,000 through 55,200.

     We cannot assume that the RTSP client can communicate with the RTP
     media agent, as the two may be independent processes. If the RTP
     timestamp shows the same gap as the NPT, the media agent will
     assume that there is a pause in the presentation. If the jump in
     NPT is large enough, the RTP timestamp may roll over and the media
     agent may believe later packets to be duplicates of packets just
     played out.

     For certain datatypes, tight integration between the RTSP layer and
     the RTP layer will be necessary. This by no means precludes the
     above restriction. Combined RTSP/RTP media clients should use the
     RTP-Info field to determine whether incoming RTP packets were sent
     before or after a seek.

   For continuous audio, the server SHOULD set the RTP marker bit at the
   beginning of serving a new PLAY request. This allows the client to
   perform playout delay adaptation.

   For scaling (see Section 12.34), RTP timestamps should correspond to
   the playback timing. For example, when playing video recorded at 30
   frames/second at a scale of two and speed (Section 12.35) of one, the
   server would drop every second frame to maintain and deliver video
   packets with the normal timestamp spacing of 3,000 per frame, but NPT
   would increase by 1/15 second for each video frame.

   The client can maintain a correct display of NPT by noting the RTP
   timestamp value of the first packet arriving after repositioning. The
   sequence parameter of the RTP-Info (Section 12.33) header provides
   the first sequence number of the next segment.

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Appendix C: Use of SDP for RTSP Session Descriptions

   The Session Description Protocol (SDP, RFC 2327 [6]) may be used to
   describe streams or presentations in RTSP. Such usage is limited to
   specifying means of access and encoding(s) for:

   aggregate control:
          A presentation composed of streams from one or more servers
          that are not available for aggregate control. Such a
          description is typically retrieved by HTTP or other non-RTSP
          means. However, they may be received with ANNOUNCE methods.

   non-aggregate control:
          A presentation composed of multiple streams from a single
          server that are available for aggregate control. Such a
          description is typically returned in reply to a DESCRIBE
          request on a URL, or received in an ANNOUNCE method.

   This appendix describes how an SDP file, retrieved, for example,
   through HTTP, determines the operation of an RTSP session. It also
   describes how a client should interpret SDP content returned in reply
   to a DESCRIBE request. SDP provides no mechanism by which a client
   can distinguish, without human guidance, between several media
   streams to be rendered simultaneously and a set of alternatives
   (e.g., two audio streams spoken in different languages).

C.1 Definitions

   The terms "session-level", "media-level" and other key/attribute
   names and values used in this appendix are to be used as defined in
   SDP (RFC 2327 [6]):

C.1.1 Control URL

   The "a=control:" attribute is used to convey the control URL. This
   attribute is used both for the session and media descriptions. If
   used for individual media, it indicates the URL to be used for
   controlling that particular media stream. If found at the session
   level, the attribute indicates the URL for aggregate control.


   This attribute may contain either relative and absolute URLs,
   following the rules and conventions set out in RFC 1808 [25].
   Implementations should look for a base URL in the following order:

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   1.     The RTSP Content-Base field
   2.     The RTSP Content-Location field
   3.     The RTSP request URL

   If this attribute contains only an asterisk (*), then the URL is
   treated as if it were an empty embedded URL, and thus inherits the
   entire base URL.

C.1.2 Media streams

   The "m=" field is used to enumerate the streams. It is expected that
   all the specified streams will be rendered with appropriate
   synchronization. If the session is unicast, the port number serves as
   a recommendation from the server to the client; the client still has
   to include it in its SETUP request and may ignore this
   recommendation.  If the server has no preference, it SHOULD set the
   port number value to zero.

     m=audio 0 RTP/AVP 31

C.1.3 Payload type(s)

   The payload type(s) are specified in the "m=" field. In case the
   payload type is a static payload type from RFC 1890 [1], no other
   information is required. In case it is a dynamic payload type, the
   media attribute "rtpmap" is used to specify what the media is. The
   "encoding name" within the "rtpmap" attribute may be one of those
   specified in RFC 1890 (Sections 5 and 6), or an experimental encoding
   with a "X-" prefix as specified in SDP (RFC 2327 [6]).  Codec-
   specific parameters are not specified in this field, but rather in
   the "fmtp" attribute described below. Implementors seeking to
   register new encodings should follow the procedure in RFC 1890 [1].
   If the media type is not suited to the RTP AV profile, then it is
   recommended that a new profile be created and the appropriate profile
   name be used in lieu of "RTP/AVP" in the "m=" field.

C.1.4 Format-specific parameters

   Format-specific parameters are conveyed using the "fmtp" media
   attribute. The syntax of the "fmtp" attribute is specific to the
   encoding(s) that the attribute refers to. Note that the packetization
   interval is conveyed using the "ptime" attribute.

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C.1.5 Range of presentation

   The "a=range" attribute defines the total time range of the stored
   session. (The length of live sessions can be deduced from the "t" and
   "r" parameters.) Unless the presentation contains media streams of
   different durations, the range attribute is a session-level
   attribute. The unit is specified first, followed by the value range.
   The units and their values are as defined in Section 3.5, 3.6 and


C.1.6 Time of availability

   The "t=" field MUST contain suitable values for the start and stop
   times for both aggregate and non-aggregate stream control. With
   aggregate control, the server SHOULD indicate a stop time value for
   which it guarantees the description to be valid, and a start time
   that is equal to or before the time at which the DESCRIBE request was
   received. It MAY also indicate start and stop times of 0, meaning
   that the session is always available. With non-aggregate control, the
   values should reflect the actual period for which the session is
   available in keeping with SDP semantics, and not depend on other
   means (such as the life of the web page containing the description)
   for this purpose.

C.1.7 Connection Information

   In SDP, the "c=" field contains the destination address for the media
   stream. However, for on-demand unicast streams and some multicast
   streams, the destination address is specified by the client via the
   SETUP request. Unless the media content has a fixed destination
   address, the "c=" field is to be set to a suitable null value. For
   addresses of type "IP4", this value is "".

C.1.8 Entity Tag

   The optional "a=etag" attribute identifies a version of the session
   description. It is opaque to the client. SETUP requests may include
   this identifier in the If-Match field (see section 12.22) to only
   allow session establishment if this attribute value still corresponds
   to that of the current description. The attribute value is opaque and
   may contain any character allowed within SDP attribute values.


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     One could argue that the "o=" field provides identical
     functionality. However, it does so in a manner that would put
     constraints on servers that need to support multiple session
     description types other than SDP for the same piece of media

C.2 Aggregate Control Not Available

   If a presentation does not support aggregate control and multiple
   media sections are specified, each section MUST have the control URL
   specified via the "a=control:" attribute.

     o=- 2890844256 2890842807 IN IP4
     s=I came from a web page
     t=0 0
     c=IN IP4
     m=video 8002 RTP/AVP 31
     m=audio 8004 RTP/AVP 3

   Note that the position of the control URL in the description implies
   that the client establishes separate RTSP control sessions to the
   servers and

   It is recommended that an SDP file contains the complete media
   initialization information even if it is delivered to the media
   client through non-RTSP means. This is necessary as there is no
   mechanism to indicate that the client should request more detailed
   media stream information via DESCRIBE.

C.3 Aggregate Control Available

   In this scenario, the server has multiple streams that can be
   controlled as a whole. In this case, there are both media-level
   "a=control:" attributes, which are used to specify the stream URLs,
   and a session-level "a=control:" attribute which is used as the
   request URL for aggregate control. If the media-level URL is
   relative, it is resolved to absolute URLs according to Section C.1.1

   If the presentation comprises only a single stream, the media-level
   "a=control:" attribute may be omitted altogether. However, if the
   presentation contains more than one stream, each media stream section
   MUST contain its own "a=control" attribute.

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     o=- 2890844256 2890842807 IN IP4
     s=I contain
     i=<more info>
     t=0 0
     c=IN IP4
     m=video 8002 RTP/AVP 31
     m=audio 8004 RTP/AVP 3

   In this example, the client is required to establish a single RTSP
   session to the server, and uses the URLs
   rtsp:// and
   rtsp:// to set up the video and audio
   streams, respectively. The URL rtsp:// controls the
   whole movie.

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Appendix D: Minimal RTSP implementation

D.1 Client

   A client implementation MUST be able to do the following :

     * Generate the following requests: SETUP, TEARDOWN, and one of PLAY
       (i.e., a minimal playback client) or RECORD (i.e., a minimal
       recording client). If RECORD is implemented, ANNOUNCE must be
       implemented as well.
     * Include the following headers in requests: CSeq, Connection,
       Session, Transport. If ANNOUNCE is implemented, the capability to
       include headers Content-Language, Content-Encoding, Content-
       Length, and Content-Type should be as well.
     * Parse and understand the following headers in responses: CSeq,
       Connection, Session, Transport, Content-Language, Content-
       Encoding, Content-Length, Content-Type. If RECORD is implemented,
       the Location header must be understood as well.  RTP-compliant
       implementations should also implement RTP-Info.
     * Understand the class of each error code received and notify the
       end-user, if one is present, of error codes in classes 4xx and
       5xx. The notification requirement may be relaxed if the end-user
       explicitly does not want it for one or all status codes.
     * Expect and respond to asynchronous requests from the server, such
       as ANNOUNCE. This does not necessarily mean that it should
       implement the ANNOUNCE method, merely that it MUST respond
       positively or negatively to any request received from the server.

   Though not required, the following are highly recommended at the time
   of publication for practical interoperability with initial
   implementations and/or to be a "good citizen".

     * Implement RTP/AVP/UDP as a valid transport.
     * Inclusion of the User-Agent header.
     * Understand SDP session descriptions as defined in Appendix C
     * Accept media initialization formats (such as SDP) from standard
       input, command line, or other means appropriate to the operating
       environment to act as a "helper application" for other
       applications (such as web browsers).

     There may be RTSP applications different from those initially
     envisioned by the contributors to the RTSP specification for which
     the requirements above do not make sense. Therefore, the
     recommendations above serve only as guidelines instead of strict

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D.1.1 Basic Playback

   To support on-demand playback of media streams, the client MUST
   additionally be able to do the following:
     * generate the PAUSE request;
     * implement the REDIRECT method, and the Location header.

D.1.2 Authentication-enabled

   In order to access media presentations from RTSP servers that require
   authentication, the client MUST additionally be able to do the
     * recognize the 401 status code;
     * parse and include the WWW-Authenticate header;
     * implement Basic Authentication and Digest Authentication.

D.2 Server

   A minimal server implementation MUST be able to do the following:

     * Implement the following methods: SETUP, TEARDOWN, OPTIONS and
       either PLAY (for a minimal playback server) or RECORD (for a
       minimal recording server).  If RECORD is implemented, ANNOUNCE
       should be implemented as well.
     * Include the following headers in responses: Connection,
       Content-Length, Content-Type, Content-Language, Content-Encoding,
       Transport, Public. The capability to include the Location header
       should be implemented if the RECORD method is. RTP-compliant
       implementations should also implement the RTP-Info field.
     * Parse and respond appropriately to the following headers in
       requests: Connection, Session, Transport, Require.

   Though not required, the following are highly recommended at the time
   of publication for practical interoperability with initial
   implementations and/or to be a "good citizen".

     * Implement RTP/AVP/UDP as a valid transport.
     * Inclusion of the Server header.
     * Implement the DESCRIBE method.
     * Generate SDP session descriptions as defined in Appendix C

     There may be RTSP applications different from those initially
     envisioned by the contributors to the RTSP specification for which
     the requirements above do not make sense. Therefore, the
     recommendations above serve only as guidelines instead of strict

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D.2.1 Basic Playback

   To support on-demand playback of media streams, the server MUST
   additionally be able to do the following:

     * Recognize the Range header, and return an error if seeking is not
     * Implement the PAUSE method.

   In addition, in order to support commonly-accepted user interface
   features, the following are highly recommended for on-demand media

     * Include and parse the Range header, with NPT units.
       Implementation of SMPTE units is recommended.
     * Include the length of the media presentation in the media
       initialization information.
     * Include mappings from data-specific timestamps to NPT. When RTP
       is used, the rtptime portion of the RTP-Info field may be used to
       map RTP timestamps to NPT.

     Client implementations may use the presence of length information
     to determine if the clip is seekable, and visibly disable seeking
     features for clips for which the length information is unavailable.
     A common use of the presentation length is to implement a "slider
     bar" which serves as both a progress indicator and a timeline
     positioning tool.

     Mappings from RTP timestamps to NPT are necessary to ensure correct
     positioning of the slider bar.

D.2.2 Authentication-enabled

   In order to correctly handle client authentication, the server MUST
   additionally be able to do the following:

     * Generate the 401 status code when authentication is required for
       the resource.
     * Parse and include the WWW-Authenticate header
     * Implement Basic Authentication and Digest Authentication

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Appendix E: Authors' Addresses

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027


   Anup Rao
   Netscape Communications Corp.
   501 E. Middlefield Road
   Mountain View, CA 94043


   Robert Lanphier
   1111 Third Avenue Suite 2900
   Seattle, WA 98101


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Appendix F: Acknowledgements

   This memo is based on the functionality of the original RTSP document
   submitted in October 96. It also borrows format and descriptions from

   This document has benefited greatly from the comments of all those
   participating in the MMUSIC-WG. In addition to those already
   mentioned, the following individuals have contributed to this

   Rahul Agarwal, Torsten Braun, Brent Browning, Bruce Butterfield,
   Steve Casner, Francisco Cortes, Kelly Djahandari, Martin Dunsmuir,
   Eric Fleischman, Jay Geagan, Andy Grignon, V. Guruprasad, Peter
   Haight, Mark Handley, Brad Hefta-Gaub, John K. Ho, Philipp Hoschka,
   Anne Jones, Anders Klemets, Ruth Lang, Stephanie Leif, Jonathan
   Lennox, Eduardo F. Llach, Rob McCool, David Oran, Maria Papadopouli,
   Sujal Patel, Ema Patki, Alagu Periyannan, Igor Plotnikov, Pinaki
   Shah, David Singer, Jeff Smith, Alexander Sokolsky, Dale Stammen, and
   John Francis Stracke.

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   1      Schulzrinne, H., "RTP profile for audio and video conferences
          with minimal control", RFC 1890, January 1996.

   2      Fielding, R., Gettys, J., Mogul, J., Nielsen, H., and T.
          Berners-Lee, "Hypertext transfer protocol - HTTP/1.1", RFC
          2068, January 1997.

   3      Yergeau, F., Nicol, G., Adams, G., and M. Duerst,
          "Internationalization of the hypertext markup language", RFC
          2070, January 1997.

   4      Bradner, S., "Key words for use in RFCs to indicate
          requirement levels", BCP 14, RFC 2119, March 1997.

   5      ISO/IEC, "Information technology - generic coding of moving
          pictures and associated audio information - part 6: extension
          for digital storage media and control," Draft International
          Standard ISO 13818-6, International Organization for
          Standardization ISO/IEC JTC1/SC29/WG11, Geneva, Switzerland,
          Nov. 1995.

   6      Handley, M., and V. Jacobson, "SDP: Session Description
          Protocol", RFC 2327, April 1998.

   7      Franks, J., Hallam-Baker, P., and J. Hostetler, "An extension to
          HTTP: digest access authentication", RFC 2069, January 1997.

   8      Postel, J., "User Datagram Protocol", STD 6, RFC 768, August

   9      Hinden, B. and C. Partridge, "Version 2 of the reliable data
          protocol (RDP)", RFC 1151, April 1990.

   10     Postel, J., "Transmission control protocol", STD 7, RFC 793,
          September 1981.

   11     H. Schulzrinne, "A comprehensive multimedia control
          architecture for the Internet," in Proc. International
          Workshop on Network and Operating System Support for Digital
          Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997.

   12     International Telecommunication Union, "Visual telephone
          systems and equipment for local area networks which provide a
          non-guaranteed quality of service," Recommendation H.323,
          Telecommunication Standardization Sector of ITU, Geneva,
          Switzerland, May 1996.

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   13     McMahon, P., "GSS-API authentication method for SOCKS version
          5", RFC 1961, June 1996.

   14     J. Miller, P. Resnick, and D. Singer, "Rating services and
          rating systems (and their machine readable descriptions),"
          Recommendation REC-PICS-services-961031, W3C (World Wide Web
          Consortium), Boston, Massachusetts, Oct. 1996.

   15     J. Miller, T. Krauskopf, P. Resnick, and W. Treese, "PICS
          label distribution label syntax and communication protocols,"
          Recommendation REC-PICS-labels-961031, W3C (World Wide Web
          Consortium), Boston, Massachusetts, Oct. 1996.

   16     Crocker, D. and P. Overell, "Augmented BNF for syntax
          specifications: ABNF", RFC 2234, November 1997.

   17     Braden, B., "Requirements for internet hosts - application and
          support", STD 3, RFC 1123, October 1989.

   18     Elz, R., "A compact representation of IPv6 addresses", RFC
          1924, April 1996.

   19     Berners-Lee, T., Masinter, L. and M. McCahill, "Uniform
          resource locators (URL)", RFC 1738, December 1994.

   20     Yergeau, F., "UTF-8, a transformation format of ISO 10646",
          RFC 2279, January 1998.

   22     Braden, B., "T/TCP - TCP extensions for transactions
          functional specification", RFC 1644, July 1994.

   22     W. R. Stevens, TCP/IP illustrated: the implementation, vol. 2.
          Reading, Massachusetts: Addison-Wesley, 1994.

   23     Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
          "RTP: a transport protocol for real-time applications", RFC
          1889, January 1996.

   24     Fielding, R., "Relative uniform resource locators", RFC 1808,
          June 1995.

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