rfc1889.txt

RFC关于RTP实时传输协议的详细规范。 原理并不复杂

   The audio conferencing application used by each conference
   participant sends audio data in small chunks of, say, 20 ms duration.
   Each chunk of audio data is preceded by an RTP header; RTP header and
   data are in turn contained in a UDP packet. The RTP header indicates
   what type of audio encoding (such as PCM, ADPCM or LPC) is contained
   in each packet so that senders can change the encoding during a
   conference, for example, to accommodate a new participant that is
   connected through a low-bandwidth link or react to indications of
   network congestion.

   The Internet, like other packet networks, occasionally loses and
   reorders packets and delays them by variable amounts of time. To cope
   with these impairments, the RTP header contains timing information
   and a sequence number that allow the receivers to reconstruct the
   timing produced by the source, so that in this example, chunks of
   audio are contiguously played out the speaker every 20 ms. This
   timing reconstruction is performed separately for each source of RTP
   packets in the conference. The sequence number can also be used by
   the receiver to estimate how many packets are being lost.

   Since members of the working group join and leave during the
   conference, it is useful to know who is participating at any moment
   and how well they are receiving the audio data. For that purpose,



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RFC 1889                          RTP                       January 1996


   each instance of the audio application in the conference periodically
   multicasts a reception report plus the name of its user on the RTCP
   (control) port. The reception report indicates how well the current
   speaker is being received and may be used to control adaptive
   encodings. In addition to the user name, other identifying
   information may also be included subject to control bandwidth limits.
   A site sends the RTCP BYE packet (Section 6.5) when it leaves the
   conference.

2.2 Audio and Video Conference

   If both audio and video media are used in a conference, they are
   transmitted as separate RTP sessions RTCP packets are transmitted for
   each medium using two different UDP port pairs and/or multicast
   addresses. There is no direct coupling at the RTP level between the
   audio and video sessions, except that a user participating in both
   sessions should use the same distinguished (canonical) name in the
   RTCP packets for both so that the sessions can be associated.

   One motivation for this separation is to allow some participants in
   the conference to receive only one medium if they choose. Further
   explanation is given in Section 5.2. Despite the separation,
   synchronized playback of a source's audio and video can be achieved
   using timing information carried in the RTCP packets for both
   sessions.

2.3 Mixers and Translators

   So far, we have assumed that all sites want to receive  media data in
   the same format. However, this may not always be appropriate.
   Consider the case where participants in one area are connected
   through a low-speed link to the majority of the conference
   participants who enjoy high-speed network access. Instead of forcing
   everyone to use a lower-bandwidth, reduced-quality audio encoding, an
   RTP-level relay called a mixer may be placed near the low-bandwidth
   area. This mixer resynchronizes incoming audio packets to reconstruct
   the constant 20 ms spacing generated by the sender, mixes these
   reconstructed audio streams into a single stream, translates the
   audio encoding to a lower-bandwidth one and forwards the lower-
   bandwidth packet stream across the low-speed link. These packets
   might be unicast to a single recipient or multicast on a different
   address to multiple recipients. The RTP header includes a means for
   mixers to identify the sources that contributed to a mixed packet so
   that correct talker indication can be provided at the receivers.

   Some of the intended participants in the audio conference may be
   connected with high bandwidth links but might not be directly
   reachable via IP multicast. For example, they might be behind an



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RFC 1889                          RTP                       January 1996


   application-level firewall that will not let any IP packets pass. For
   these sites, mixing may not be necessary, in which case another type
   of RTP-level relay called a translator may be used. Two translators
   are installed, one on either side of the firewall, with the outside
   one funneling all multicast packets received through a secure
   connection to the translator inside the firewall. The translator
   inside the firewall sends them again as multicast packets to a
   multicast group restricted to the site's internal network.

   Mixers and translators may be designed for a variety of purposes. An
   example is a video mixer that scales the images of individual people
   in separate video streams and composites them into one video stream
   to simulate a group scene. Other examples of translation include the
   connection of a group of hosts speaking only IP/UDP to a group of
   hosts that understand only ST-II, or the packet-by-packet encoding
   translation of video streams from individual sources without
   resynchronization or mixing. Details of the operation of mixers and
   translators are given in Section 7.

3.  Definitions

   RTP payload: The data transported by RTP in a packet, for example
        audio samples or compressed video data. The payload format and
        interpretation are beyond the scope of this document.

   RTP packet: A data packet consisting of the fixed RTP header, a
        possibly empty list of contributing sources (see below), and the
        payload data. Some underlying protocols may require an
        encapsulation of the RTP packet to be defined. Typically one
        packet of the underlying protocol contains a single RTP packet,
        but several RTP packets may be contained if permitted by the
        encapsulation method (see Section 10).

   RTCP packet: A control packet consisting of a fixed header part
        similar to that of RTP data packets, followed by structured
        elements that vary depending upon the RTCP packet type. The
        formats are defined in Section 6. Typically, multiple RTCP
        packets are sent together as a compound RTCP packet in a single
        packet of the underlying protocol; this is enabled by the length
        field in the fixed header of each RTCP packet.

   Port: The "abstraction that transport protocols use to distinguish
        among multiple destinations within a given host computer. TCP/IP
        protocols identify ports using small positive integers." [3] The
        transport selectors (TSEL) used by the OSI transport layer are
        equivalent to ports.  RTP depends upon the lower-layer protocol
        to provide some mechanism such as ports to multiplex the RTP and
        RTCP packets of a session.



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RFC 1889                          RTP                       January 1996


   Transport address: The combination of a network address and port that
        identifies a transport-level endpoint, for example an IP address
        and a UDP port. Packets are transmitted from a source transport
        address to a destination transport address.

   RTP session: The association among a set of participants
        communicating with RTP. For each participant, the session is
        defined by a particular pair of destination transport addresses
        (one network address plus a port pair for RTP and RTCP). The
        destination transport address pair may be common for all
        participants, as in the case of IP multicast, or may be
        different for each, as in the case of individual unicast network
        addresses plus a common port pair.  In a multimedia session,
        each medium is carried in a separate RTP session with its own
        RTCP packets. The multiple RTP sessions are distinguished by
        different port number pairs and/or different multicast
        addresses.

   Synchronization source (SSRC): The source of a stream of RTP packets,
        identified by a 32-bit numeric SSRC identifier carried in the
        RTP header so as not to be dependent upon the network address.
        All packets from a synchronization source form part of the same
        timing and sequence number space, so a receiver groups packets
        by synchronization source for playback. Examples of
        synchronization sources include the sender of a stream of
        packets derived from a signal source such as a microphone or a
        camera, or an RTP mixer (see below). A synchronization source
        may change its data format, e.g., audio encoding, over time. The
        SSRC identifier is a randomly chosen value meant to be globally
        unique within a particular RTP session (see Section 8). A
        participant need not use the same SSRC identifier for all the
        RTP sessions in a multimedia session; the binding of the SSRC
        identifiers is provided through RTCP (see Section 6.4.1).  If a
        participant generates multiple streams in one RTP session, for
        example from separate video cameras, each must be identified as
        a different SSRC.

   Contributing source (CSRC): A source of a stream of RTP packets that
        has contributed to the combined stream produced by an RTP mixer
        (see below). The mixer inserts a list of the SSRC identifiers of
        the sources that contributed to the generation of a particular
        packet into the RTP header of that packet. This list is called
        the CSRC list. An example application is audio conferencing
        where a mixer indicates all the talkers whose speech was
        combined to produce the outgoing packet, allowing the receiver
        to indicate the current talker, even though all the audio
        packets contain the same SSRC identifier (that of the mixer).




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RFC 1889                          RTP                       January 1996


   End system: An application that generates the content to be sent in
        RTP packets and/or consumes the content of received RTP packets.
        An end system can act as one or more synchronization sources in
        a particular RTP session, but typically only one.

   Mixer: An intermediate system that receives RTP packets from one or
        more sources, possibly changes the data format, combines the
        packets in some manner and then forwards a new RTP packet. Since
        the timing among multiple input sources will not generally be
        synchronized, the mixer will make timing adjustments among the
        streams and generate its own timing for the combined stream.
        Thus, all data packets originating from a mixer will be
        identified as having the mixer as their synchronization source.

   Translator: An intermediate system that forwards RTP packets with
        their synchronization source identifier intact. Examples of
        translators include devices that convert encodings without
        mixing, replicators from multicast to unicast, and application-
        level filters in firewalls.

   Monitor: An application that receives RTCP packets sent by
        participants in an RTP session, in particular the reception
        reports, and estimates the current quality of service for
        distribution monitoring, fault diagnosis and long-term
        statistics. The monitor function is likely to be built into the
        application(s) participating in the session, but may also be a
        separate application that does not otherwise participate and
        does not send or receive the RTP data packets. These are called
        third party monitors.

   Non-RTP means: Protocols and mechanisms that may be needed in
        addition to RTP to provide a usable service. In particular, for
        multimedia conferences, a conference control application may
        distribute multicast addresses and keys for encryption,
        negotiate the encryption algorithm to be used, and define
        dynamic mappings between RTP payload type values and the payload
        formats they represent for formats that do not have a predefined
        payload type value. For simple applications, electronic mail or
        a conference database may also be used. The specification of
        such protocols and mechanisms is outside the scope of this
        document.

4.  Byte Order, Alignment, and Time Format

   All integer fields are carried in network byte order, that is, most
   significant byte (octet) first. This byte order is commonly known as
   big-endian. The transmission order is described in detail in [4].
   Unless otherwise noted, numeric constants are in decimal (base 10).



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