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2. RTP and RTCP Packet Forms and Protocol Behavior

  1. RTP and RTCP Packet Forms and Protocol Behavior

    The section "RTP Profiles and Payload Format Specifications" of RFC 3550 enumerates a number of items that can be specified or modified in a profile. This section addresses these items. Generally, this profile follows the default and/or recommended aspects of the RTP specification.

    RTP data header: The standard format of the fixed RTP data header is used (one marker bit).

    Payload types: Static payload types are defined in Section 6.

    RTP data header additions: No additional fixed fields are appended to the RTP data header.

    RTP data header extensions: No RTP header extensions are defined, but applications operating under this profile MAY use such extensions. Thus, applications SHOULD NOT assume that the RTP header X bit is always zero and SHOULD be prepared to ignore the header extension. If a header extension is defined in the future, that definition MUST specify the contents of the first 16 bits in such a way that multiple different extensions can be identified.

    RTCP packet types: No additional RTCP packet types are defined by this profile specification.

    RTCP report interval: The suggested constants are to be used for the RTCP report interval calculation. Sessions operating under this profile MAY specify a separate parameter for the RTCP traffic bandwidth rather than using the default fraction of the session bandwidth. The RTCP traffic bandwidth MAY be divided into two separate session parameters for those participants which are active data senders and those which are not. Following the recommendation in the RTP specification [1] that 1/4 of the RTCP bandwidth be dedicated to data senders, the RECOMMENDED default values for these two parameters would be 1.25% and 3.75%, respectively. For a particular session, the RTCP bandwidth for non-data-senders MAY be set to zero when operating on unidirectional links or for sessions that don't require feedback on the quality of reception. The RTCP bandwidth for data senders SHOULD be kept non-zero so that sender reports can still be sent for inter-media synchronization and to identify the source by CNAME. The means by which the one or two session parameters for RTCP bandwidth are specified is beyond the scope of this memo.

    SR/RR extension: No extension section is defined for the RTCP SR or RR packet.

    SDES use: Applications MAY use any of the SDES items described in the RTP specification. While CNAME information MUST be sent every reporting interval, other items SHOULD only be sent every third reporting interval, with NAME sent seven out of eight times within that slot and the remaining SDES items cyclically taking up the eighth slot, as defined in Section 6.2.2 of the RTP specification. In other words, NAME is sent in RTCP packets 1, 4, 7, 10, 13, 16, 19, while, say, EMAIL is used in RTCP packet 22.

    Security: The RTP default security services are also the default under this profile.

    String-to-key mapping: No mapping is specified by this profile.

    Congestion: RTP and this profile may be used in the context of enhanced network service, for example, through Integrated Services (RFC 1633) [4] or Differentiated Services (RFC 2475) [5], or they may be used with best effort service.

    If enhanced service is being used, RTP receivers SHOULD monitor packet loss to ensure that the service that was requested is actually being delivered. If it is not, then they SHOULD assume that they are receiving best-effort service and behave accordingly.

    If best-effort service is being used, RTP receivers SHOULD monitor packet loss to ensure that the packet loss rate is within acceptable parameters. Packet loss is considered acceptable if a TCP flow across the same network path and experiencing the same network conditions would achieve an average throughput, measured on a reasonable timescale, that is not less than the RTP flow is achieving. This condition can be satisfied by implementing congestion control mechanisms to adapt the transmission rate (or the number of layers subscribed for a layered multicast session), or by arranging for a receiver to leave the session if the loss rate is unacceptably high.

    The comparison to TCP cannot be specified exactly, but is intended as an "order-of-magnitude" comparison in timescale and throughput. The timescale on which TCP throughput is measured is the round- trip time of the connection. In essence, this requirement states that it is not acceptable to deploy an application (using RTP or any other transport protocol) on the best-effort Internet which consumes bandwidth arbitrarily and does not compete fairly with TCP within an order of magnitude.

    Underlying protocol: The profile specifies the use of RTP over unicast and multicast UDP as well as TCP. (This does not preclude the use of these definitions when RTP is carried by other lower- layer protocols.)

    Transport mapping: The standard mapping of RTP and RTCP to transport-level addresses is used.

    Encapsulation: This profile leaves to applications the specification of RTP encapsulation in protocols other than UDP.

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