The Internet transport protocol for real-time flows is RTP [#!rtp!#]. This provides a standard format packet header which gives media specific timestamp data, as well as payload format information and sequence numbering amongst other things. RTP is normally carried using UDP. It does not provide or require any connection setup, nor does it provide any enhanced reliability over UDP. For RTP to provide a useful media flow, there must be sufficient capacity in the relevant traffic class to accommodate the traffic. How this capacity is ensured is independent of RTP.
RTP media timestamps units are flow specific - they are in units that are appropriate to the media flow. For example, 8KHz sampled PCM encoded audio has a timestamp clock rate of 8KHz. This means that inter-flow synchronisation is not possible from the RTP timestamps alone.
Every original RTP source is identified by a source identifier, and this source id is carried in every packet. RTP allows flows from several sources to be mixed in gateways to provide a single resulting flow. When this happens, each mixed packet contains the source id's of all the contributing sources.
Each RTP flow is supplemented by Real-Time Control Protocol (RTCP) packets. There are a number of different RTCP packet types. RTCP packets provide the relationship between the real time clock at a sender and the RTP media timestamps, and provide textual information to identify a sender in a session from the source id.
This is described in more detail in chapter five.