[Live-devel] RTP Audio without sip

[Ross Finlayson]

RTP and RTCP are different protocols, but which work together.  RTP packets deliver the media data.  RTCP is used (in the reverse direction) to monitor/report the quality of the RTP stream (including reporting packet loss rates).  Also, if you have separate RTP streams for audio and video, then RTCP (in the forward direction) is used to provide time synchronization between them.

For more information about the purpose of RTCP, see
	https://www.rfc-editor.org/rfc/rfc3550.html#section-6
and
	https://en.wikipedia.org/wiki/RTP_Control_Protocol

Although you should implement RTCP as well as RTP, RTCP can probably be omitted if - as in your case - you have just a single media stream (in your case, just audio).  But implementing RTCP is very easy; just call
	RTCPInstance::createNew()
as we do in the demo applications (e.g., “testMP3Receiver” and “testAMRAudioStreamer”), and RTCP packets will be sent/received automatically by our code.


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/