RTP Synchonization Src / Contributing Src

[kcarpenter]

SSRC: We do use SSRC in WAVE (standard RTP of course). SSRCs are calculated for each new distinct talk spurt which helps not only with clearing buffers and reducing latency on receiving endpoints (WAVE or otherwise). On client endpoints such as Desktop and Dispatch, a new SSRC is calculated each time the PTT button is pushed. On the Media Server, a new SSRC is calculated after 5 seconds (if I recall correctly) of transmission inactivity.

CSRC: We do not transmit RTCP on channel communications (only on IP telephony calls). In either case, we do not put together a CSRC list as this approach does not scale in WAVE-type environments. Rather, we use WAVE.s control channel to indicate status and presence of talkers.

Please note that SSRC of a transmitter is NOT preserved if the audio from that transmitter is relayed through a Media Server or Dispatch endpoint. Instead, the SSRC used is that of the Media Server and/or Dispatch endpoint for that unique transmission stream.

[jeremycorkery]

I am using wave on a number of different sites, all connected together over a mesh satellite system. When testing in the lab I have the satellite modems connected together by the IF (coax) and WAVE will work.
WAVE will break between sites when I do either (or both) of the following:
-Introduce crypto hardware, and / or
-Put the transmission over the RF Satellite path (to space and back).
b/w is usually about 1 to 2 Mbps in most scenarios.
I believe the problem relates to latency. The satellite is a TDMA system, although I nail up quite a bit of bandwidth between sites so the b/w is there.
Does anyone have any experience using WAVE over satellite links, with delay... any tuning I could try. This is kind of urgent to get this going, and I hope it is possible.
Further information: I have a Supernode Multicast topologies in each site, each with a media server... and unicast links between them.