[Live-devel] ssrc of rtp packets

[Simon Schampijer]

Hi,

as I described in the last mail - we now have a new structur of our
distributed virtual concert.
new implementation = new problems :)

n-Sender sends their stream (RTP packets) to one multicast address.
1-receiver receives the RTP packets from that multicast address.

->we wish now to check the ssrc of the packets to know where the audio
data should be written to (we got one ringbuffer for each source). In
our continuePlaying() (here we put the audio data of the rtp packets to
the ringbuffer) method of our from MediaSink derrived Sink class we do
the following to get the number of active Sources : 
rtpSource->receptionStatsDB().numActiveSourcesSinceLastReset();
(got from the playCommon example)
This works out fine - but we need also the ssrc to identify the packets.

We havn't found the right solution yet - so any idea is welcome

Thanks in advance 

Simon Schampijer