On Mon, Jul 28, 2008 at 10:19 AM, Klaus Darilion < klaus.mailinglists at pernau.at> wrote: > Hi Benny! > > I have modified the streamutil.c sample to send/receive from multicast > IP and subscribe to the multicast group. This works fine if there is > only one sender in the multicast group. If there are multiple senders to > the same mutlicast group, the audio played back by pjsip is broken. > > That's probably because you use one stream (i.e. one jitter buffer) for everyone. If the multiple senders are talking at the same time, I can imagine that the jitter buffer will get confused because of this. Even when talkers are talking in turns, the jitter buffer will also get confused if senders have different RTP timestamp reference (in other words, you need to make sure that all participants have the same RTP timestamp clock). The later problem probably could be alleviated with using the jitter buffer in fixed mode. To solve this issue I want to have multiple choices, e.g: > - conference all the incoming streams > - use only the first stream (and some timeout to change to another > stream if the current stream stops) > - manual selection of the stream (SSRC) > > Can you give me some hints how to implement this? > > If you want to create one stream for each participant, you'd need to implement some sort of wrapper to the media transport, which will check the SSRC in incoming RTP and distribute it to the appropriate stream instance (or create the stream if it doesn't exist). This kind of adapter is explained a bit in http://trac.pjsip.org/repos/wiki/SRTP#pjmedia_transport_srtpImplementation -benny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080728/97866c72/attachment.html
