Hi Nanang So of course the Conference bridge allows two streams, one ingoing and one outgoing: A ----in--------> B <---out------ What I mean by removing A from B's conference bridge is to end both streams status = pjsua_conf_disconnect(info.slot_id,0); status = pjsua_conf_disconnect(0,info.slot_id); On the RTP Level, i thought that this corresponds to a "manual" silence supression, both clients set the "s" bit in the rtp packets. Or how are those RTP Streams still maintained, since with every SIP Call a RTP stream is generated. So as you said this obviously should save me some bandwidth. My question also is where can I see those packets exchanged between those two clients, which loglevel should i use, or is there any other way like wireshark to caputre those packets? or maybe is the "dq function of pjsua already enough to measure this somehow? Cheers Thomas 2008/4/22 Nanang Izzuddin <nanang at pjsip.org>: > Hi Thomas, > > "Removing A from the B's conference bridge" is a bit ambiguous > sentence though :) > Assumed in the B side, the conference bridge only has 2 ports (sound > port is port 0 and stream port is port 1) then issuing command: > - "cd 0 1" will make B stop sending RTP to A (this will save traffic, > but NAT session may be gone) > - "cd 1 0" will not save traffic. > These scenarios don't seem to involve silence suppression. > > You can test the silence suppression by not talking or muting mic (to > make it safe from keyboard stroke sound :D), then you can check the > traffic by using command 'dq' periodically in pjsua. Please make sure > the VAD is enabled (just don't put --no-vad in the pjsua's param). > > > Regards, > nanang > > > On 22/04/2008, Thomas Plotkowiak <plotti at gmx.net> wrote: > > Scenario with Psjua: > > > > A initiates SIP call with B. > > > > B adds A to the conference bridge. > > B is able to hear A. > > --> In RTP protocoll audio is transmitted from A to B. > > > > What if i now remove A from the conference bridge and still have the > call. > > Does this lead to silence supresssion on B or A side? Do I save traffic > by > > doing this? > > > > How could I prove that I am saving traffic with this? Can I see it > somewhere > > happening in the logs? > > > > Any comments or info about this topic would be appreciated. > > > > > > Cheers > > Thomas > > > > > > RFC3389 on Silence Suppression: RTP allows discontinuous transmission > > (silence suppression) on any audio payload format. The receiver can > detect > > silence suppression on the first packet received after the silence by > > observing that the RTP timestamp is not contiguous with the end of the > > interval covered by the previous packet even though the RTP sequence > number > > has incremented only by one. The RTP marker bit is also normally set on > such > > a packet. > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip at lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080423/9bec87a1/attachment-0001.html
