Hello,
On 07/03/2017 13:10, Grant Bagdasarian wrote: > > Hi, > > > > One of our customers is using a SEMS box to place two outbound calls > using our sip trunk. > > Once the first call is connected a second call is placed and when the > second call answers their server sends a re-invite to switch audio > ports so the rtp traffic doesn’t flow through their server anymore but > is routed inside our platform. > > Basically, they just switch SDP’s of both calls. > > It seems like a random issue, and is not really reproducible, except > for placing multiple calls and sometimes both parties can hear each > other, other times they can’t, because rtpengine fails (I think) to > update the endpoint and keeps sending rtp back to their server for one > of the call legs. > > > > We tried to reproduce the case using a freeswitch box and it worked > every time. After the reinvite, the rtp remained within our platform. > > The signaling in both cases still goes through the freeswitch or sems > for call control. > > > > Does anyone have experience with this case? Or seen the issue before > where rtpengine keeps sending rtp to the original endpoint? > Have your checked to see if the sip messages are received/processed in the expected order? In some very rare situations, it happened that the re-invite was sent very fast by callee after just sending the 200ok, so that the re-invite arrived to the proxy/rtprelay before the 200ok, so at the end the sdp from 200ok was taken as the last relevant one for the peer. I put there rtprelay, because I faced this issue where I had rtpproxy, but maybe the issue is exposed by the rtpengine as well. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Mar 6-8 (Europe) and Mar 20-22 (USA) - www.asipto.com Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
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