Hi Maxim,
Hard to do because it's in production. I have a serious finding since yesterday on how this happened. My understanding is that the function "ts_less" returns FALSE into "rtp_resizer.c" because the timestamp between the two packets is > (1 << 31) [for example: 3740425320]. That's result in a drop of any following packets as I can see it into a capture. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Maxim Sobolev Envoyé : vendredi 6 mars 2015 07:44 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] RTPProxy issue? Hi Igor, that's bit strange, since the rtpproxy is not checking any of the rtp flags including marker bit. It would help if you can post a tcpdump capture of the streams in question along with the log output of the rtpproxy running at the "dbug" level. Thanks! On Mar 5, 2015 5:54 AM, "Igor Potjevlesch" <igor.potjevle...@gmail.com <mailto:igor.potjevle...@gmail.com> > wrote: I reviewed again a call trace and I can be more precise: a RTP packet comes with a new SSRC and the Marker bit set to "True". This packet is properly forwarded. Then, just after this packet, another RTP packet containing a new SSRC with the huge timestamp and the Marker bit set to "True" is coming from the UA. The RTPProxy stops forward since this packet. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com <mailto:igor.potjevle...@gmail.com> ] Envoyé : jeudi 5 mars 2015 11:34 À : mico...@gmail.com <mailto:mico...@gmail.com> ; 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] RTPProxy issue? Hello, Thank you. Just to let you know, the RTPProxy is running in bridging mode. Regards, Igor. De : sr-users [mailto:sr-users-boun...@lists.sip-router.org] De la part de Daniel-Constantin Mierla Envoyé : jeudi 5 mars 2015 09:33 À : Kamailio (SER) - Users Mailing List Objet : Re: [SR-Users] RTPProxy issue? Hello, maybe Maxim (cc-ed) will be able to provide more insights. Cheers, DAniel On 04/03/15 16:59, Igor Potjevlesch wrote: Hello, I discovered an issue related to the handling of "timestamp" and/or "Marker bit" with rtpproxy (I use the latest Extension 20081224). The call-flow is the following: one UA places a call to A and put this call on hold. Then, the same UA call another number B. Individual streams are ok. When the UA tries to transfer A with B, the RTPProxy receive a RTP packet with a huge timestamp and the Marker bit set to "True". Just after this RTP packet, RTPProxy stop forward the RTP packets from A to B. B to C is still working. Anyone have an idea? Regards, Igor. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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