Re: [asterisk-users] RTP timestamps
Hi, One more interesting fact, i see correlation with DTMF features, after i disabled corresponding options on dial commands (like htw) the timestamps on rtp are constantly growing and no more one way audio problems after call transfer, hold, parking etc. So it seems there is a bug related to rtp, rfc2833 and timestamp calculation. Or maybe some misconfigured features ? Has anyone seen this behaviour before ? Greetings, Liivo 27.10.2009 16:53, Liivo Vöörmann kirjutas: Hi Alex, Yes, it's almost the same, except the fact that in my case timestamps sometimes decrease drastically. In internal network I have Snom 3xx phones with upgraded firmware, internal leg has no issues, i captured both legs and phones-asterisk part is ok, the other part, asterisk-provider has these issues which are mentioned above. Greetings, Liivo 27.10.2009 15:28, Alex Balashov kirjutas: Liivo, I wonder if you are dealing with this general class of issues: https://issues.asterisk.org/view.php?id=11491 -- Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RTP timestamps
Hi All, Could somebody explain me how the timestamps are computed in asterisk while bridging two sip channels ? I've got situation with my provider, who changed some things in config and added some codecs (that much i know) and after that we got one way audio issues. It seems that the problem is with RTP timestamps. Within one outgoing stream the RTP timestamps are growing, as it should be, but sometimes while the asterisk plays MOH (or somebody transfers call to another extension) the timestamps on RTP packets will fall to past. Providers media gateway dosn't like that. The marker bit is correctly set but it seems like that dosn't change anything. Sequences and SSRC-s are Ok, no packet loss. Has anyone seen something like this before and knows what is the cause and how to fix this? I've tried many changes in config and upgraded to 1.6.1 but it didnt change anything, currently running asterisk 1.4.26.1 on 64 bit intel platform with opensuse. Here is the tcpdump view from wireshark, xxx is providers ip and yyy is asterisk: 6218207.717454xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54364, Time=1987711680 6219207.717481yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22826, Time=2202453496 6220207.737442xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54365, Time=1987711840 6221207.757430xxx.xxx.xxx.xxxyyy.yyy.yyy.yyyRTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54366, Time=1987712000 6222207.759283yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22827, Time=736089280, Mark 6223207.765349yyy.yyy.yyy.yyyxxx.xxx.xxx.xxxRTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22828, Time=736089440 Help! Greetings, Liivo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP timestamps
Hi Alex, Yes, it's almost the same, except the fact that in my case timestamps sometimes decrease drastically. In internal network I have Snom 3xx phones with upgraded firmware, internal leg has no issues, i captured both legs and phones-asterisk part is ok, the other part, asterisk-provider has these issues which are mentioned above. Greetings, Liivo 27.10.2009 15:28, Alex Balashov kirjutas: Liivo, I wonder if you are dealing with this general class of issues: https://issues.asterisk.org/view.php?id=11491 -- Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
