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:
6218 207.717454 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy RTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54364, Time=1987711680 6219 207.717481 yyy.yyy.yyy.yyy xxx.xxx.xxx.xxx RTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22826, Time=2202453496 6220 207.737442 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy RTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54365, Time=1987711840 6221 207.757430 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy RTP PT=ITU-T G.711 PCMA, SSRC=0x85048346, Seq=54366, Time=1987712000 6222 207.759283 yyy.yyy.yyy.yyy xxx.xxx.xxx.xxx RTP PT=ITU-T G.711 PCMA, SSRC=0x35276954, Seq=22827, Time=736089280, Mark 6223 207.765349 yyy.yyy.yyy.yyy xxx.xxx.xxx.xxx RTP 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