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