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chance4minutes
2014 Jun 18
1
Making sense of SDP for debugging of missing audio in SIP trunk
...is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to have started. When debugging with tcpdump, I have seen that all the successful calls have SDP negotiation that
look like this:
(inside INVITE request body from SIP carrier)
v=0
o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46
s=sip call
c=IN IP4 38.126.208.46
t=0 0
m=audio 30552 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
(inside 200 OK response body fro...