El 18/06/14 13:44, Alex Villacís Lasso escribió:
I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the
same setup, the calling party 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 from asterisk)
v=0
o=root 835643920 835643920 IN IP4 201.234.196.171
s=Asterisk PBX 11.10.0
c=IN IP4 201.234.196.171
t=0 0
m=audio 12112 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
a=nortpproxy:yes
In contrast, all failing calls have SDP negotiation that looks 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 30534 RTP/AVP 0 18
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=sendrecv
(inside 200 OK response body from asterisk)
v=0
o=root 1719775538 1719775538 IN IP4 201.234.196.171
s=Asterisk PBX 11.10.0
c=IN IP4 201.234.196.171
t=0 0
m=audio 11678 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes
Apart from this, I see RTP packets being exchanged in both directions in both
scenarios.
From what I can see, the SIP provider sends packets that differ in that the
following four properties are present in the successful case and missing in the
failing case:
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
What can I deduce from this? Is there some configuration on my asterisk that
can be tweaked so the failing requests can be handled properly?
Is there additional information needed to make sense of this scenario?
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