We have a customer reporting poor quality calls when they come to us via
a particular provider. The SIP traces look perfectly normal both on the
ingress to us and egress to another telco. No additional sip messages
after the call has been answered until the BYE is received. However in
the asterisk logs I am getting this :-
2014-02-05 13:45:03 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
2014-02-05 13:45:04 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
2014-02-05 13:45:05 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
2014-02-05 13:45:06 | C-00108c80] rtp_engine.c: -- Locally bridging
SIP/vmpubopensips1-001dfd43 and SIP/magrathea2-001dfd44
Any idea what could be causing this?
I am running asterisk 11.2-cert2.
I am going to get call redirected via our test box and turn on full
verbosity in the logs and capture a full tcpdump but any ideas would be
welcome.
Thanks
Gareth
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