05-10-2017 11:18 AM - edited 03-17-2019 10:17 AM
Hi Community!
Hoping you can help me with this one.
I've had reports from some of my users that they get no audio on random incoming calls. When they answer the call, the recipient cannot hear anything, and the caller is disconnected.
I took some debugs and noticed in the failing calls that the SIP invite and the c line of the SDP have IP addresses that do not match. Debugs of successful calls have IP addresses that both match in SIP invite and SDP.
Here is the output I gathered (some values are masked to protect the innocent):
Failed call
008948: May 10 10:36:57.048: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:~~~~~~5056@XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 12.194.141.181:5060;branch=z9hG4bKhak5ks200o6858hjj6m0.1
From: "MYCALLER" <sip:2~~~~~~787@12.194.141.181:5060>;tag=7638500593592712_c1b07.2.1.1487225703669.0_12746247_25332790
To: <sip:~~~~~~5056@XXX.XXX.XXX.XXX>
Call-ID: 09764958280100544@12.114.48.16
CSeq: 2 INVITE
P-Asserted-Identity: "MYCALLER" <sip:2~~~~~~787@12.194.141.181:5060>
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Max-Forwards: 65
Contact: <sip:12.194.141.181:5060;transport=udp>
Content-Length: 265
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 31400 8983 IN IP4 12.194.141.190
s=SIP Media Capabilities
c=IN IP4 12.194.141.190
t=0 0
m=audio 25184 RTP/AVP 18 0 100
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:30
Successful call
009173: May 10 10:58:18.552: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:~~~~~~5056@XXX.XXX.XXX.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 12.194.141.181:5060;branch=z9hG4bKnb35493078q64065nlp0.1
From: "MYCALLER" <sip:2~~~~~~592@12.194.141.181:5060>;tag=09002470632800741_c2b04.1.1.1485844275705.0_15931885_31660235
To: <sip:~~~~~~5056@XXX.XXX.XXX.XXX>
Call-ID: 5288979150146234@12.114.232.34
CSeq: 2 INVITE
P-Asserted-Identity: "MYCALLER" <sip:2~~~~~~592@12.194.141.181:5060>
Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Max-Forwards: 65
Contact: <sip:12.194.141.181:5060;transport=udp>
Content-Length: 265
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 31304 8304 IN IP4 12.194.141.181
s=SIP Media Capabilities
c=IN IP4 12.194.141.181
t=0 0
m=audio 27116 RTP/AVP 18 0 100
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=sendrecv
a=maxptime:30
Is this behavior normal and expected?
Regards,
Jason
05-10-2017 10:46 PM
Hi Jason,
This is normal. This means that your provider is having two sonus controllers which is normal to load balance in the calls in provider environment.
Now its clear that when signaling and media goes through same controller, calls are successful. When signaling and media are through two different controllers, the call is failing.
This sounds to be a problem with one of the controllers but if you share full debug for failed call, I might be able to confirm that.
05-11-2017 06:44 AM
05-11-2017 09:04 AM
I was able to resolve this by adding a dial peer for the extra IP address.
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