Hello,
In the company I work for, we are running (really old) CUCM 11.5.1 / IP Phone 7841 and using SIP TRUNK as external gateway.
Some people in my comany reports blank calls (no audio in either direction) for certain numbers while other ones are working properly.
As I'm not TOIP guy and have minimal knowledge in SIP protocol, I spend a lot of time monitoring TOIP and looking for wireshark captures trying to figure out what change between working calls and non working ones.
I can note that we get 2 SIP INVITE messages for calls without audio and only one for call with audio working properly.
For both scenarios (working / non working) CUCM send a first SIP INVITE that contains IP Phone in Owner Address inside SIP Message Body.
During the non working scenario, CUCM generate a second SIP invite that contains CUCM IP address as Owner Address.
In the working scenario, RTP are established between IP Phone and SIP gateway which (as far as I'm aware) is expected?.
In the non working scenario, RTP are between CUCM and SIP gateway and both party can't hear each other.
I try to figure why CUCM generate the second SIP INVITE message (as I suppose that this is related to our issue without being sure of that) and I notice that in the non working scenario, remote party send to CUCM a SIP 200 SDP message containing two audio codecs inside "Media Attribute".
I read in this post that a SIP INVITE could be generate if I get a 200 OK response with multiple codec?
Finally, after digging a bit more, I was looking for SIP Profile Configuration associated to my SIP Gateway containing these informations:
Working call flow:
Non working call flow:
SDP message received from remote party:
v=0
o=- 3512106348 798068497 IN IP4 SIP_GW_IP
s=-
c=IN IP4 SIP_GW_IP
t=0 0
m=audio 6886 RTP/AVP 8 18 101
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
a=ptime:20
First SIP invite message Body during non working scenario:
v=0
o=CiscoSystemsCCM-SIP 83589867 1 IN IP4 IP_PHONE_IP
s=SIP Call
c=IN IP4 IP_PHONE_IP
b=AS:64
t=0 0
m=audio 32484 RTP/AVP 8 0 18 101
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Second SIP invite message Body during non working scenario (IP is the CUCM IP):
v=0
o=CiscoSystemsCCM-SIP 83589867 2 IN IP4 CUCM_IP
s=SIP Call
c=IN IP4 CUCM_IP
b=AS:64
t=0 0
m=audio 63970 RTP/AVP 8 101
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Am I close to find the root cause of the issue with this audio codec and is there anything to change on the CUCM side to make it works again?
Waiting for your precious help guys !
Thanks a lot !
