Cisco 8861 - Codec Problem with Router

[AlexWurzelsepp]

Hi Experts,

we experience a problem making calls through a sip router. Incoming calls are working fine, outgoing calls don't work.

On the router and the phone logs I find similar error messages, indicating a problem, negotiating the right codec.

Here is a snipped of the log:

CSeq: 102 INVITE
Warning: 399 0.0.0.0 "no fitting codec"
User-Agent: FRITZ!OS
Content-Type: application/sdp
Content-Length: 421

v=0
o=user 1721170 1721170 IN IP4 192.168.10.208
s=call
c=IN IP4 192.168.10.208
t=0 0
m=audio 7082 RTP/AVP 9 8 0 2 102 100 99 101 97 120 121
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:120 PCMA/16000
a=rtpmap:121 PCMU/16000
a=rtcp:7083

----------

I tried all available settings on the phones web settings - without a change.

other ip phones work without problems on the same router.

Any idea?

Best rgds,
Alex

[BalajiSivaraj49175]

CallManager allocates MTP for call flow, and selects g711u codec, as the SIP provider claims to support that codec.

Please work with your Telco to not send g711u as a supported codec to CallManager, so that this issue doesn't occur.

[Nithin Eluvathingal]

There is no such setting on phone web page you can change for Codec. can you explain a little more about your setup. What call control platform you use ?

Is this a CME ? If yes would you be able to share the config.

2.12.0.0

[AlexWurzelsepp]

There are severaloptions to change the codec - but not the ones, the router would like to see first (with 16k bandwidh).

the setup is easy: the phone, a router (a sip router made by aAVM) and the sip proider.

the router serves 10 phone numbers registered at one provider - and on the other side several ip phones that work well - except the cisco.

if you need specific info, let me know pls.

[Ibrahim AlQumairi]

There is a mismatch in codec mapping, so to be sure which codec you need to use in outgoing calls, check the incoming calls SIP invite for which codec they use, and after that try to configure that codec in the router voice class codec.

Also don't forget to activate the MTP in the CUCM trunk "Media Termination Point Required"