SIP: Internal Error (200, codec mismatch)

[DJ2018]

Hi Guys,

CUBE is a 2811 running on 151-4.M7

I am getting rusty. need some help to identify the problem.

Call Flow: IP Phone -> CME -> CUBE -> ITSP

This was a working call flow for around a month and stopped working.

I turned on 

#debug ccsip messages

#debug voip ccapi inout

From basic troubleshooting done i could see that my CUBE is initiating the BYE message towards the provider.

right before sending the Bye i see below error. 

000451: *Nov 10 09:28:24.055: %VOICE_IEC-3-GW: SIP: Internal Error (200, codec mismatch): 

I do not understand why i get this error.

This is the SDP sent by CME to CUBE in Invite.

c=IN IP4 192.168.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

This is the SDP sent by CUBE to provider in Invite.

m=audio 19578 RTP/AVP 18 101
c=IN IP4 10.1.1.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

This is the OK message received from Provider:

a=rtpmap:18 G729/8000/1 <-- not sure what is that extra /1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:80
a=sendrecv

Then call disconnect with codec mismatch.

debug attached.

[Mohammed al Baqari · Accepted solution]

Hi,

Try to turn annex-b on cme and cube and see if it work. voice service voip
> sip > g729 anne

[mezdeke11]

thank you a lot, i solve my problem now . 

IN TERMINAL MONITOR 
%VOICE_IEC-3-GW SIP INTERNAL ERROR (200 CODEC MISMATCH).

With adding the commands :  voice service voip> sip > g729 anne

all is ok now.

Thank You

[Ayodeji Okanlawon · Accepted solution]

Hi,

The extra 1 indicates the encoding parameter used. It implies the number of audio channels used for transmission. The reason why you dont see this in most SIP messages is that it is an optional parameter if a single audio channel is used.

The issue you are having is indeed related to codec mismatch. According to the RFC, it states that there is no annexb attribute present in SDP, the default value assumed will be "YES" Hence in your case, since your CUBE is configured not to support annexb but your provider wants to do annexb, your call is failing.

You can either enable annexb using voice-class coded or use Mohammed's suggestion

[DJ2018]

This does not make any sense. 

Call worked but SDP looks the same

Invite from CME-> CUBE [ i will verify the CME configuration ]

a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:40

Invite from CUBE -> Provider

a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

200 Ok from Provider to CUBE  <== (No annexb) 

a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:80
a=sendrecv

200 OK from CUBE to CME

t=0 0
m=audio 18840 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Does this mean that Provider is talking to CUBE in Annexb=Yes and

CUBE to CME in Annexb=No ?

Does this consume transcoding?