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3620 resetting outgoing SIP call using G.729 immediatelly after 200 message

Hi everybody

My 3620 received a PSTN call and then forwards this to a softswitch. It proposes G.729 as part of the INVITE message:

a=rtpmap:18 G729/8000

The call gets connected and the same codec type is used in this message.

This is the SDP section in message 200:

t=0 0

m=audio 31374 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=rtpmap:101 telephone-event/8000

But then the router decides to disconnect and send a BYE message.

After debugging with "debug ccsip all" we find some events like:

Dec 12 16:46:06.289 pst: Codec (No Codec ) is not in preferred list

Dec 12 16:46:06.289 pst: Dynamic Payload :101 in SDP Body

Dec 12 16:46:06.293 pst: sipSPIDoAudioNegotiation: No matching voice codec found for m-line 1

We have no problems when using G.711 ulaw.

Any ideas why this fails. This is not a configuration issue; we have the same one in production. We think about a IOS bug or DSP failure.


Re: 3620 resetting outgoing SIP call using G.729 immediatelly af

Some codec compression techniques require more processing power than others. Codec complexity is broken into two categories named medium and high complexity.

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