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

ewaizel
Level 1
Level 1

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.

1 Reply 1

ebreniz
Level 6
Level 6

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

http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml