SIP Trunk Cisco <-> Asterisk with ILBC Codec

[evandrojnabor]

I'm having problems with this situation

I've configured asterisk to permit codecs preferece 1 - ilbc and 2 - alaw

When I make a call from Cisco FXS port -> Asterisk extension, everything works fine, the codec chosen is ILBC and the voice quality is ok.

Calls from Asterisk extension -> Cisco FXS extension always use codec alaw and the voice quality is not good.

I'm using snmp to monitor the interfaces to know what codec the devices are using.

This is my Cisco router configuration

http://pastebin.com/pwMTxGEv

This is the log from debug ccsip all

http://pastebin.com/d9d9hUwr

Thanks

[Ayodeji Okanlawon · Accepted solution]

From the logs, here is what I see...

26.m=audio 16568 RTP/AVP 97 101
27.a=rtpmap:97 iLBC/8000
28.a=fmtp:97 mode=30

Jan 22 17:31:44.295: //-1/xxxxxxxxxxxx/SIP/Info/rtpAvpCodec_to_voipCodec: Dynamic Payload :97 in SDP Body

The Asterisk box is negotiating payload 97 for ilbc codec....The problem with this is that this is a pre-configured payload type on cisco gateways as "cisco-codec-fax-ack"

You need to change the payload type for 97 on the router...

conf t

rtp payload-type cisco-codec-fax-ack 104

After you make the change..test again and send debug ccsip all from the cisco router

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[evandrojnabor]

Very nice, the problem is solved!

Thank you very much!

[Ayodeji Okanlawon]

You are most welcome!

Please rate all useful posts

"opportunity is a haughty goddess who waste no time with those who are unprepared"