Asterisk Reinvite and Codec Negotiation Problem

Asterisk Asterisk Support
1

Hi, i’m trying to use canreinvite=yes to allow for G722 direct link between phone (for High Bandwith audio).

It seems that when using reinvite, the phones are effectively switching to peer to peer direct audio path, but Asterisk does modify the allowed codecs, restricting to the configured codecs (allow=…).

As G722 is not permitted with Asterisk, then after reinvite the phones keep the original codec.

This seems absurde. It would be nice if we had the option to allow for direct codec negotiation during reinvite.

This would allow for High Bandwith audio, G722, G729.1, Speex, and others, to pass through direct links.

If someone has a workaround to get this work i would be happy to try

Thanks.

Olivier.

2

Asterisk 1.4.0 does have support for the G722 codec. Try it out, it may fix your problem.

Matt Brooks
Digium, Inc.

3

this is because * is NOT a proxy, it’s a b2bua. When you pass a call thru *- it’s not passing anything really from call 1 to call 2 except the voice data. to * they are two separate calls. Thus asterisk won’t allow or passthru a codec it knows nothing about.

4

Thanks a lot for your answer.

Is it possible to upgrade easily from * 1.2.12 to 1.4 if we are on Trixbox 2.0 ?

Thanks.

5

that i am not sure of. i don’t know much about trixbox, but i’d assume it would take at least some doing as 1.4 is significantly different from 1.2…