[asterisk-users] Change codec when dial from SIP to DAHDI

d tbsky tbskyd at gmail.com
Thu Sep 25 21:21:17 CDT 2014

2014-09-26 3:34 GMT+08:00 Eric Wieling <EWieling at nyigc.com>:
> You will find not transcoding much less useful that one might imagine.
    can you give some more hint about the topic?

in my testing, if the sip phone use G.722 and the sip trunk use G.711,
I can hear the quality is not as good as both side use G.711.

    but you maybe right when both legs use G.711 but transcoding in
the middle. the quality seems not so bad but I have test it very

    thanks a lot for your help!!


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