[asterisk-users] strange delay behaviour in SIP call with same codec

Terry Wilson twilson at digium.com
Thu Oct 20 09:37:09 CDT 2011


> I have canreinvite and directmedia to 'no' - and there is no NAT
> between the phones and asterisk...

Hmm. In that case, I'm not sure. You could take a look at the output of "rtp set debug on" when the call is going on to see what is going on with the audio.



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