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

[Stefano Sasso]

Hello,
  I have a strange audio delay behaviour when placing a call between two SIP
devices using the same codec.
In my example, I have two devices forced to use GSM codec.
When placing a call, the first ~9sec have no audio, then the audio starts
trasmitting.
If I force one phone to use GSM and the other ULAW/ALAW, everything works
fine.

Ideas on how to solve?

thanks,
bye,
stefano
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