asterisk users - Oct 2011 - strange delay behaviour in SIP call with same codec

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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> 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.
If I had to guess, I'd say that you don't have
canreinvite/directmedia=no in sip.conf and there is possibly a NAT between the
phones and Asterisk. When they have the same codec and directmedia is enabled,
the phones will try to communicate directly to each other. It sounds like it is
taking a while for firewalls to allow this traffic through (since both phones
have to send packets out to the other before the whole is opened up in many
setups). When the codecs are forced to be different, the media will go through
Asterisk to get to the phones instead, alleviating your issue.

Terry