[asterisk-users] Asterisk cuts off intial voice path on bridging SIP channel

[Johansson Olle E]

27 jun 2008 kl. 10.35 skrev Mayur:

> I am using asterisk-1.4.21 and it is configured to pass media  
> through it for SIP calls. I have observed that if the callee answers  
> the call and starts speaking immediately for e.g. ‘Hello one two  
> three’, the caller would get to hear only ‘one two three’. From  
> packet captures I can see that asterisk receives all the RTPs from  
> the callee but it truncates the ‘Hello’ word from the voice path  
> when passing the stream on the other side.
> The signaling gets complete between caller and callee, so asterisk  
> should bridge the channels immediately. I am using canreinvite=no  
> and nat=yes option in sip.conf.
> Has anyone observed this issue why asterisk is cutting of the  
> initial voice?
>

I haven't observed it like this, but I now that when we send audio  
over NAT, it takes a while to set up all media channels. We need to  
receive RTP from both phones, in order to get a hole through the NAT  
and be able to send audio out. That usually means that some RTP  
packets we send before this happens is lost. Make sure you have turned  
off silence suppression in both telephones, so that phones has no  
delay in sending audio, even if it's just silence.

Regards,
/O