[Asterisk-Users] About Audio Latency from PSTN to SIP

[Andrew Kohlsmith]

I'm Andrew.

On April 14, 2005 10:01 pm, Qiao Yuansong wrote:
> My asterisk box and sip phone are not behind a nat, the sip phone and
> asterisk box are connected by LAN, so the delay is not caused by network
> congestion, and furthermore, there is no delay from sip to pstn.
>
> [sip phone]------LAN------[Asterisk with X100P]------[PSTN]
> sip to pstn (no delay)
> pstn to sip (half or one second delay)

This doesn't make any sense; the streams are identical.  Are different codecs 
being negotiated when the call origination is one side then the other?   

put

disallow=all
allow=ulaw

in sip.conf, under [general] and comment out all other allow/disallow lines.  
Restart asterisk and try again.  Something basic is not right.

-A.