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

[Qiao Yuansong]

Thanks.

I tried your suggestion, and it make no use. 

--- 
Best regards,
 Qiao Yuansong
 mailto: qys at iscas.ac.cn

Friday, April 15, 2005, 10:21:16 AM, you wrote:

> 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.
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