Thanks for your reply :).
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)
could you tell me Mr. Andrew Kohsmith's email? I want to add him to my contact list.
Thanks.
---
Best regards,
Qiao Yuansong
mailto: [EMAIL PROTECTED]
Thursday, April 14, 2005, 9:08:19 PM, you wrote:
> --- Qiao Yuansong wrote:
>> At the beginning of a call, the latency is not very
>> long, but it becomes more and more obvious along
>> with time. If the call keep 10 minutes, the delay
>> will be about half or one second.
>>
>> Anyone knows the reason, and any suggestion?
> Are you running your asterisk from behind a nat? My
> asterisk is behind a nat and I have the same problem
> with iax. Two other guys includong Mr. Andrew Kohsmith
> has the same issue and he is working on this problem.
> Today, he sent me an ip address to dial where he had
> echo test. The RTT (ping round trip time) from iax was
> low and the same almost all the time and the jitter
> was down to zero to this his server.
> the high jitter (variation in packet delay) causing a
> compounding problem that eventually cause the
> communication break down.The question becomes what is
> the source of the jitter: network layer, low internet
> bandwidth or some iax.conf or sip.conf configuration
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