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

[Qiao Yuansong]

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: qys at iscas.ac.cn

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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