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

[chawki hammoud]

--- Qiao Yuansong <qys at iscas.ac.cn> 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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