[Asterisk-Users] Sounds gets choppy after 30 seconds

[Jed Stafford]

Well after doing some more testing it seems the
problem is really when going out via SIP to our
termination gateway, IAX seems to be working well.

Is there anything SIP specific I should be looking at
which might be causing this issue.

Regards,

-Jed

--- Jed Stafford <jeds at yahoo.com> wrote:
> This is driving me crazy, when making an outgoing
> call
> the first 30 seconds is always perfect, then the
> party
> on the receiving end can always hear me perfectly
> but
> after 30-60 seconds the audio coming back to me from
> them starts to get choppy and drops out.
> 
> I've tried this with multiple devices, from multiple
> locations some behind NAT, others not. This is using
> the ulaw codec, although i've tried it with alaw as
> well. Problem happens via IAX as well as a SIP
> channel, both calling PSTN numbers.
> 
> Network performance to the Asterisk server is good,
> 15-20ms, performed ping tests for 1-2 hours with
> almost no packet loss.
> 
> I'm willing to check or post anything needed here,
> but
> I need some fresh ideas since i've checked
> everything
> I can.
> 
> Regards,
> 
> -J
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