[Asterisk-Users] Sounds gets choppy after 30 seconds

[Peter Dean]

Is your network equipment QoS aware? If so have you implemented a high
priority for VoIP (IAX, SIP, etc) and lower priority (SSH, HTTP, etc)
for other networking traffic?

Also it doesn't hurt to have a QoS policy on your Linux (if you are
using Linux) box where Asterisk is running too.

Plus your networked VoIP devices too.

This is what I have needed to do as I progressed from Proof-of-Concept
to Beta testing.

BTW, both iax.conf and sip.conf carry the tos parameter.
e.g. tos=0x18


On Mon, 28 Mar 2005 21:36:51 -0800 (PST), Jed Stafford <jeds at yahoo.com> wrote:
> 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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