[Asterisk-Users] Dead air on 7960 sip at start of call.

[Sam Tilders]

On Thu, Jun 24, 2004 at 01:45:55PM +0100, Chris Glover wrote:
> Hi,
> 
> I've noticed the same thing with my 7960 running SIP 7.1. I came to the
> conclusion that the Cisco does some clever echo cancellation, and that
> pause is the echo canceller training. I don't think it's an Asterisk
> problem. My Grandstream 101 works perfectly.

I had thought that it might be the echo canceller too... but of 400 calls
today, I've had less than 20 reports of it occurring. Some only 2 seconds
of silence, most around 5 to 6 seconds and a few over 20 seconds.

I would have thought that's an awful long time for an echo canceller and
fairly inconsistent. Or perhaps that's normal for echo cancellers?

- Sam

> On Thu, 24 Jun 2004, Sam Tilders wrote:
> 
> > Hi,
> >
> > Despite much searching, I can't find anything quite like the problem
> > we seem to be experiencing with our recently activated asterisk pbx.
> >
> > Of four people answering several calls per five minutes, I
> > have reports of the occasional call starting with dead air.
> >
> > The user picks up the call from the queue, and they can hear that the line
> > is too quiet. If they start talking, the caller either doesn't
> > hear them, or only hears the end of the first sentence.
> >
> > Reports are that this silence at the start of the call lasts
> > from 4-5 seconds, sometimes over 10 seconds and more rarely over
> > 20 seconds.
> >
> > After the period of silence the call proceeds with no further problems.
> >
> > I have taken packet dumps of extended periods, there is nothing consistently
> > different about the calls with silent start compared to those without the
> > problem. The SIP INVITE is sent to the phone, the phone rings, and OK's the
> > call. The RTP stream begins straight away.
> >
> > I'm using cvs-head as of 2004-06-10, CVS-D2004.06.09.14.00.00, recently
> > updated from v1-0_stable dated a few weeks before. This problem
> > existed with both versions.
> >
> > The phones are Cisco 7960's with SIP image 6.3, the calls are incoming
> > on an E1 connected to a TE410P. Calls are queued and the phones
> > statically defined as agents in the queue which has a ringall
> > strategy. There is no queue announcement played to the answering user.
> >
> > Asterisk in vvv verbose doesn't log anything different when this occurs.
> >
> > So, has anyone else heard of something like this? Or have a suggestion
> > for further investigation?
> >
> > Cheers.
> >
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-- 
-- 
Sam Tilders
sam at jovianprojects.com.au
(Move to Jupiter)