[asterisk-users] Audio going blank for a few seconds andthencomes back. What could be the reason?

[Rob Schall]

If its all local network, then I would agree with you. In our situation,
we had people using both SIP and IAX over a home high-speed and we ran
into the problem I mentioned. We also tried to setup a IAX trunk between
2 locations where one end was on a normal high-speed connection. We
would see no more than 2-3 seconds of silence though. Any more than
that, and I agree, something much larger is the problem. However, in our
case, the connection was the problem. When we did packet trapping, we
could see the handful of packets missing, which made sense to us.

Since this certainly not this situation though, I would do the packet
capturing like everyone else is recommending. Something has to be odd there.

Zeeshan Zakaria wrote:
> Rob, as I mentioned before, here the main trunk is a T1 PRI on which
> this customer face this problem. Local phones are connected to the
> Asterisk server on their local network, and then calls go through the
> PRI. There is a VoIP trunk too only for long distance, and same
> problem happens there. So I was thinking its the network issue.
>
> On 6/1/07, *Rob Schall* <rschall at callone.net
> <mailto:rschall at callone.net>> wrote:
>
>     We have the same problem with our system. Unless you have a solid
>     (not just high speed) connection between the 2 parties, you're
>     going to get silence a few times during the call. We had set up a
>     user on a business comcast high-speed, thinking that would be more
>     than enough. Turned out though, with most high speed solutions,
>     there is some limited packet loss and its just to be expected. You
>     internet browsers, etc, would normally just re-request the packet
>     and move on, but with a stream, you're out of luck. The only real
>     solution is to have a dedicated T1 or mpls connection or something
>     like that for perfect quality. We have solid connections between
>     our offices and haven't had a problem yet.
>
>     Steve Hanselman wrote:
>>
>>     You can use tcpdump or ethereal (wireshark now) to capture the
>>     stream and then see if there was loss during the call, just leave
>>     a capture going then get your users to mark out the time at which
>>     they encountered the silence, compare this to the server time
>>     (e.g. their watch to the server) to get a time difference, then
>>     figure out what time you need to look at in the trace.
>>
>>      
>>
>>      
>>
>>     Steve
>>
>>      
>>
>>     ------------------------------------------------------------------------
>>
>>     *From:* asterisk-users-bounces at lists.digium.com
>>     <mailto:asterisk-users-bounces at lists.digium.com>
>>     [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of
>>     *Zeeshan Zakaria
>>     *Sent:* 01 June 2007 13:02
>>     *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>     *Subject:* Re: [asterisk-users] Audio going blank for a few
>>     seconds andthencomes back. What could be the reason?
>>
>>      
>>
>>     There are some remote extensions connected on this system, and
>>     calling long distance is purely on voip. These remote extensions
>>     also face the same thing, i.e. audio going blank for a few
>>     seconds, when dialing long distance. So in this case, no PRI is
>>     involved. Its either the server, or the network. Now I don't know
>>     how to find out what is it and why?
>>
>>     On 6/1/07, *Steve Hanselman* <SteveH at brendata.co.uk
>>     <mailto:SteveH at brendata.co.uk>> wrote:
>>
>>     I think this is more related to the PRI, we've been seeing this
>>     for a few weeks now, and our environment is bridged PRI-PRI on
>>     the same board.
>>
>>      
>>
>>     Steve
>>
>>      
>>
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> -- 
> Zeeshan A Zakaria
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