[asterisk-users] Diagnosing call problem

Mitch Claborn mitch_ml at claborn.net
Thu Mar 21 12:34:43 CDT 2013


I did open a ticket with SFL support and sent them the packet trace.

Interestingly, using Bria we sometimes see similar, though not exactly 
the same, symptoms.  That would make me wonder about the TCP stack on 
the client machine, or similar.

Bria on Ubuntu is not terribly stable.  Bria on the Mac works very well, 
but that's a pretty expensive solution.

We are close to ditching the soft phones entirely for this call center 
and going to the Digium D40.  I put one of those in service this morning 
and the calls are noticeably clearer and there have been no reported 
problems.


Mitch

On 03/21/2013 09:48 AM, Matthew J. Roth wrote:
> Mitch Claborn wrote:
>>
>> Thank you for that most excellent post.  I had guessed at most of the
>> SDP fields and meaning.
>
> No problem.  I actually like looking at SIP traces for some reason.
>
>> I have wireshark traces from the client and the RTP packets are not in
>> the trace, which I think means that the client software is simply not
>> producing them.  I have opened a ticket with SFL phone support and will
>> post here if I find anything.
>
> That's a reasonable conclusion.  Just make sure that you get some traces of good
> calls to verify that your tests are valid.
>
>> I did test the "muted microphone" theory.  SFLphone continues to send
>> RTP packets even when the mic is muted, so that doesn't seem to be the
>> cause.
>
> It's always a good idea to rule out PEBKAC before spending a lot of time
> diagnosing a problem.
>
>> I've also compared the call initiation SIP and SDP packets between a
>> call that fails and one that works correctly.  I can discern no
>> difference other than things like port numbers and call IDs.
>>
>> Tomorrow I'll be trying one of my agents on Bria instead of SFL - maybe
>> that will make a difference.
>
> It really seems like it may be a problem with the softphone.  I'm sure the
> developers of SFLphone will appreciate your feedback, because not sending RTP is
> a pretty serious bug.
>
> I'll keep an eye on this thread and help out if I can.
>
> Regards,
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                 http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users
>



More information about the asterisk-users mailing list