[asterisk-users] debug strategy for one-way audio calls

[Olivier]

Le 5 mai 2013 12:19, "Marie Fischer" <marie at vtl.ee> a écrit :
>
>
> On 04.05.2013, at 20:20, Olivier <oza_4h07 at yahoo.fr> wrote:
> > Le 2 mai 2013 13:23, "Marie Fischer" <marie at vtl.ee> a écrit :
> >>
> >> from time to time, we get so-called simplex / one-way audio calls,
where one party cannot hear the other. The only thing in common is that is
does happen with calls via SIP trunk, not ISDN and not internal calls.
Nothing strange in verbose and SIP logs. Could even be some weird
intermittent firewall issue I guess.
> >>
> > Which audio flow is missing ? Inbound ?
> >
> > I suppose it should be easier to automatically detect missing inbound
audio.
>
> Not sure about older calls, but outbound was missing the last few times.
We use call recording via MixMonitor and the recording has both flows, so I
guess rtp debug would have shown both as well.

Yes, I agree: rtp debug would probably show both flows.

So your asterisk box is quite probably sending rtp data to a SIP trunk
which do not forward it to the other party.
And this does happen from time to time (not on every call), and for a
remote party which is independant from both your provider and your own
infrastructure, right ?

This is very strange.

I would try to find conditions with which I get missing outbound audio,
100% of time but I don't have a clue on how to do it successfully.

What does your SIP provider say about this ?
Have you met this with another SIP provider ?

>
> >> Apart from logging all traffic 24/7 via tcpdump (not really
convenient), can you give me some ideas how to debug this kind of issue?
> >> Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters.
>
> --
>
> marie
>
>
>
>
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