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

[Gopalakrishnan N]

@Marrie For one way audio as a debug strategy you can enable RTP debug and
see whether you have both way packets flow SENT and GOT.

Regards


On Thu, May 2, 2013 at 6:05 PM, Johan Wilfer <lists at jttech.se> wrote:

> 2013-05-02 13:19, Marie Fischer skrev:
> > Hello everybody,
> >
> > 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.
> >
> > 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.
> >
>
> Voipmonitor.org is great for debugging voip. You can either use only the
> sniffer (opensource) and use mysql + the pcap files or you can also buy
> the commercial webgui. Either way, it's a great product.
>
> /Johan
>
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