[asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?

[Kristian Kielhofner]

On Sat, Sep 27, 2008 at 5:54 PM, Philip Prindeville
<philipp_subx at redfish-solutions.com> wrote:
> I've got the following situation.  I'm running Asterisk 1.4.18 on a
> firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones
> behind it.
>
> I'm peering SIP with a Coppercom switch sitting behind an SBC.
>
> On outbound calls, I get 2-way voice, no worries.
>
> On inbound calls, I get one-way voice (I can hear the caller but they
> can't hear me).
>
> I've looked at tcpdumps of the RTP traffic, and the addresses and port
> numbers correspond to what's in the SIP INVITE/OK messages (assuming
> that they don't somehow get munged by NAT after tcpdump looks at them --
> there is no NAT device upstream of my Asterisk firewall).
>
> I'll look into using Record() or Monitor() to capture the phone call,
> but if there's any conversion being done by codecs then that won't
> eliminate the possibility that the code itself is misconfigured or buggy
> and generating a bad stream on one of the legs...
>
> Anyone have an idea about how to best go about troubleshooting this?
>
> Thanks,
>
> -Philip
>

Philip,

  We were recently having a few call quality problems with one of our
carriers, including very mysterious one way audio on specific pieces
of equipment.

  I created a call generator using Playtones/Monitor to record all
four audio paths (only two are important) of a successful call and
analyze the resulting recordings with ecasound to detect distortion,
one-way audio, audio drops, etc.  After several hundred calls we were
able to get the carrier to correct the offending pieces of equipment.

  I'm looking into a way to do this in real time but for now this
collection of scripts works pretty well.  It's not ready for release
but I could get it to you shortly for some testing.

-- 
Kristian Kielhofner
http://blog.krisk.org