[asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?

[Matt Fredrickson]

On Wed, May 18, 2016 at 9:44 AM, Olivier <oza.4h07 at gmail.com> wrote:
> I've got the following setup:
>
> PSTN ---- ITSP ---- SDSL Modem-Router --<SIP ?>-- Gateway --<BRI>---
> Asterisk with B410P --- SIP Phones

Wow.

> Both SDSL Modem-Router and Gateway are managed by my ITSP.
>
> Some calls coming from PSTN and forwarded to an other PSTN number have a
> poor voice quality.

How are you forwarding them?  Is it in such a way that you remain in
the audio path, or do you get out of the audio path in the forward?

> How can I best illustrate this ?

It depends on what let has the bad audio.  If it's on the SIP side
(RTP to RTP) a pcap file will show you your perspective of audio
losses.  Received RTCP reports should show you the other side's
perspective of audio losses as well.

> A friend advised me to simply record incoming DAHDI channel, for instance.
> How can I then translate record WAV file into meaningful figures ?

If DAHDI is still in the picture in the forward scenario, that would
be another place to monitor the audio.

> More generaly, what would you suggest ?

Try to capture each leg (IP side, using tcpdump/wireshark) and on
DAHDI using dahdi_monitor or something equivalent.  Figure out if any
of your legs of audio quality issues.  If you don't see anything, it's
something at their end.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA