On Wed, May 18, 2016 at 9:44 AM, Olivier <[email protected]> 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 -- _____________________________________________________________________ -- 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
