[asterisk-users] Voice "broken" during calls

Antony Stone Antony.Stone at asterisk.open.source.it
Tue Jun 16 03:48:10 CDT 2020

On Tuesday 16 June 2020 at 08:18:51, Luca Bertoncello wrote:

> > sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
> > sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &
> eth0 is my DSL interface and eth1 my phone interface?

Well, one is internal (phone) and the other is external (DT), doesn't matter 
which way round.

> tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de &
> tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of my
> phone) &

Looks like you name your Banana interfaces very similarly to mine :)

However, I would be careful with that first one, containing "host tel.t-
online.de".  I don't use DT, so I can't be sure, but I guess this is the SIP 
server to which you register with the account credentials...

It *may not* be the same machine as handles the RTP packets - that is 
negotiated separately between Asterisk (or the Thomson, when it's connected 
directly to DT) as part of the SIP INVITE / Acknowledge.

So, you *could* find that you capture all of the SIP traffic and none of the RTP 
traffic.  On the other hand, you might get everything.

You can be pretty sure it's worked if you do the above and then find that the 
two packet capture files are approximately the same size.  If the DT one is 
significantly smaller (by which I mean a factor of at least ten different), then 
omit the "host" parameter on that capture and try again...


A few words to be cautious of between American and English:
 - momentarily
 - suspenders
 - chips
 - pants
 - jelly
 - pavement
 - vest
 - pint (and gallon)
 - pissed

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