asterisk users - Jun 2020 - Voice broken during calls (again...)

Am 22.06.2020 um 21:30 schrieb Michael Maier:
> Did you check to prevent transcoding?
could you explain what do you mean and how to check it?
>> On the Gateway (Banana PI), where the Asterisk server also runs, the
>> load is about 0.50 during calls and it has a Gbps LAN.
> 
> What's running on this device on parallel? What about other network
> traffic - not necessarily to the internet interface?
On the BananaPI? Nothing other PPP, Bind, NTP, Firewall (iptables) and
Asterisk.
>> I can't believe, the problem is here...
> 
> That's irrelevant. You have to ensure, that the driver doesn't have
any
> problems. Reducing the queue sizes of the interface may help.
I don't understand what you mean...
> - Are you using NAT or is asterisk running on the device which runs the
> ppp-interface?
Asterisk runs on PPP interface
> - What's the modem you are using? What about the wiring between APL and
> modem? Is it done correctly? [2]
Zyxel VMG1312B30A. It works correctly and using the Internet (upload and
download) is not a problem
> - Did you configure prioritization for the up-stream regarding RTP and
> SIP? This is done with the tc tool.
Yes
> - Did you correctly configure tos? For Deutsche Telekom you may use
> tos=0xb8 (pjsip). You have to verify it with Wireshark with your traces.
> You have to set it to the same value as the packages which are received
> from their server.
I use SIP, not PJSIP... Do I have to do that, too? Which value?
> - You have to use the DNS of Deutsche Telekom which they provide during
> the ppp-login because they usually provide optimal sip servers for you
> (regarding distance). You're RTT of ping (18 ms) is pretty bad. I'm
> having here 5 ms to the primary server (Telekom provides 3). See
> 
>     dig +noall +answer _sip._udp.tel.t-online.de SRV
> 
> e.g. (don't know the hostname for the business infrastructure)
I have a forwarding to the DNS servers of Telekom configured in my bind,
since the Gateway has to manage the internal domains, too...

Regarding the ping time: wich line do you have? I have a DSL 50Mbps.
Maybe your times are better due to a faster line?

What is your opinion about the tests I did today with the friend and his
phone as VoIP-peer?

Thanks
Luca Bertoncello
(lucabert at lucabert.de)

A thing I forgot to report...
My Asterisk listen on an high port (*not* 5060), since I had many
problems in the past with someone trying to use my Asterisk with brute
force attack...

I really don't think, this can be the problem, but better to report all...

Regards
Luca Bertoncello
(lucabert at lucabert.de)

Would you mind repeating the test with canreinvite=no set for all you
phones and mobile phones?

What is your upload bitrate? Is it guaranteed?

I would try also to test the PMTU:

Try:

ping -M  do -s 2000 ${ip address of the sip server}

You should receive icmp asking for lowering the packet size.

The LTE phones could have lower MTU and thus overcome PMTU problem.

Marek


2020-06-22 21:48 GMT+02:00, Luca Bertoncello <lucabert at
lucabert.de>:
> A thing I forgot to report...
> My Asterisk listen on an high port (*not* 5060), since I had many
> problems in the past with someone trying to use my Asterisk with brute
> force attack...
>
> I really don't think, this can be the problem, but better to report
all...
>
> Regards
> Luca Bertoncello
> (lucabert at lucabert.de)
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users