[asterisk-users] Voice broken during calls (again...)

Michael Maier m1278468 at mailbox.org
Mon Jun 22 14:30:50 CDT 2020

Am 22.06.20 um 16:48 schrieb Luca Bertoncello:
> Hi list!
> So, now I have a business contract and a technician was here to check
> the DSL...
> Nothing found, except that for 50Mbps I need now vectoring. Really
> nice... A couple of years ago I could get 50Mbps without vectoring.
> Of course, Deutsche Telekom said nothing about this change...
> Well, I got it working, and now I have 48Mbps down and 10Mbps up.
> I _REALLY CAN'T_ believe, that this is not enough...

This is enough if you're doing it correctly. But that's your job to do it correctly - not Telekom's one.

> The problem with many little disruptions during calls is always here.

Not surprising. That's most probably not a problem of the provider. VoIP of Deutsche Telekom mostly is pretty perfect 
regarding voice quality and availability.

> I tried changing the codecs and changing some settings in the SIP
> configuration of the peers.
> No changes...

Not surprising.

Did you check to prevent transcoding?

> 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?

> 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.

> @all german users using Telekom: how did you configured your Asterisk?

- At first, you have to trace down the problem and analyze those traces when the problem occurred. This could be done 
with pcapsipdump[1] on both sides (internal and external).

	pcapsipdump -i ppp0 -p -d /tmp/pcapsipdump &

will trace the connection to Telekom. You have to add another process to another device to trace the internal call.
Use Wireshark to analyze the dumps. Wireshark understands VoIP. (I assume you are using SIP / RTP on all legs.) Now you 
can see on which side the problem happens and how it looks like.
- Are you using NAT or is asterisk running on the device which runs the ppp-interface?
- What's the modem you are using? What about the wiring between APL and modem? Is it done correctly? [2]
- Did you configure prioritization for the up-stream regarding RTP and SIP? This is done with the tc tool.
- 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.
- 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)


[1] https://sourceforge.net/projects/pcapsipdump/
[2] https://telekomhilft.telekom.de/t5/Telefonie-Internet/Das-richtige-Kabel-zwischen-APL-und-TAE-Dose/ta-p/3499089

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