[asterisk-users] Voice "broken" during calls

Luca Bertoncello lucabert at lucabert.de
Sat Jun 13 15:30:28 CDT 2020

Am 13.06.2020 um 22:09 schrieb Antony Stone:

Hi Antony

> You are *assuming* that it's the codec causing the difference.

Well, I really don't know what I can think, now...

> We don't know that.
> Let me get this clear, to make sure I understand (differences emphasised):
> 1. You use *a VoIP softphone app* on your mobile, which is registered by SIP, 
> to your Asterisk server, over your home *wireless network*, to place a call to 
> some external number, you have a conversation and *the quality is excellent*.
> 2. You use your *Thomson ST2022*, which is also registered by SIP, to your 
> home Asterisk server, over your home *cabled* network, to place a call to some 
> (the same???) external number, you have a conversation and the quality is *not 
> excellent*.
> Is that an accurate summary of your situation?

Not really...

1) I have an Android phone, using the integrated Android VoIP-subsystem,
connected to my Asterisk at home, over LTE or other network *outside my
home network*. Today I called my mother using this method (I was in the
home network of my parents in law, about 20km von my home network, so
definitly *not* in my wireless...). The quality was excellent and it was
confirmed from my father in law, too...
2) I have a Thomson ST2022 connected to my Asterisk over Ethernet
(cabled network). If I call for example my mother or my parents in law,
the conversation is "broken", eg: both partner can hear little
"interruption", about 1/10 seconds in the conversation...

This is the situation...

I tried to connect the Thomson ST2022 directly to the server of Deutsche
Telekom via VoIP (excluding the Asterisk, but of couse using NAT, since
the phone does not have a public IP but just an IP in my internal
network) and then I called my father in law. Same problem... :(
I didn't get my Android phone connected to the server of Deutsche
Telekom to check how it works *outside my home network*... Not sure why
it doesn't work...

Some other information:

1) Asterisk runs on a Linux-Box (on a BananaPI) with Debian 10. Asterisk
was installed from Debian repositories.
2) The Linux-Box is directly connected to the Internet (no NAT) with a
DSL-Modem and PPPoE. Public IPv4 and IPv6 addresses are configured in a
network interface of the Linux-Box.
3) I use iptables+tc to manage a traffic shaping, privileging the VoIP
connection. If you want, I have no problem to send the
traffic-shaping-script to the list.
4) The DSL connection has a speed of 50Mbps down and 10Mbps up, and I
really think, it should be enough...
5) The phones are connected with Gbps-Ethernet to the Linux-Box.
6) On my Asterisk I configured a second VoIP-Provider (MessageNet, in
Italy), but just to *receive* calls. My contract with MessageNet does
not allow me the call someone using this connection. If someone calls my
number by MessageNet, I have the same problem I have with Deutsche
Telekom, altought not so strong, eg. the "interruptions" are not so
frequent as by calls via Deutsche Telekom... Btw: by MessageNet I must
use *gsm* as Codec, otherwise a connection will be extablished, but no
Voice can be heared...

I really appreciate any idea.
Of course, it could be possible that there is a problem on Telekom-side,
but it does not explain why I have the same problems, altought not often
as by Telekom, by MessageNet, too...

Thanks a lot
Luca Bertoncello
(lucabert at lucabert.de)

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