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

[Luca Bertoncello]

Am 22.06.2020 um 17:41 schrieb Marek Greško:

Hi

> try pinging your sip peer ip address following way:
> 
> ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress}
> 
> Post several lines and the statistics.

root at bpi:/etc/asterisk# ping -n -M do -s 1300 -i 0.1 -c 100 tel.t-online.de
PING tel.t-online.de (217.0.128.133) 1300(1328) bytes of data.
1308 bytes from 217.0.128.133: icmp_seq=1 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=2 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=3 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=4 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=5 ttl=57 time=18.1 ms
1308 bytes from 217.0.128.133: icmp_seq=6 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=7 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=8 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=9 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=10 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=11 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=12 ttl=57 time=17.7 ms
1308 bytes from 217.0.128.133: icmp_seq=13 ttl=57 time=17.8 ms
1308 bytes from 217.0.128.133: icmp_seq=14 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=15 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=16 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=17 ttl=57 time=18.2 ms
1308 bytes from 217.0.128.133: icmp_seq=18 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=19 ttl=57 time=18.4 ms
1308 bytes from 217.0.128.133: icmp_seq=20 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=21 ttl=57 time=18.2 ms
1308 bytes from 217.0.128.133: icmp_seq=22 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=23 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=24 ttl=57 time=17.8 ms
1308 bytes from 217.0.128.133: icmp_seq=25 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=26 ttl=57 time=18.1 ms
1308 bytes from 217.0.128.133: icmp_seq=27 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=28 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=29 ttl=57 time=18.1 ms
1308 bytes from 217.0.128.133: icmp_seq=30 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=31 ttl=57 time=18.3 ms
1308 bytes from 217.0.128.133: icmp_seq=32 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=33 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=34 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=35 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=36 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=37 ttl=57 time=18.1 ms
1308 bytes from 217.0.128.133: icmp_seq=38 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=39 ttl=57 time=18.0 ms
1308 bytes from 217.0.128.133: icmp_seq=40 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=41 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=42 ttl=57 time=18.1 ms
1308 bytes from 217.0.128.133: icmp_seq=43 ttl=57 time=18.1 ms
1308 bytes from 217.0.128.133: icmp_seq=44 ttl=57 time=18.1 ms
1308 bytes from 217.0.128.133: icmp_seq=45 ttl=57 time=17.9 ms
1308 bytes from 217.0.128.133: icmp_seq=46 ttl=57 time=18.1 ms
^C
--- tel.t-online.de ping statistics ---
46 packets transmitted, 46 received, 0% packet loss, time 4527ms
rtt min/avg/max/mdev = 17.784/18.058/18.454/0.190 ms, pipe 2

But now I made a test with a friend of mine, and I think the results are
very interesting...

So, we configured his mobile phone (Android) to use my Asterisk as peer.
We created also a VoIP account on the phone.
The phone was *NOT* in my WLAN.

The friend called my phone (Thomson ST2022 in local LAN). This was a
VoIP call inside Asterisk (two peers speaking together). Deutsche
Telekom was *NOT* used now!
I can hear very good the friend, without "broken voice", but *he* just
hear "like a robot with sore throat" and can't understand any word...
The same if I call ihm from my phone (via VoIP).

I tried to call my wife's phone from my phone (both in the LAN, both
Thomson ST2022). Excellent quality in both direction.

Last test: I configured my Android phone and added a VoIP-account on my
Asterisk, so now I have my Android as peer in my Asterisk.
Then I called my friend's phone (also logged in my Asterisk).
First test was with my mobile phone in my WLAN and his phone via LTE.
Terrible quality on his side (he hear me very bad), good quality on my
side (I hear ihm good).
Second test with *both my phone and my friend's phone* via LTE:
excellent quality in both directions.

Conclusion (maybe!): it can *not* be a problem in the DSL connection and
*maybe* it is not a problem in the communication with the Server of
Deutsche Telekom, since I have many problems to communicate between two
peers in local Asterisk if one is over LTE and the other in local LAN
(but curiously *not* if both peers are in local LAN or both via LTE).

Ergo: this *must* be a problem in my Asterisk...

So the questions:

1) can someone confirm or contradict my conclusions?
2) assuming are my conclusions correct, can someone suggest me where can
I search the problem?

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