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

Am 23.06.2020 16:22, schrieb Marek Greško:
> It seems your problems lie in something other. Most probably it is not
> mtu problem. All my suspections are contradicted. If it is true you
> have inter vlan voice quality problems, it is definitely something
> different. Formerly I assumed you were trying only LTE vs LAN using
> internet.
I'm not sure what you mean with the last sentence...
I tried to connect to my Asterisk via LAN or via DSL (either via LTE or 
other DSL).
Then I noticed that if I call another peer in same network (= both peers 
via DSL or both peers in the same VLAN), the quality is very good, 
otherwise is very poor.

But why should Asterisk have problem if the peers are in different 
networks it's for me a really big mistery...

This evening I'll try to capture the pakets in a call between two peers 
connected to Asterisk via LTE, two peers connected in the same LAN and a 
peer connected via LTE and the other in LAN, then maybe it's possible to 
find the problem...

But if you have any other idea, I'm very happy to hear it! ;)

Thanks
Luca Bertoncello
(lucabert at lucabert.de)

I interchanged LAN and LTE in the sentence.

Do you have some kind of NAT in fron of asterisk? Or is your asterisk
having public IP? Could you share sip.conf (without passwords)? One
LAN client, one LTE and general section.

Marek


2020-06-23 16:29 GMT+02:00, Luca Bertoncello <lucabert at
lucabert.de>:
> Am 23.06.2020 16:22, schrieb Marek Greško:
>> It seems your problems lie in something other. Most probably it is not
>> mtu problem. All my suspections are contradicted. If it is true you
>> have inter vlan voice quality problems, it is definitely something
>> different. Formerly I assumed you were trying only LTE vs LAN using
>> internet.
>
> I'm not sure what you mean with the last sentence...
> I tried to connect to my Asterisk via LAN or via DSL (either via LTE or
> other DSL).
> Then I noticed that if I call another peer in same network (= both peers
> via DSL or both peers in the same VLAN), the quality is very good,
> otherwise is very poor.
>
> But why should Asterisk have problem if the peers are in different
> networks it's for me a really big mistery...
>
> This evening I'll try to capture the pakets in a call between two peers
> connected to Asterisk via LTE, two peers connected in the same LAN and a
> peer connected via LTE and the other in LAN, then maybe it's possible
to
> find the problem...
>
> But if you have any other idea, I'm very happy to hear it! ;)
>
> Thanks
> Luca Bertoncello
> (lucabert at lucabert.de)
>
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Am 23.06.2020 um 17:04 schrieb Marek Greško:
> I interchanged LAN and LTE in the sentence.
OK...
> Do you have some kind of NAT in fron of asterisk? Or is your asterisk
No, Asterisk has a public IP. No NAT in front of Asterisk...
> having public IP? Could you share sip.conf (without passwords)? One
> LAN client, one LTE and general section.
Of course:

my outgoing configuration:
[pbxluca]
type=peer
defaultuser=<my username @telekom>
secret=<my very password>
dtmfmode=rfc2833
host=tel.t-online.de
context=luca_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=03511111111
fromdomain=tel.t-online.de
usereqphone=yes
canreinvite=yes
insecure=port,invite
nat=no
qualify=yes
qualifyfreq=600
disallow=all
allow=alaw
allow=ulaw

my phone configuration:
; Lucas Telefon
[00493511111111]
fullname = 00493511111111
secret = <very secret password>
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
deny=0.0.0.0/0.0.0.0
permit=192.168.200.0/255.255.255.0
dial=SIP/00493511111111

my mobile phone:
; Lucas Handy
[00491772222222]
fullname = 00491772222222
secret = <secret>
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
dial=SIP/00491772222222
allow = all

sip.conf:
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0:25572
tcpenable=yes
tcpbindaddr=0.0.0.0:25572
tlsenable=no
tlsbindaddr=0.0.0.0:25573
transport=udp
srvlookup=no
minexpiry=480
defaultexpiry=480
disallow=all
allow=alaw
allow=ulaw
allow=ilbc
allow=g729
allow=g723
allow=gsm
language=de
alwaysauthreject = yes
tlscertfile=/etc/asterisk/ssl/asterisk.pem
tlscafile=/etc/asterisk/ssl/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
callcounter = yes
t38pt_udptl = yes
faxdetect = no
register =>
03511111111:xxxxxxxx:333333333333-0001 at t-online.de@pbxluca/00493511111111
register =>
03511111112:xxxxxxxx:333333333333-0001 at t-online.de@pbxfax/00493511111112
register =>
03511111113:xxxxxxxx:333333333333-0001 at t-online.de@pbxanika/00493511111113
register => 555555555:xxxxxxxx at messagenet/555555555
register => lucabertoncello:xxxxxxxxx at rebvoice/lucabertoncello
jbenable = no
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = fixed

Thanks
Luca Bertoncello
(lucabert at lucabert.de)