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

[Michael Keuter]

You could also use the 'mtr' command under Linux.

> Am 22.06.2020 um 17:41 schrieb Marek Greško <mgresko8 at gmail.com>:
> 
> Hello,
> 
> 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.
> 
> Were you also thinking about MTU problems? Not very probable, but one
> never knows.
> 
> Marek
> 
> 
> 2020-06-22 17:18 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>:
>> Am 22.06.2020 um 17:01 schrieb Telium Technical Support:
>>> I don't know if there was a prior email with more details, but....
>>> 
>>> Latency is as important as speed.  Have you checked latency between your
>>> device and pop?  What about QoS at your location, and does your ITSP
>>> support/respect QoS?
>> 
>> That's a very good idea...
>> Could you suggest me how can I check it?
>> The Gateway is a Linux with Debian 9.
>> 
>>> Could problem be inside your network?  Have you tested/optimized internal?
>> 
>> Really difficult to believe... If I call another VoIP-phone in my
>> network (using the "internal number") the quality is excellent.
>> 
>> If I call my wife using the "external number", the quality is very bad...
>> 
>> Thanks
>> Luca Bertoncello
>> (lucabert at lucabert.de)


Michael

http://www.mksolutions.info