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

[Michael Maier]

On 23.06.20 at 21:10 Luca Bertoncello wrote:
> Am 23.06.2020 um 21:08 schrieb Michael Maier:
>> On 23.06.20 at 08:05 Luca Bertoncello wrote:
>>> Am 23.06.2020 07:27, schrieb Luca Bertoncello:
>>>
>>> I again
>>>
>>>>> Do not change MTU. Probably there will be another problem. I expect
>>>>> packet size 1466 would pass and higher will have the same result. It
>>
>> RTP-VoIP-packets never reach this size. Size is about 214 bytes.
> 
> OK, so it must be something other...
> 
> But I really don't have any idea what... :(

Your basic architecture looks good to me - now you have to start the analysis of the problem with pcapsipdump and wireshark as I wrote before to get an idea what actually happens at
all. You most probably won't come any further without doing any analyzing. You have to learn it. It will take some, or even more, time. You can't do it in just few hours or maybe
even days or weeks. It is work or even hard work to learn and to do it.

That's my problem: It's impossible for me to assist because I can't see any effort on your side to learn. I won't fix your problem. You have to fix it yourself. All I can do, is, to
show you a way to *find* your problem (I can't know your problem) and may be to give some hints how to fix it (once you've found it). Finding / localizing problems and fixing them
are two completely different things. But before you fix a problem, you have to know the problem. Therefore: go and find your problem by starting the analysis. That's the first thing
to do.


Regards
Michael