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

[Michael Maier]

Am 24.06.20 um 08:10 schrieb Luca Bertoncello:
> Am 24.06.2020 05:05, schrieb Michael Maier:
> 
> Hi
> 
>> Your basic architecture looks good to me - now you have to start the
> 
> Nice to hear it...
> 
>> 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.
> 
> Well, that's the very problem...
> I don't know *how* to analyze it... Or, better, how to read the data...
> I know, I can use tcpdump, sngrep and many other tools, but I don't know what I have to expect and how to decide, that a 
> paket is wrong...
> Can someone help me to learn it?
Google is your friend as usual. Try *for example* those search patterns as *entry point*:
wireshark rtp stream analysis
wireshark voip mitschneiden

https://support.yeastar.com/hc/en-us/articles/360007606533-How-to-Analyze-SIP-Calls-in-Wireshark
https://www.innosoft.at/news/169/voip-grundlagen-wireshark-analyse-von-sip-telefonie
https://sharkfestus.wireshark.org/sharkfest.12/presentations/BI-7_VoIP_Analysis_Fundamentals.pdf


Regards
Michael