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

[Marek Greško]

Would you mind repeating the test with canreinvite=no set for all you
phones and mobile phones?

What is your upload bitrate? Is it guaranteed?

I would try also to test the PMTU:

Try:

ping -M  do -s 2000 ${ip address of the sip server}

You should receive icmp asking for lowering the packet size.

The LTE phones could have lower MTU and thus overcome PMTU problem.

Marek


2020-06-22 21:48 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>:
> A thing I forgot to report...
> My Asterisk listen on an high port (*not* 5060), since I had many
> problems in the past with someone trying to use my Asterisk with brute
> force attack...
>
> I really don't think, this can be the problem, but better to report all...
>
> Regards
> Luca Bertoncello
> (lucabert at lucabert.de)
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users