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

[Luca Bertoncello]

Am 23.06.2020 09:19, schrieb Administrator:

Hi Daniel

> Audio has nothing to do with SIP signaling 5060 port. Look at your 
> rtp.conf

You're right...
I have to restrict to the ports I configured in rtp.conf...
So like:

iptables -A FORWARD -p tcp -m multiport --ports -ports 10000:15100 
--tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128

?

Or I just have to use:

iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 
128

instead of:

iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS  
--clamp-mss-to-pmtu

?

Thanks
Luca Bertoncello
(lucabert at lucabert.de)