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

[Administrator]

Hello

Le 23/06/2020 à 09:06, Luca Bertoncello a écrit :
> Am 23.06.2020 08:43, schrieb Luca Bertoncello:
>
> And another thing, I discovered right now...
>
>> Could you suggest me something to restrict the problem?
>> Currently, I think the problem can be:
>>
>> 1) on Asterisk
>> 2) on my Gateway/Firewall
>
> A couple of years ago I added this entry in my firewall:
>
> /sbin/iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS  
> --clamp-mss-to-pmtu
>
> since I had the problem downloading data from an Internet site using 
> my tablet.
> I found this site explaining that:
>
>    https://lartc.org/howto/lartc.cookbook.mtu-mss.html
>
> I really forgot this entry, but now I checked all entries in my 
> Firewall, and I see it, with my remark...
> Now, the last line of the HowTo:
>
> --------------------------------
> # iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS 
> --set-mss 128
>
> This sets the MSS of passing SYN packets to 128. Use this if you have 
> VoIP with tiny packets, and huge http packets which are causing 
> chopping in your voice calls.
> --------------------------------
>
> Could it be the problem? Right now I'm not at home, so I cannot test 
> it, but maybe I can add an entry like:
>
> iptables -A FORWARD -p tcp -m multiport --ports 5060,<my high port for 
> SIP> --tcp-flags SYN,RST SYN -j TCPMSS --set-mss 128
>
> and change the previous entry like:
>
> iptables -A FORWARD -p tcp -i intlan0 --tcp-flags SYN,RST SYN -j 
> TCPMSS  --clamp-mss-to-pmtu
>
> to limit the behaviour on the internal LAN...
>
> Your opinion?
Audio has nothing to do with SIP signaling 5060 port. Look at your rtp.conf

-- 
Daniel