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
--
_____________________________________________________________________
-- 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