SIP vs NAT, the configuration
Quoted 3 lines · François Legal
I guess yes, but most of the packets will go through the RELATED match, and depending on your setup, a lot of packets not related at all to SIP/VOIP could go through that match.
my setup:
no NAT, static routing, my soft-phones can call each other without problems. (I configured a local asterisk server just for this test)
on my router (debian lenny):nf_conntrack_sip ports=5060 sip_direct_signalling=1 sip_direct_media=1 iptables -t mangle -A FORWARD -m helper --helper sip -j CLASSIFY --set-class 1:100
iptables -L -n -t mangle -v -x does not match all voip traffic, just some random packets. Can anyone reproduce this?
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Quoted 2 lines · François Legal
Could you log the packets that match instead of marking, then send the log to the list ?
Hi François,
I just managed to solve it by setting "directmedia=update" on my asterisk configuration. If directmedia=yes RTP traffic is between the two clients, with directmedia=update almost all RTP traffic is via asterisk and in this way the nf_conntrack_sip module can match it. The sip_direct_media=[0|1] parameter of nf_conntrack_sip had no influence on my test.