25 Mar
2022
25 Mar
'22
5:54 a.m.
Hello all,
I am trying to periodically send SIP OPTIONS to all connected WebRTC clients from
Kamailio. The functionality is similar to qualify=yes of Asterisk. Following are the
configuration changes I have made to get this working.
#!define FLB_NATSIPPING 7
<snip/>
loadmodule "nathelper.so"
<snip/>
# ----- nathelper params -----
modparam("nathelper|registrar", "received_avp",
"$avp(RECEIVED)")
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "natping_interval", 20)
modparam("nathelper", "sipping_from",
"sip:pinger@kamailio.org")
<snip/>
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER")) {
xlog("L_INFO", "Processing REGISTER in
route[REGISTRAR]\n");
if(isflagset(FLT_NATS)) {
setbflag(FLB_NATB);
# do SIP NAT pinging
xlog("L_INFO", "Setting FLB_NATSIPPING\n");
setbflag(FLB_NATSIPPING);
}
if (!save("location")) {
sl_reply_error();
}
xlog("L_INFO", "Successfully processed REGISTER in
route[REGISTRAR]\n");
exit;
}
}
When the WebRTC client registers, I can see the log: Setting FLB_NATSIPPING, but SIP
OPTIONS packets are not seen. I am checking it using the Chrome console, at client side
as well as sipdump module in server side.
Do I have to do any additional configuration? I am not posting the full config file here
so that its easy to focus on the relevant parts, but can do that if needed.
Thanks and regards,
X.