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.