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", "$avp(RECEIVED)") modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "natping_interval", 20) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") <snip/>
# Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { xlog("L_INFO", "Processing REGISTER in route[REGISTRAR]\n");
if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # do SIP NAT pinging xlog("L_INFO", "Setting FLB_NATSIPPING\n"); setbflag(FLB_NATSIPPING); }
if (!save("location")) { sl_reply_error(); } xlog("L_INFO", "Successfully processed REGISTER in route[REGISTRAR]\n"); exit; } }
When the WebRTC client registers, I can see the log: Setting FLB_NATSIPPING, but SIP OPTIONS packets are not seen. I am checking it using the Chrome console, at client side as well as sipdump module in server side.
Do I have to do any additional configuration? I am not posting the full config file here so that its easy to focus on the relevant parts, but can do that if needed.
Thanks and regards,
X.
Hello,
nathelper OPTIONS keepalive/ping is done only for udp contacts.
The usrloc module in the latest version of kamailio has the capability of sending keepalive for all contacts.
Cheers, Daniel
On 25.03.22 10:54, Xuo Guoto wrote:
Hello all,
I am trying to periodically send SIP OPTIONS to all connected WebRTC clients from Kamailio. The functionality is similar to qualify=yes of Asterisk. Following are the configuration changes I have made to get this working.
#!define FLB_NATSIPPING 7
<snip/>
loadmodule "nathelper.so"
<snip/>
# ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$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.
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