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:[email protected]")
> <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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--
Daniel-Constantin Mierla -- www.asipto.com
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__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
* [email protected]
Important: keep the mailing list in the recipients, do not reply only to the
sender!
Edit mailing list options or unsubscribe:
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