Hello,
yes call record_route function here:
# record routing
if (!is_method("REGISTER|MESSAGE"))
record_route();
and fix_nated_contact here:
onreply_route[handle_nat] {
if (nat_uac_test("private-contact")) {
fix_nated_contact();
}
if ($socket_in =~ "wss") {
fix_nated_contact();
}
if (has_body_part("application/sdp") && t_check_status("200")) {
route(RTPENGINE);
}
}
Regards
El 6/02/2025 a las 8:11 a. m., Răzvan Crainea escribió:
Hello!
Are you calling record_route? Also, make sure you call
fix_nated_contact() on the 200 OK. Read this blog post for more
information:
https://blog.opensips.org/2017/02/22/troubleshooting-missing-ack-in-sip/
Best regards,
Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com
On 1/31/25 3:55 PM, VoIP via Users wrote:
Good morning everyone,
I'm trying to implement this type of scenario:
WSS -> load_balancer -> UDP Gateway (Asterisk)
Everything works up to the 200 OK received from the gateway and
forwarded from OpenSIPs to the WebRTC clients.
I don't see the ACK sent from the WebRTC client to OpenSIPs to commit
the 200OK.
WebRTC -> UDP and UDP -> WebRTC calls between users work correctly
and analyzing the 200 OK of a call between users and a call via
load_balancer, the truth is that I do not find differences that
justify this type of error.
I'm writing a tutorial, for now in Spanish, dedicated to the subject
but without that piece I can't finish it.
Thank you in advance for the help
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Users mailing list
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
