Changing:

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);
        }
}

with:

onreply_route[handle_nat] {
        if ($socket_in =~ "wss") {
                fix_nated_contact();
        } else {
                if (nat_uac_test("private-contact")) {
                        fix_nated_contact();
                }
        }
*if (has_body_part("application/sdp") && (t_check_status("200") || t_check_status("183"))) {*
                route(RTPENGINE);
        }
}

solve the issue

This because my UDP SIP trunk reply with a 183 with SDP Annex while call between users (UDP or WebRTC) reply with 180 Ringing without Annex

Thank you. Very soon a Tutorial about WetRTC and calls UDP -> WSS WSS -> UDP branch based.


El 6/02/2025 a las 8:21 a. m., VoIP via Users escribió:

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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