[OpenSIPS-Users] LOAD_BALANCER module and calls WSS -> UDP [SOLVED]

[VoIP]

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
>>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
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