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

[VoIP]

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
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