hello all,
I want to use kamailio like as a proxy SIP (outbound proxy) to transcode tls to udp like us:
call flow is: Softphone --> tls --> Kamailio server (kamailio listen on 443 port) --> udp --> Asterisk server (asterisk listen on 5060 port)
I know Kamailio can convert tls to udp.
According tm documentation http://kamailio.org/docs/modules/devel/modules/tm.html, I think I must use t_relay_to_udp().
Do you confirm I must replace t_relay() function by t_relay_to_udp() function?
Thank for your help.
Regards Abdoul.
Hello,
On 18/01/2017 18:07, Abdoul Osséni wrote:
hello all,
I want to use kamailio like as a proxy SIP (outbound proxy) to transcode tls to udp like us:
call flow is: Softphone --> tls --> Kamailio server (kamailio listen on 443 port) --> udp --> Asterisk server (asterisk listen on 5060 port)
I know Kamailio can convert tls to udp.
According tm documentation http://kamailio.org/docs/modules/devel/modules/tm.html, I think I must use t_relay_to_udp().
Do you confirm I must replace t_relay() function by t_relay_to_udp() function?
You have to replace it only for the initial request sent to Asterisk. Then be sure you do record routing and the requests within the dialog should be let via t_relay().
Another option is to set $du to asterisk address for the initial INVITE and then use t_relay() always.
Cheers, Daniel
Thanks: )
Regards Abdoul.
2017-01-18 19:55 GMT+01:00 Daniel-Constantin Mierla miconda@gmail.com:
Hello,
On 18/01/2017 18:07, Abdoul Osséni wrote:
hello all,
I want to use kamailio like as a proxy SIP (outbound proxy) to transcode tls to udp like us:
call flow is: Softphone --> tls --> Kamailio server (kamailio listen on 443 port) --> udp --> Asterisk server (asterisk listen on 5060 port)
I know Kamailio can convert tls to udp.
According tm documentation http://kamailio.org/docs/modules/devel/modules/tm.html, I think I must use t_relay_to_udp().
Do you confirm I must replace t_relay() function by t_relay_to_udp() function?
You have to replace it only for the initial request sent to Asterisk. Then be sure you do record routing and the requests within the dialog should be let via t_relay().
Another option is to set $du to asterisk address for the initial INVITE and then use t_relay() always.
Cheers, Daniel
-- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Abdoul Osséni -
Daniel-Constantin Mierla