Hello Guys,
I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp, but AFTER the call is established, kamailio hangs the call with "SIPS required"...
Has this happened to anyone?
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337 ᐧ
Kamailio could be ending the call, though it may also be one of the endpoints.
Anyway, if your clients are dialing sips: URIs, then it is required that the signaling be TLS end-to-end. If you are trying to translate TLS to TCP, you should use sip:user@domain.com;transport=tls. This should enforce TLS from the client -> proxy, but allow the proxy to use its preferred transport.
The reason the call wouldn't end until it's established is because it's not until this time that the any party receives a list of Record-Route headers. If using sips: and a record-route comes back that indicates that a hop did not use TLS, the call would end.
Best, Colin
On Fri, May 12, 2017 at 12:44 PM, Juha Heinanen jh@tutpro.com wrote:
David Villasmil writes:
I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on
tcp,
but AFTER the call is established, kamailio hangs the call with "SIPS required"...
Are you sure that it is K that hangs the established call?
-- Juha
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
Thanks for the replies guys!
Juha was right, it's the client... funny thing, though:
When calling:
Client->kamailio->asterisk->gw This works fine...
But when calling:
Client->kamilio->freeswitch->gw This does NOT work...
I'm thinking maybe there's some topology hiding somewhere, so that the client doesn't realizes the siganlling is being downgraded...
Any ideas? (I will take a look at the other kam's config)
David
ᐧ
Regards,
David Villasmil email: david.villasmil.work@gmail.com phone: +34669448337
On Fri, May 12, 2017 at 6:48 PM, Colin Morelli colin.morelli@gmail.com wrote:
Kamailio could be ending the call, though it may also be one of the endpoints.
Anyway, if your clients are dialing sips: URIs, then it is required that the signaling be TLS end-to-end. If you are trying to translate TLS to TCP, you should use sip:user@domain.com;transport=tls. This should enforce TLS from the client -> proxy, but allow the proxy to use its preferred transport.
The reason the call wouldn't end until it's established is because it's not until this time that the any party receives a list of Record-Route headers. If using sips: and a record-route comes back that indicates that a hop did not use TLS, the call would end.
Best, Colin
On Fri, May 12, 2017 at 12:44 PM, Juha Heinanen jh@tutpro.com wrote:
David Villasmil writes:
I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on
tcp,
but AFTER the call is established, kamailio hangs the call with "SIPS required"...
Are you sure that it is K that hangs the established call?
-- Juha
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-
Colin Morelli -
David Villasmil -
Juha Heinanen