12 May
2017
12 May
'17
7:36 p.m.
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(a)gmail.com
phone: +34669448337
ᐧ
7:48 p.m.
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(a)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(a)lists.kamailio.org
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8:13 p.m.
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(a)gmail.com
phone: +34669448337
On Fri, May 12, 2017 at 6:48 PM, Colin Morelli <colin.morelli(a)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(a)tutpro.com> wrote:
David Villasmil writes:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
2755
days inactive
2755
days old
3 comments
3 participants
participants (3)
-
Colin Morelli -
David Villasmil -
Juha Heinanen