[SR-Users] TLS to TCP

David Villasmil david.villasmil.work at gmail.com
Fri May 12 19:13:22 CEST 2017


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 at gmail.com
phone: +34669448337

On Fri, May 12, 2017 at 6:48 PM, Colin Morelli <colin.morelli at 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 at 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 at 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
>>
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>
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