Am Thu, 20 Jun 2024 14:14:11 -0000
schrieb smartin114--- via sr-users <[email protected]>:

> Invite to carrier is good, sent over tls and so the reply from the carrier, 
> the 200 ok.  The ACK is sent via UDP and so is the Bye, which I didn't 
> include.

I fear I need more information and a complete example to see what might
be going wrong.

INVITE: FS => TLS => Kamailio => UDP => Carrier
some more messages
200 OK: Carrier => UDP => Kamailio => TLS => FS
and the ACK to the 200 OK, they are all fine?

So the Problem is the BYE when sent which direction? From the Carrier?

I assume this is the case.

Basically you need to look at the INVITE from the FS.

Does the Contact: header contain a transport=tls attribute? (According
to your example, it does)

So this information needs to be stored on the Carrier side (or whatever
B2BUA an the Carrier side handles the call) and when the carrier issues
a BYE the R-URI of that BYE needs to contain transport=tls as this is
the information, the last HOP towards the Fs side will be using when
all Route: Header have been consumed.

If this is NOT the case and transport=tls is missing in the BYE R-URI
the IMHO SIP implementation Carries side is 'broken', you face the same
issue I have had. :-)

If you are able to put other attributes in the invite Contact header,
and they are sent back the BYE from the Carrier, then you can copy the
transport value to a custom contact attribute (I use t=) and restore the
transport attribute on the R-URI from that.

Mit freundlichen Grüssen

-Benoît Panizzon-
-- 
I m p r o W a r e   A G    -    Leiter Commerce Kunden
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