I'm not sure this is a task for a proxy like Kamailio, even with its fancy new async relay features. To truly customise how successive INVITEs are dealt with like this, I think it needs to be handled by a UAS that has complete latitude in terms of how it computes state.


On 4 February 2014 10:13:33 GMT-05:00, Moritz Graf <moritz.graf@g-fit.de> wrote:
Hi,

sure. That's why it's called "transaction oriented".
On a SIP-Theory basis, the second INVITE opens a new transaction!! For
correlation the Call-ID keeps the same! This is how all of the big
carriers I know handle OverlapDialing.

The problem is:
* I am receiving an INVITE.
* At this point I don't know if there will be a second INVITE or not. So
I have to wait.
* If the timer elapses, ok handle this INVITE.
* But if there IS a second (third..) INVITE, I have to cancel the first
and then handle the second.

Sounds like nobody has done this before... any suggetions appriciated.

greetz


Am 04.02.2014 15:56, schrieb Alex Balashov:
 Indeed, Section 3.1 in the referenced RFC covers this pretty well.
 
 On 4 February 2014 09:47:38
GMT-05:00, "Olle E. Johansson"
 <oej@edvina.net> wrote:
 
 
     On 04 Feb 2014, at 15:43, Moritz Graf <moritz.graf@g-fit.de> wrote:
 
 
         Shortly explained what RFC3578 is: In a open numbering plan you
         never
         know if the INVITE you received is already complete, or if there are
         more numbers coming in. One way of accomplishing this is to set up a
         timer. If the timer elapses you assume the number is complete.
         If not,
         you are receiving a new INVITE with one digit more. Now you have to
         close the old transaction with a "484 - Address
         Incomplete"-response and
         start the timer again. (Find the algorithm I want to implement
         attached)
 
     You are not allowed to have two open INVITEs, the second one SHOULD get
     a 491 response. The client should not send a new INVITE if the old
INVITE
     transaction is not complete.
 
     I don't th
      ink
     overlap dialing in SIP was ever meant to be timer based. Consider
     that the first INVITE can go to a different proxy than the second. There's no
     dialog, no route set.
 
     /O

 
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     sr-users@lists.sip-router.org
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 --
 Sent from my Nexus 10, with all the figments of autocorrect that might
 imply.
 
 Alex Balashov - Principal
 Evariste Systems LLC
 235 E Ponce de Leon Ave
 Suite 106
 Decatur, GA 30030
 United States
 Tel: +1-678-954-0670
 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/


--
Sent from my Nexus 10, with all the figments of autocorrect that might imply.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/