[SR-Users] Question about sip call setup : three-way handshake
Giovanni Maruzzelli
gmaruzz at gmail.com
Wed Jun 21 14:30:04 CEST 2017
Maybe an unrelated OPTIONS ?
On 21 June 2017 at 14:20, Mititelu Stefan <stefan.mititelu92 at gmail.com>
wrote:
> Maybe PRACK, RFC 3262, is what you are looking for?
>
> ---
> Stefan
>
> On Jun 21, 2017 2:46 PM, "Abdoul Osséni" <abdoul.osseni at gmail.com> wrote:
>
> Hello,
>
> During three-way handshake (SIP call setup - INVITE-200 OK - ACK) and
> before generating or sending the final 200 OK call, is there a way to
> confirm the caller is still available (no network issue)?
>
> I want to detect broken connections (ex. the uac uses 3G network) before
> generating/sending/forwarding the final 200 OK.
>
> Thanks.
>
> Regards
> Abdoul
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Sincerely,
Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20170621/e8e0ac12/attachment.html>
More information about the sr-users
mailing list