[asterisk-users] PJSIP reInvite

Jöran Vinzens vinzens at sipgate.de
Fri Aug 16 00:53:41 CDT 2019

Hi all,

So the scenario is:

A -> Asterisk -> B

after B send back 200 OK Asterisk is answering the call to A. Directly
after the Answer Asterisk generates a ReInvite to A and the only difference
between the 200 OK sdp and the reInvite sdp are the offered codecs which
are forwarded from B to A. Here i do not understand why this could not be
done in the 200OK to A?

As far as i understood you Josh, there is no way to prohibit this kind of
reInvite? It is not about route Optimization just for some more options for
the A Party.


On Thu, Aug 15, 2019 at 4:07 PM Joshua C. Colp <jcolp at digium.com> wrote:

> On Thu, Aug 15, 2019, at 8:23 AM, Jöran Vinzens wrote:
> > Hi All,
> >
> > We are using asterisk 16.5 and having an issue with the first re-invite
> > after the call has been established.
> > We can see the call gets up and you see in the logs the bridge type has
> > changed and after that a re-invite is triggered.
> >
> > Is there any possibility to deactivate this kind of reInvite? We have
> > some race conditions while have multiple asterisk in the call flow and
> > the different asterisk systems are sending this reInvites out parallel.
> > While an invite is pending on a system it is not accepting another
> > incoming reInvite from peer.
> >
> > With chan_SIP canreinvite=no solved the issue. But it seems there is
> > nothing similar in PJSIP.
> The "direct_media" option controls just that, direct media. A reinvite can
> occur for other reasons (such as attempting to renegotiate streams to be of
> better quality or to update connected line information). Have you
> determined which case is occurring?
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
> --
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Jöran Vinzens - vinzens at sipgate.de

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
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