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. BR Jöran 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Jöran Vinzens - vinzens at sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190816/15c319c2/attachment.html>
On Fri, Aug 16, 2019, at 3:06 AM, Jöran Vinzens wrote:> 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?Noone has written the functionality to do this. The information isn't exchanged back at such a point or used to construct the answer 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.There's no ability currently to disable this. -- 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
HI All, thanks for your help. We will make our setup work correctly with reInvites. BR Jöran On Fri, Aug 16, 2019 at 11:28 AM Joshua C. Colp <jcolp at digium.com> wrote:> On Fri, Aug 16, 2019, at 3:06 AM, Jöran Vinzens wrote: > > 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? > > Noone has written the functionality to do this. The information isn't > exchanged back at such a point or used to construct the answer 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. > > There's no ability currently to disable this. > > -- > 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- Jöran Vinzens - vinzens at sipgate.de sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190816/a1b84f5a/attachment.html>