On Thu, 16 Apr 2020 at 13:15, Rohrbach, Gerald <[email protected]> wrote:
> Maxim, > > > > I do not know much about the technical background. > > OM is using kurento that´s my understanding. > > > > I do not know the integration depth of the old Red5 SIP in OM. > > > > For me it is not necessary, that the audio users are listed by name or > somehow, they should just listen and should be able to talk be mobile phones > > or classic phones. That of course would be a nice feature if they are > somehow listed in the room, but probably a lot of work. > > > > Before inventing the wheel again, maybe we can use another open source > project and can combine it. > > I have seen some kurento SIP projects. If the sip connection works, a > deeper integration can be done. > > Asterisks is good documented and easy to setup. > > > > Unfortunately all this takes time and needs deep understanding of the > technology OM uses. > > Probably you know much about kurento and definitely about OM. > > https://www.kurento.org/kurento-architecture > > > > What`s your opinion about this way? You know the technology best. Could > this speed up the process? > I have no idea ATM :) Investigation is actually big part of this task > > > > > Gerald > > > > > > > > > > > > > > *Von:* Maxim Solodovnik [mailto:[email protected]] > *Gesendet:* Mittwoch, 15. April 2020 18:18 > *An:* Openmeetings user-list <[email protected]> > *Betreff:* Re: SIP setup / testing > > > > Old one was RTMP based (since OM was based on Red5 which is RTMP) > > Now it is KMS and WebRTC > > So multi-media part need to be written from scratch ... > > > > On Wed, 15 Apr 2020 at 23:14, Rohrbach, Gerald <[email protected]> > wrote: > > Maxim, > > > > Audio is he only interesting with SIP. > > I will try to read about the old implementation > > > > Gerald > > > > > > *Von:* Maxim Solodovnik [mailto:[email protected]] > *Gesendet:* Mittwoch, 15. April 2020 17:34 > *An:* Openmeetings user-list <[email protected]> > *Betreff:* Re: SIP setup / testing > > > > > > > > On Wed, 15 Apr 2020 at 22:20, Rohrbach, Gerald <[email protected]> > wrote: > > Maxim, > > > > what all needs to be done for the SIP stuff? > > > > The only thing required: is to implement it > https://issues.apache.org/jira/browse/OPENMEETINGS-2239 > > > > > > I would spend some time in this, as we have some use cases for it. > > So at least I can setup an asterisks for testing purpose. > > Do you see a realistic chance to get this working in the next weeks? > > > > So far I have lots of day-time job and no idea how to implement this > > I doubt it is weeks :( > > Audio/Video transfer part need to be totally re-implemented > > > > > > I read on the older version it was working, so maybe there is not too much > to do. > > I´m not asterisk expert, but at least I have used it in some areas. > > > > Asterisk integration will work (rooms, users etc.) > > Multimedia server has been changed, so I doubt it will be fast and easy > > > > > > > > Gerald > > > > > > -- > > Best regards, > Maxim > > > > > -- > > Best regards, > Maxim > -- Best regards, Maxim
