There are 3 potential ways to do this as Jitsi is an SFU (selective forwarding unit) and FreeSWITCH is an MCU (multi-point control unit)
Wait for FreeSWITCH to support the SFU like Asterisk has done or just use Asterisk instead.
Use a gateway like Jigasi to bridge between both
Run parallel conferences on FS and JVB (Jitsi Videobridge) and make clients open two webrtc peerconnections to both of them.
Option 1 is non-starter for me. I have commited to FS for now.
I started out with option 2 but could not get it to work a while back. Part of the problem was that it was hard-wired to use port 5060 which made it difficult to run OF, JVB, FS and Jigasi all on the same server box.
I then moved to option 3 which works fine for me. In this mode, video is by SFU and audio is by both SFU and MCU. MCU is used specificaly by telephony participants. I use the ability in FS to mute all audio exclusively between JVB users in the MCU. Only the audio between participants that join by telephone and JVB users get mixed by FS. It works on both incoming and outgoing telephone calls to FS.
As I use the same alphanumeric conference names from JVB in FS, some extra work is required to the FS dialplan to actually make it work.
However, the lastest version of JitsiMeet is causing an issue with SIP.js. The JitsiMeet chat stops working when I am using both libraries and I have disabled this feature in my clients Etherlynk and Pade for now.
There are 3 potential ways to do this as Jitsi is an SFU (selective forwarding unit) and FreeSWITCH is an MCU (multi-point control unit)
Option 1 is non-starter for me. I have commited to FS for now.
I started out with option 2 but could not get it to work a while back. Part of the problem was that it was hard-wired to use port 5060 which made it difficult to run OF, JVB, FS and Jigasi all on the same server box.
I then moved to option 3 which works fine for me. In this mode, video is by SFU and audio is by both SFU and MCU. MCU is used specificaly by telephony participants. I use the ability in FS to mute all audio exclusively between JVB users in the MCU. Only the audio between participants that join by telephone and JVB users get mixed by FS. It works on both incoming and outgoing telephone calls to FS.
As I use the same alphanumeric conference names from JVB in FS, some extra work is required to the FS dialplan to actually make it work.
However, the lastest version of JitsiMeet is causing an issue with SIP.js. The JitsiMeet chat stops working when I am using both libraries and I have disabled this feature in my clients Etherlynk and Pade for now.