When conferencing with SIP you usually have two options:
Is there a way of having all the pros by parallel calling all targets and sending them the multicast address and port they need to listen to? We're currently using Asterisk, but I was looking at FreeSWITCH or Kamailio/OpenSIPs, with little luck.
There are no realization like that.
But you can do it by modify client soft to accept special sip message with multicast stream.
After that you have send in dialplan that stream when you want it listen multicast.
Asterisk can do multicast without any issues
Also you can do send multicast address in sip INVITE. If client support that(most clients not support), you can get calls with multicast rtp in downstream direction and usual in other direction.
if you don't use transcoding, FreeSWITCH will be able to handle hundreds of simultaneous sessions from a single multicore server. So it's probably not worth the effort to build something special. Also you're assuming that all listeners are connected with SIP, and multicast routing is done properly in the whole network. In real life, you may get duplicated RTP streams, or some users which are not receivuibg the multicast stream, and so on. So, IMHO it's not worth the effort, and a normal multipoint SIP conference will do just fine.
also, see here: https://wiki.freeswitch.org/wiki/Mod_esf
if your endpoints can statically listen to a specific multicast group and port, you can easily send them a multicast RTP stream in PCMU.