Main features
This is a short but not exhaustive list of supported features on this beta version:

Powerful MCU (Multipoint Control Unit) for audio and video mixing
Stereoscopic (spatial) 3D and stereophonic audio
Full (1080p) and Ultra (2160p) HD video up to 120fps
Conference recording to a file (containers: .mp4, .avi, .mkv or .webm)
Revolutionary way to share presentations: documents are "streamed" in the video channel to allow any SIP client running on any device to participate
Smart adaptive audio and video bandwidth management
Congestion control mechanism
SIP registrar
4 SIP transports (WebSocket, TCP, TLS and UDP)
SA (direct connection to SIP clients) and AS (behind a server, such as Asterisk, reSIProcate, openSIPS, Kamailio…) modes
Support for any WebRTC-capable browser (WebRTC demo client at http://conf-call.org/)
Mixing different audio and video codecs on a single bridge (h264, vp8, h263, mp4v-es, theora, opus, g711, speex, g722, gsm, g729, amr, ilbc)
Protecting a bridge with PIN code
Unlimited number of bridges and participants
Connecting any SIP client (Mobiles, Tablets, Desktops, Set-top-boxes, Smart TVs...)
Easy interconnection with PSTN
NAT traversal (Symmetric RTP, RTCP-MUX, ICE, STUN and TURN)
RTCP Feedbacks (NACK, PLI, FIR, TMMBN, REMB…) for better video experience
Secure signalling (WSS, TLS) and media (SDES-SRTP and DTLS-SRTP)
Continuous presence
Smart algorithm to detect speakers and listeners
Different video patterns/layouts
Multiple operating systems (Linux, OS X, Windows …)
100% open source and free (no locked features)
Full documentation
…and many others