We (Doubango Telecom) are proud to announce the beta version of our open source SIP TelePresence system (https://groups.google.com/group/opentelepresence)
This is a short but not exhaustive list of supported features on this
beta version:
This short list is a good starting point to help you to understand what
you could expect from our TelePresence system.
Technical guide: http://conf-call.org/technical-guide.pdf
Google code website: https://code.google.com/p/telepresence/
Minimal WebRTC Demo client to test all features: http://conf-call.org/
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)
- 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
- Support for any WebRTC-capable browser (WebRTC demo client at
- Mixing different audio and video codecs on a single bridge (h264,
- Protecting a bridge with PIN code
- Unlimited number of bridge and participants
- Connecting any SIP client
- Easy interconnection with PSTN
- NAT traversal (Symmetric RTP, RTCP-MUX, ICE, STUN and TURN)
- RTCP Feedbacks (NACK, PLI, FIR, TMMBN, REMB...) for better video
- 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 (https://code.google.com/p/telepresence/w/list,
- ...and many others
This short list is a good starting point to help you to understand what
you could expect from our TelePresence system.
Technical guide: http://conf-call.org/technical-guide.pdf
Google code website: https://code.google.com/p/telepresence/
Minimal WebRTC Demo client to test all features: http://conf-call.org/