The Open Source SIP TelePresence system

We (Doubango Telecom) are proud to announce the beta version of our open source SIP TelePresence system (https://code.google.com/p/telepresence/)

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

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 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

experience

• Secure signaling (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 shortlist is a good starting point to help you to understand what you could expect from our TelePresence system.

Google code website: https://code.google.com/p/telepresence/