the open source SIP TelePresence system

We (Doubango Telecom) are proud to announce the beta version of our open source SIP TelePresence system (

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
  • 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
  • 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

This short list is a good starting point to help you to understand what
you could expect from our TelePresence system.

Technical guide:
Google code website:
Minimal WebRTC Demo client to test all features:
Created by: alice27, Last modification: Mon 17 of Jun, 2013 (21:05 UTC)
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