the open source SIP TelePresence system
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Configuration.md
Configuration_AVPF_tail_length.md
Configuration_Audio.md
Configuration_Audio_Video_codecs.md
Configuration_Bridge_configuration.md
Configuration_Debugging_the_system.md
Configuration_NAT_Firewall_traversal.md
Configuration_Presentation_sharing.md
Configuration_RTP_buffer_size.md
Configuration_Recording_conference_to_a_file.md
Configuration_SIP_network_transports.md
Configuration_SIP_registration.md
Configuration_Security.md
Configuration_Video.md
Makefile.am
README.md
SSL_CA.der
SSL_CA.pem
SSL_Priv.pem
SSL_Pub.der
SSL_Pub.pem
Support_BuildingSourceCode.md
Support_FAQ.md
Support_Known_issues.md
Support_Roadmap.md
Support_Testing_the_system.md
Support_Tips.md
Technical_Audio_mixer_design.md
Technical_Bandwidth_management_and_congestion_control.md
Technical_Call_session_management.md
Technical_NAT_and_firewall_traversal.md
Technical_Presentation_sharing.md
Technical_Protecting_a_bridge_with_password.md
Technical_Recording_conference_to_a_file.md
Technical_SA_versus_AS_modes.md
Technical_Security.md
Technical_Stereoscopic_spatial_3D_audio.md
Technical_Video_mixer_design.md
Technical_Video_quality.md
autogen.sh
configure.ac
opentelepresence.sln
self-signed-certs.sh
telepresence.cfg
telepresence.vcproj

README.md

Source code freely provided to you by Doubango Telecom ®. This is a good and viable open source alternative to Google Hangouts.

Demonstration Presentation sharing

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

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

Getting started

  1. Read the technical guide for more information on how to build, install and run the system
  2. Test the system as explained here
  3. Share issues and technical questions on our developer group
  4. Find our roadmap here

Even if any SIP client could be used we highly recommend for this beta version to use our WebRTC demo client to ease debugging.

Technical help

Please check our issue tracker or developer group if you have any problem.
We highly recommend reading our Technical guide.
Please check the list of known issues before reporting.