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Hi i would like to be a beta tester please�best regards,
------
Ibrahim
On Tue, Jun 11, 2013 at 11:21 PM, Mamadou <diopm...@doubango.org> wrote:
Hello,
We've promised you *two* new amazing projects in 2013. The good news is that one of them is about to enter in the beta phase :)
This first project is an open source SIP TelePresence system (https://code.google.com/p/telepresence/).
If you've already asked to be beta tester then, you should have received an invitation to be part of our dev-group (https://groups.google.com/group/opentelepresence) in order to get the source code. If not, just post a message here.
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 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, http://conf-call.org/technical-guide.pdf?svn=1)
�- �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/
Happy testing and thanks for using our products :)
--
Mamadou DIOP - Technology Evangelist
Doubango Telecom - Paris, France
http://www.doubango.org
Click here to call me!
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�
�
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�
�
I would love to join the group too for the beta testing.thanks
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I would like to server as a beta tester please. please add me to your list for early source code access
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Hi,I'm Bill Baek in Korea.I want to get the Doubang source code, but i don't know how can i do for next.Would you know and answer to me how can i do ?Best Regards,Billy
2013�� 6�� 12�� ������ ���� 8�� 21�� 24�� UTC+9, Mamadou ���� ��:
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Hello : Mamadou
I am not a programmer I like to be a tester is some I can work here
Thanks Luis
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Hello,
We've promised you *two* new amazing projects in 2013. The good news is that one of them is about to enter in the beta phase :)
This first project is an open source SIP TelePresence system (https://code.google.com/p/telepresence/).
If you've already asked to be beta tester then, you should have received an invitation to be part of our dev-group (https://groups.google.com/group/opentelepresence) in order to get the source code. If not, just post a message here.
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 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, http://conf-call.org/technical-guide.pdf?svn=1)
- …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/
Happy testing and thanks for using our products :)
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To unsubscribe from this group and stop receiving emails from it, send an email to doubango+u...@googlegroups.com.
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Hello,
We've promised you *two* new amazing projects in 2013. The good news is that one of them is about to enter in the beta phase :)
This first project is an open source SIP TelePresence system (https://code.google.com/p/telepresence/).
If you've already asked to be beta tester then, you should have received an invitation to be part of our dev-group (https://groups.google.com/group/opentelepresence) in order to get the source code. If not, just post a message here.
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 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, http://conf-call.org/technical-guide.pdf?svn=1)
- …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/
Happy testing and thanks for using our products :)
Hi Sir,
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