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Web based multiple way video chat and real time file sharing tool.
...p=PHP+Video+Conference
Demo: https://www.videowhisper.com/demos/conference/
Video Conference key features
+ many to many 2 way video chat
+ 3 default video panels
+ unlimited extra video panels
+ user list with online participants
+ room list with avatars
+ file sharing
+ private messaging
+ profile picture , info and link
+ push to talk
+ P2P group streaming support
+ codec setup (H264, Speex, NellyMoser)
+ acoustic echo cancellation
+ session timer and control with scripts
+ 100% web based for clients
PHP Video Conference Edition:
+ Webcam Video Conferencing Rooms
+ Create Private Rooms
+ Simple Setup
+ Easy to Install
+ Full PHP Source Code
+ Easy to Integrate
+ Profile Picture and Avatar from Webcam Snapshot
Asterisk and FreePBX on Debian, by Astiostech Sdn Bhd Malaysia
DOWNLOAD | EXTRACT | USE
A ready to use Virtual Image or VMDK of Asterisk 11/13, FreePBX 2.11/12.0.
Network:
After booting, run #ifconfig -a
Then either
#dhclient ethX
or edit manually /etc/network/interfaces
#ifup ethX
More details can be found here:
Asterisk 11: http://highsecurity.blogspot.com/2013/04/sanjay-asterisk-debian-based-asterisk.html
Asterisk 13:...
VOIP client/server in python >= 2.6. Audio in/out: ossaudiodev (UNIX like) or SoX. Network: bzip2 compression, speex or ogg audio compression, you can configure all, minimum bytes per second: 350-400 in speex mode: U8, 6 kHz, quality 0, bzip2, buf 4K
VoIP's project with Speex codec, speech detecting and crypto coding with traffic counting. All code is a pure Java, so completely cross-platforms. Traffic is about 9 MB per hour in both ends, sources: http://www.open-source-soft.narod.ru/arrow.7z
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A patent-free audio codec designed especially for voice (unlike Vorbis which targets general audio) signals and providing good narrowband and wideband quality. This project is complementary to the Ogg Vorbis codec.
Among the functionality: Include a large variety of codecs (G711, GSM, and SPEEX) - Protocol SIP - Other technical functionalities the support of DTMF (tonalities) although support ENUM (to employ numbers of SIP instead of the addresses of SIP).
Open Source IM and voice client using Jabber and SIP protocols, with great audio quality thanks to speex and a nice and clean interface. Programmed in Java and C++. It supports chat, voice and file transfers.