Search Results for "speex"

VoIPmonitor is open source network packet sniffer with commercial frontend for SIP SKINNY MGCP RTP and RTCP VoIP protocols running on linux. VoIPmonitor is designed to analyze quality of VoIP call based on network parameters - delay variation and packet loss according to ITU-T G.107 E-model which predicts quality on MOS scale. Calls with all relevant statistics are saved to MySQL or ODBC database. Optionally each call can be saved to pcap file with either only SIP / SKINNY protocol or...

...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

REPOSITORY MOVED TO GITHUB!! https://github.com/medooze/media-server Video Multiconference Media Server with WebRTC support. Provide Multiconference and video broadcasting services to any SIP service. Supports VP8, H264, MP4V-ES, H263 and H263P, continuous presence, RTMP flash broadcasting, adhoc conferences, load balancing and administrative WEB interface. JSR309 driver implementation under development. .

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

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.

IHU is a VoIP application for Linux (using Qt and Speex), with low latency, crypted stream, minimal use of bandwith, and without intermediary servers. It is the easiest way to talk real-time with your friends (like phone) on the internet or LAN.

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).

SpeexComm is a Speex based multiplatform voice communication engine meant mainly for gaming.

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.