Search Results for "audio linux"
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Showing 35 open source projects for "audio linux"
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jfPhone
VoIP/SIP SoftPhone (formerly known as jphonelite)
jfPhone is a VoIP/SIP SoftPhone for Desktops. -
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Mumble
Low-latency, high quality voice chat for gamers
Mumble is an open source, low-latency, high quality voice chat software primarily intended for use while gaming. It includes game linking, so voice from other players comes from the direction of their characters, and has echo cancellation so the sound from your loudspeakers won't be audible to other players. -
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Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.
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app_swift provides an interface into the Cepstral TTS libraries to stream text-to-speech audio from the Asterisk PBX without creating files.
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Simple testing tool to generate RTP data packets and send it via netwok interface or save into pcap file. Primarily intended for use with SIPp application to test speech quality with different codecs.
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A Sip live audio feeding agent. The agent captures audio from sound card and sends live audio stream(uLaw) to caller(sip phone) using RTP. It is based on Peers 0.3(http://peers.sourceforge.net/). Can be used in IP telephony to broadcast live audio.
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CHAROZT is an Asterisk MOH (Music on Hold) Uploader. It is a Linux-based application installed in Linux desktops that converts MP3 to WAV for Asterisk supported MOH. It can also upload converted files to the Asterisk Server.
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Kiax is a softphone (soft phone, VoIP client) with a simple and comfortable user interface for making VoIP calls to Asterisk PBX. It depends on the iaxclient library to use Asterisk's IAX2 protocol for easy call configuration and audio setup.
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A lightweight cross platform IP telephony client using the IAX protocol, designed for use with the asterisk open source PBX.
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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.
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Scx-tools is a set of utilities for Skype VoIP client. The project goal is to provide several useful tools for such task as Skype calls recording and management. Currently only the call recorder (version 0.1.1) is available.
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vox-attendant is a telephony voice enabled virtual receptionist designed to work with VoiceXML and includes a distribution for the Voxeo Prophecy platform. It provides a web based interface for managing the directory.
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libfindrtp is a C library providing functions used to identify Real-time Transport Protocol (RTP) session endpoint ports and network addresses from active VoIP signaling traffic.
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Enable Linux firewall to support connection tracking and NAT of H.323 protocol. It supports RAS, Fast Start, H.245 Tunnelling, Call Forwarding, Signal Proxy/Softswitch, RTP/RTCP and T.120 based audio, video, fax, chat, whiteboard, file transfer, etc.
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The man goal of Dr.Ta/MOD project is to create a brand new communication in vehicle. Under multi-network environment, Dr.ta/Mod use SCTP multi-homing feature to process multi-path association. Please read more in our website http://mod.maple.tw
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SIPNet is a p2p-sip framework, it support IM,audio,video based on p2p-sip protocols, and support file sharing like other p2p programs such as bittorrent. All SIPNet clients join to form a P2P overlay net and find each other using DHT algorithm.
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Priax is a small graphical multiplatform (Linux and win32) IAX2 client written in C, using libiaxclient and SDL.
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GibPhone is a highly extensible VoIP/IM client for the .NET framework that uses a powerful plugin engine to allow for UI extensions and any call control stack / media payload / transport protocol combination.
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Audio/Video Conference software for Instant Messengers. It aims to provide Audio/Video conferencing for as many Instant Messengers as possible through a modular design.
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Javafon is an open source peer-to-peer audio-communication application written in Java. The audio signal is compressed with JSpeex. Furthermore it supports encryption through AES and Blowfish.
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Speech profile of Person is created that contains elementary sounds uttered. Profile is 1 time download for listeners. The actual audio sample is encoded based on profile. Decode using the profile stored earlier by User, and the audio can be regenerated.
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A library to provide team-based voice communication over the net, meant for easy integration with multiplayer games.
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Simple command line IP telephonly application for Linux, that doesn't conform to either H.323 or SIP, doesn't use RTP and is not interoperable with any other program. It does work well though, and has quite good audio quality.
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Speak Freely is a Cross Platform Internet telephony (Voice Chat) application which provides high quality voice grade audio with GSM and CELP compression and encryption with DES, Blowfish, and IDEA ciphers. Refering to recent news: The maintainers will never add any unwanted extra software in the zip download file but we can only speak for our selves.
