Search Results for "audio linux"
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Showing 57 open source projects for "audio linux"
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Amazon Connect connect-rtc-js
Provide softphone support to AmazonConnect customers
connect-rtc.js provides softphone support to AmazonConnect customers when they choose to directly integrate with AmazonConnect API and not use the AmazonConnect web application. It implements Amazon Connect WebRTC signaling protocol and integrates with browser WebRTC APIs to provide a simple contact session interface that can seamlessly integrate with Amazon Connect StreamJS. In a typical amazon-connect-streams integration, connect-rtc-js is not required on parent page. Softphone call... -
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jfPhone
VoIP/SIP SoftPhone (formerly known as jphonelite)
jfPhone is a VoIP/SIP SoftPhone for Desktops. -
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JavaComm
Desktop communication tool: chat, telephony, multimedia
JavaComm is a client–server communication platform offering chat, telephony, screen sharing, conferencing, a media player, and configurable media streams. Users can define and manage their own streams for flexible playback. All server-based communication runs over HTTPS (SSL/TLS), while file transfers are handled directly between users without passing through the server. The JavaComm Server runs on any Jakarta EE Web Profile 10 platform with the required extensions, using Tomcat 10.1.52 as... -
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VOIP-VOICE-TO-TEXT&ANALYS
Convert VoIP calls to text and analyze them with AI
The VoIP voice-to-text software for Issabel is an intelligent, AI-based solution that converts calls into accurate Persian text. After each call, the audio file is sent to the GPT-4O AI engine, producing editable transcripts. The software also provides AI-powered call analysis, extracting key points, customer requests, satisfaction levels, and sensitive topics, all stored in the database. This helps sales and support teams make faster decisions, improve response quality, and enhance customer... -
MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
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oreka
Enterprise telephony recording and retrieval system
Enterprise telephony recording and retrieval system with web based user interface. The project currently supports recording voice from VoIP SIP, Cisco Skinny (aka SCCP), raw RTP and audio sound device and runs on multiple operating systems and database systems. It can record audio from most PBX and telephony systems such as BroadWorks, Metaswitch, Asterisk, FreeSwitch, OpenSIPS, Avaya, Nortel, Mitel, Siemens, Cisco Call Manager, Cosmocom, NEC, etc... It is amongst others being used in Call... -
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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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JAVIER
JAvascript Voicexml InterpretER
JAVIER is a JAvascript Voicexml InterpretER, designed (but not restricted) to run inside a web browser, its main engine has less than 1000 lines of code. It's maybe, the tiniest but (almost) FULL VoiceXML implementation. -
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NoiseGator (Noise Gate)
A simple noise gate app intended for use with VOIPs like Skype.
Ever wanted to cut out background noise when talking with others on Skype? Now it's possible! NoiseGator is a light-weight noise gate application that routes audio through an audio input to an audio output. In real-time the audio level is analysed and if the average level is higher than the threshold the audio bypasses as normal. However, if the average level goes below the threshold, the gate closes and the audio is cut. When used with a virtual audio cable it can act as a noise gate... -
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mod_MumbleLink
Positional Audio Communication for Minecraft with Mumble
A Mod so that Minecraft now natively supports Mumble's positional audio feature. This means: Directional and positionally attenuated VOIP in relation to the game world. Please visit the Forum for information about the newest Version! Main Forum-Thread: http://www.minecraftforum.net/forums/mapping-and-modding/minecraft-mods/1272675 TheSkorm's Fork on GitHub: https://github.com/TheSkorm/mod_mumblelink Mumble: http://mumble.sourceforge.net Donations are greatly appreciated:... -
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GNU Telephony
GNU Telephony Services and Applications
GNU Telephony intends anyone to use free as in freedom software for telephony, and to do real-time collaboration freely and privately over the Internet, with the freedom to do so on any platform they choose to use. -
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baresip
Baresip is a modular SIP User-Agent with audio and video support
Baresip is a portable and modular SIP User-Agent with audio and video support. the latest source code can be found here: https://github.com/alfredh/baresip -
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Teamspeak 3 Love Plugin
A plugin for Teamspeak3. This plugin allows you to autofollow a user.
A plugin for Teamspeak3. This plugin allows you to follow a user while he switches through channels. For the love menu just right click any name in the server view. -
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O'Packet Platform is a platform which aims to offer a tool that users could operate network packet (Ethernet, IP, TCP, UDP, SCTP, RTP... ) and application packet like audio/vieo packet and DTMF packet ( 2833), which refered to all rfc protocol.
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Zanzibar is a complete, standards based IVR. It includes an MRCPv2 Server with ASR and TTS engines as well as an voiceXML interpreter so that you can deploy and run voiceXML applications. It integrates with VOIP PBX’s (like Asterisk) using SIP and RTP.
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A JNI wrapper for pjsip. You can use this wrapper to develop Java applications using the pjsip library. At the moment only the pjsua API is implemented. If you would like to obtain a commercial license, or need customisations, please contact us.
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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
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Cairo sets out to provide an enterprise grade, MRCPv2 compliant speech solution utilizing existing open source speech resources such as FreeTTS and Sphinx-4.
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Use this when attachments aren't offered with your email service. (i.e. Proxy Mail). Simply Supply a link to download the 11KB Audio Player and Copy/Paste your Message into the Email. See Website For Demo. http://audiotext.sourceforge.net
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DTMF detector library and/or application that reads in the specified audio file and returns/outputs whether DTMF detected and/or list of detected digits. May have GUI and command line interfaces.
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Rabboni VOIP is an Open Source Cross Platform Softphone developed with trolltech QT LIBRARY it uses PJSIP - Open Source SIP Stack
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OpenAL mixer aims to provide a multiplatform API built on top of OpenAL to get/set volume values (including support for capture devices) and list, select and mute input and output lines.
