Search Results for "jack audio voip"
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Showing 52 open source projects for "jack audio voip"
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VoIP monitor
VoIP SIP and SKINNY quality analyzer and packet / audio recording tool
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 SIP/RTP... -
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jfPhone
VoIP/SIP SoftPhone (formerly known as jphonelite)
jfPhone is a VoIP/SIP SoftPhone for Desktops. -
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cerberuscms2
Cerberus Content Management System
Cerberus Content Management System is a dynamic, secure and infinitely expandable CMS designed after a Unix-Like model. It is a custom written Web Application Framework ( W.A.F. ) with a consistent and custom written Pre-Hyper-Text-Post-Processor Programming Code Framework ( P.C.F. ). This Web Application Software Project' aim is to be the fastest and most secure Web Application Framework, Web Application Programming Code Framework, Text, Voice and Video Communications Platform and Content... -
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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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ViaVoip
A portable peer to peer voice-chat/walkie-talkie.
ViaVoip is a simple Voice Over IP application that can be used when you need to talk, chat, or send files through the internet, but you can't or don't want to make use of any third party services. Its peer to peer design allows the two end points to connect directly to each other, without any central server nor account registration. It runs on Windows, Linux, Mac OS X and Android, and is portable, that is you don't need any setup, just get a copy and run it from any storage... -
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A Python library to create sophisticated multilingual IVR applications. NOTICE. The repository is frozen, please find the latest version of the software at https://github.com/sippy/vapp
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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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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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Irdroid - Infrared Remote for Android
The Irdroid app / module is a infrared remote control for Android
Almost everyone uses infrared remote controls at home and developing and using infrared on most of the Android Devices requires the usage of external hardware. You will learn about the design details, the tools and software used to design a project that enables any Android device to control infrared equipment. You will get an introduction to the hardware functionality of Irdroid v1.0 modules, battery powered infrared remote control modules that connects to the Android device Audio jack... -
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Libre Software to manage audio/video connection from a PC to a libre hardware real time stethoscope hardware, developed to allow doctors make a remote diagnose in real time in isolated rural areas of developing countries or for medicine teaching
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ChatAnt
client-server full java chat audio-video file sharing
client-server full-java chat, provided with audio&video (JMF), custom rooms, custom forums, instant- messaging and file sharing -
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This is a P2P VoIP system. It support audio and video chat through kademlia-based dht overlay. The p2p module use Kademlia protocols; Audio codec is G.729. An simple Introduction here: http://www.needdo.com/2007/09/ppphonekademliap2p-voip/ . My Personal blog is http://www.idolf.cn and my email is dolfcao@gmail.com .
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OfficeSIP Softphone and Messenger are two enterprise VoIP SIP clients written in C# in .NET Framework. The SIP clients make use of Microsoft UCC API SDK, ensuring the highest quality of audio and video. Compatible with Office Communications Server. See also open source, cross-platform: 1) simple messenger Brief Msg at http://briefmsg.org 2) MUVConf is a multi-user video conferencing, see demo video http://youtu.be/YrBU-Aqtvrk, download https://code.google.com/p/muvconf/downloads/list
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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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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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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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This project is a SIP based softphone developed as a Java applet that can be embedded into a website. The applet has a simple user interface and is intended to be used by journalists to make and deposit audio recordings via a SIP proxy.
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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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