Search Results for "audio"

...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/RTCP/T.38/udptl protocols. VoIPmonitor can also decode audio.

...It provides enterprise-grade VoIP communication with a focus on performance, security, and extensibility. Features Ultra-Fast: Startup in <0.5s, minimal CPU usage HD Audio: Opus and G.711 codecs with adaptive jitter buffer Security: TLS transport, SDES-SRTP encryption, Windows DPAPI credential protection Multi-Account: Up to 5 simultaneous SIP accounts Plugin SDK: C# SDK for custom integrations (CRM, call recording, automation) Plugin Marketplace: Community and premium extensions Headset Support: Jabra, Poly, Sennheiser, Logitech with HID controls Call Quality: Real-time MOS Score, Jitter, Packet Loss and RTT monitoring Enterprise: Silent MSI installer, GPO support, JSON centralized config Compatibility: 3CX, Asterisk, FreePBX, OVH, Twilio, RingCentral, and more Language packages : https://sipline.feelautom.fr/languages

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 experience. ...

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.

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

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 for a either a sound input(microphone) or sound output(speakers). ...

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: https://sourceforge.net/donate/?...

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

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.

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

client-server full-java chat, provided with audio&video (JMF), custom rooms, custom forums, instant- messaging and file sharing

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

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.

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.

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

app_swift provides an interface into the Cepstral TTS libraries to stream text-to-speech audio from the Asterisk PBX without creating files.

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

Fideliphone: High Quality Internet Telephony

PJSGUA is an Open Source SIP UAC based on PJSUA reference UAC from PJSIP project

PHP based Viewer for Voice Servers like Mumble.

Open Unified Recording (OUR) is a full featured Linux based Open Source VoIP/SIP Call Recording engine, indexing and retrieval system. The system resides on the network and passively captures SIP sessions. Project is sponsored by UnifiedRecording.com

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.

Apreta is about sharing information in real-time meetings. It provides desktop sharing, presentation sharing, a shared whiteboard, and audio conferencing.

Vorilon is a Voice Teleconferencing software consisting of a Client and Server applications to be used for Game Voice chatting and General Voice Teleconferencing.

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.