Trending repositories for topic voip
Mumble is an open-source, low-latency, high quality voice chat software.
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
Baresip is a modular SIP User-Agent with audio and video support
Main repository! MikoPBX - is free, easy to setup PBX for small business based on Asterisk 16 core
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
Main repository! MikoPBX - is free, easy to setup PBX for small business based on Asterisk 16 core
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
Mumble is an open-source, low-latency, high quality voice chat software.
Baresip is a modular SIP User-Agent with audio and video support
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
Mumble is an open-source, low-latency, high quality voice chat software.
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
Baresip is a modular SIP User-Agent with audio and video support
Main repository! MikoPBX - is free, easy to setup PBX for small business based on Asterisk 16 core
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
Main repository! MikoPBX - is free, easy to setup PBX for small business based on Asterisk 16 core
Mumble is an open-source, low-latency, high quality voice chat software.
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
Baresip is a modular SIP User-Agent with audio and video support
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
A fully featured browser based WebRTC SIP phone for Asterisk
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
Mumble is an open-source, low-latency, high quality voice chat software.
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing.
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
qTox is a chat, voice, video, and file transfer IM client using the encrypted peer-to-peer Tox protocol.
MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing.
API server and Web GUI for FreeSwitch written in Golang and Angular
Main repository! MikoPBX - is free, easy to setup PBX for small business based on Asterisk 16 core
Dockerized FreePBX 16 with Asterisk 16, PHP 7.4, dedicated MySQL database, Hashicorp Vault integration and data persistence
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
Android app for Mumble voice conference system. — NOTE: mirror of https://gitlab.com/quite/mumla Please file issues and merge requests over there. (Mirror of the Humla library here: https://github.com...
How I replaced Skype with Twilio to make phone calls from my computer
A fully featured browser based WebRTC SIP phone for Asterisk
SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. SaraPhone is fully integrated with FusionPBX. Based on SIP.js, Sar...
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a ...
Mumble is an open-source, low-latency, high quality voice chat software.
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Open source video conferencing app built on latest WebRTC SDK. Android/iOS/MacOS/Web
Baresip is a modular SIP User-Agent with audio and video support
qTox is a chat, voice, video, and file transfer IM client using the encrypted peer-to-peer Tox protocol.
Dockerized FreePBX 16 with Asterisk 16, PHP 7.4, dedicated MySQL database, Hashicorp Vault integration and data persistence
Open source video conferencing app built on latest WebRTC SDK. Android/iOS/MacOS/Web
API server and Web GUI for FreeSwitch written in Golang and Angular
The only self-hostable discord music bot you will ever need, has a cool website too ✌️
MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing.
Flutter plugin of Waterbus. Build video call or online meeting app with SFU. Supports Android/iOS/MacOS/Web
A simple SIP server (proxy) for handling VoIP calls based on SIP using C++ on Windows & Linux platforms.
The Open Source Contact Center Solution (mirror of https://gitlab.com/omnileads/ominicontacto)
Flutter Video SDK - Build your own video app experience using Dart, Flutter and the Stream Video Messaging API.
openduplex uses speech-to-text, artificial intelligence and text-to-speech, to call businesses and make appointments for you
Pure python VoIP/SIP/RTP library. Currently supports PCMA, PCMU, and telephone-event
Minimalist Windows / OSx / Linux SIP Softphone with API Control
A utility to automate the installation, maintenance, and debugging of Asterisk/DAHDI, while integrating additional patches to provide the richest telephony experience
