Trending repositories for topic voip
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.
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
Baresip is a modular SIP User-Agent with audio and video support
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
DjangoPBX - A full-featured domain based multi-tenant PBX driven by Django and FreeSwitch.
A fully featured browser based WebRTC SIP phone for Asterisk
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
DjangoPBX - A full-featured domain based multi-tenant PBX driven by Django and FreeSwitch.
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
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 fully featured browser based WebRTC SIP phone for Asterisk
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
Baresip is a modular SIP User-Agent with audio and video support
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.
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
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.
Mumble is an open-source, low-latency, high quality voice chat software.
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
🚀starRTC,即时通讯(IM)系统,免费IM系统(含单聊,群聊,聊天室,文件传输),免费一对一视频聊天,VOIP,语音对讲(回音消除),直播连麦,视频直播,RTSP拉流,RTMP推流,webRTC服务端,在线教育,白板,小班课,在线会议,视频会议,视频监控,局域网直连(无需服务器),兼容webRTC, 支持webRTC加速,P2P高清传输,安卓、iOS、web互通,支持门禁对讲,可视对讲,电...
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
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.
DjangoPBX - A full-featured domain based multi-tenant PBX driven by Django and FreeSwitch.
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
:satellite: A curated list of awesome Real Time Communications resources
MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing.
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 Fork from http://git.freeswitch.org/git/freeswitch-contrib/tree/dvarnes/java/esl-client
A fully featured browser based WebRTC SIP phone for Asterisk
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.
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.
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing.
Baresip is a modular SIP User-Agent with audio and video support
Open source video conferencing app built on latest WebRTC SDK. Android/iOS/MacOS/Web
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
DjangoPBX - A full-featured domain based multi-tenant PBX driven by Django and FreeSwitch.
MagnusBilling is a fast, secure, efficient, high availability, VOIP Billing.
API server and Web GUI for FreeSwitch written in Golang and Angular
Open source video conferencing app built on latest WebRTC SDK. Android/iOS/MacOS/Web
:satellite: A curated list of awesome Real Time Communications resources
Main repository! MikoPBX - is free, easy to setup PBX for small business based on Asterisk 16 core
Pure python VoIP/SIP/RTP library. Currently supports PCMA, PCMU, and telephone-event
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
How I replaced Skype with Twilio to make phone calls from my computer
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 ...
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.
Open source video conferencing app built on latest WebRTC SDK. Android/iOS/MacOS/Web
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
Baresip is a modular SIP User-Agent with audio and video support
免费IM系统,IM即时通信消息系统(含一对一文字聊天,群聊,聊天室),免费一对一voip实时通话,录屏,webrtc服务端,免费直播连麦,互动直播,视频直播,RTSP拉流,RTMP推流,语音对讲,免费在线会议,视频会议等服务端程序,支持物联网平台,✨万水千山总是情,来个star行不行✨
Open source video conferencing app built on latest WebRTC SDK. Android/iOS/MacOS/Web
DjangoPBX - A full-featured domain based multi-tenant PBX driven by Django and FreeSwitch.
The only self-hostable discord music bot you will ever need, has a cool website too ✌️
A simple SIP server (proxy) for handling VoIP calls based on SIP using C++ on Windows & Linux platforms.
API server and Web GUI for FreeSwitch written in Golang and Angular
openduplex uses speech-to-text, artificial intelligence and text-to-speech, to call businesses and make appointments for you
Flutter Video SDK - Build your own video app experience using Dart, Flutter and the Stream Video Messaging API.
A utility to automate the installation, maintenance, and debugging of Asterisk/DAHDI, while integrating additional patches to provide the richest telephony experience
Pure python VoIP/SIP/RTP library. Currently supports PCMA, PCMU, and telephone-event
Android notification full screen incoming call for React Native
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.
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 ...