WebRTC
With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, allowing developers to build powerful voice- and video-communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms. The technologies behind WebRTC are implemented as an open web standard and available as regular JavaScript APIs in all major browsers. For native clients, like Android and iOS applications, a library is available that provides the same functionality. The WebRTC project is open source and supported by Apple, Google, Microsoft and Mozilla, amongst others. This page is maintained by the Google WebRTC team.
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mediasoup client side C++ library
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Jun 1, 2023 - C++
Cutting Edge WebRTC Video Conferencing
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Sep 30, 2025 - C++
基于EasyRTC技术接入到EasyGBS平台的SDK(包含多种AMD/ARM硬件平台)及相关调用示例方法,通过EasyRTC协议接入EasyGBS能够充分利用WebRTC技术的领先优势,使得产品的整体体验上升一个层次。
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Telegram VoIP Contest (Bossy Gnu's submission )
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Oct 26, 2020 - C++
Open Source Communication Provider based on WebRTC and Cloud technologies
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Jul 31, 2017 - C++
A simple C++ broadcasting library fro audio coming from custom sources (e.g. generated by a program).
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Dec 22, 2024 - C++
Developing a distributed voice call app using WebRTC and Qt
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Feb 11, 2025 - C++
Created by Google
Released May 4, 2018
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