There are 56 repositories under voip topic.
即时通讯(IM)系统
Mumble is an open-source, low-latency, high quality voice chat software.
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
WebRTC plugin for Flutter Mobile/Desktop/Web
🚀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,电视盒子,树莓派,海思,全志,任天堂switch,云游戏,OTT设备,物联网平台,C语言自研方案,支持二次开发成类微信,类映客等APP,✨万水千山总是情,来个star行不行✨,更多示例请访问:
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 Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.
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✨
Async PHP client API for the telegram MTProto protocol
开源即时通讯(野火IM)系统Android端,支持Android 4.x —— 最新
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
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)
iOS CallKit framework and Android ConnectionService for React Native
SIPVicious OSS is a VoIP security testing toolset. It helps security teams, QA and developers test SIP-based VoIP systems and applications. This toolset is useful in simulating VoIP hacking attacks against PBX systems especially through identification, scanning, extension enumeration and password cracking.
Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of linphone-iphone (git://git.linphone.org/linphone-iphone.git)
:books: WebRTC (Web Real-Time Communications) 中文教程
Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP
Go implementation of the MediaDevices API.
A fully featured browser based WebRTC SIP phone for Asterisk