voip
There are 764 repositories under voip topic.
pion/webrtc
Pure Go implementation of the WebRTC API
wildfirechat/im-server
即时通讯(IM)系统
fonoster/fonoster
🚀 The open-source alternative to Twilio.
mumble-voip/mumble
Mumble is an open-source, low-latency, high quality voice chat software.
processone/ejabberd
Robust, Ubiquitous and Massively Scalable Messaging Platform (XMPP, MQTT, SIP Server)
qTox/qTox
qTox is a chat, voice, video, and file transfer IM client using the encrypted peer-to-peer Tox protocol.
flutter-webrtc/flutter-webrtc
WebRTC plugin for Flutter Mobile/Desktop/Web
webrtc-rs/webrtc
A pure Rust implementation of WebRTC
starrtc/starrtc-android-demo
🚀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：
signalwire/freeswitch
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.
starrtc/starrtc-server
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✨
danog/MadelineProto
Async PHP client API for the telegram MTProto protocol
wildfirechat/android-chat
即时通讯，聊天，野火IMAndroid客户端，支持Android 4.x —— 最新
kamailio/kamailio
Kamailio - The Open Source SIP Server for large VoIP and real-time communication platforms -
asterisk/asterisk
The official Asterisk Project repository.
miroslavpejic85/mirotalksfu
🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 4k, compatible with all browsers and platforms.
pjsip/pjproject
PJSIP project
onsip/SIP.js
A simple, intuitive, and powerful JavaScript signaling library
baresip/baresip
Baresip is a modular SIP User-Agent with audio and video support
sipcapture/homer
HOMER - 100% Open-Source SIP, VoIP, RTC Packet Capture & Monitoring
sipsorcery-org/sipsorcery
A WebRTC, SIP and VoIP library for C# and .NET. Designed for real-time communications apps.
fonoster/routr
⚡ The future of programmable SIP servers.
wildfirechat/ios-chat
即时通讯，聊天，野火IM iOS客户端
BelledonneCommunications/linphone-android
Linphone.org mirror for linphone-android (https://gitlab.linphone.org/BC/public/linphone-android)
64characters/Telephone
SIP softphone for Mac
react-native-webrtc/react-native-callkeep
iOS CallKit framework and Android ConnectionService for React Native
EnableSecurity/sipvicious
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.
pion/stun
A Go implementation of STUN
BelledonneCommunications/linphone-iphone
Linphone is a free VoIP and video softphone based on the SIP protocol. Mirror of linphone-iphone (git://git.linphone.org/linphone-iphone.git)
Tinywan/WebRTC-tutorial
:books: WebRTC (Web Real-Time Communications) 中文教程
element-hq/element-call
Group calls powered by Matrix
pion/mediadevices
Go implementation of the MediaDevices API.
creytiv/re
Generic library for real-time communications with async IO support
tiredofit/docker-freepbx
Dockerized FreePBX 15 w/Asterisk 17, Seperate MySQL Database support, and Data Persistence and UCP
MarshalX/tgcalls
Voice chats, private incoming and outgoing calls in Telegram for Developers
InnovateAsterisk/Browser-Phone
A fully featured browser based WebRTC SIP phone for Asterisk