🏆 WebRTC - SFU - Simple, Secure, Scalable Real-Time Video Conferences Up to 8k, compatible with all browsers and platforms.

Pluggable WebRTC softphone and communication platform.

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, SaraPhone works with all WebRTC compliant servers: FreeSWITCH, Asterisk, OpenSIPS, Kamailio, etc. SaraPhone gets its name from Giovanni's wife, Sara.

一对一呼叫、邀请呼叫、音视频通话、多人通话，适合陌生人交友、在线教学、在线医疗、智能终端等场景；更能轻松实现腾讯会议，Zoom会议，钉钉课堂等多人音视频交互类场景。一对一音视频，一对多音视频。

Open-source event-driven AI powered Softphone

OpenTok demo application

HOMER 7 Docker Images

SIPREC recording server based on drachtio and rtpengine

React Native SIP App

sip capture server by hep。work with OpenSIPS, Kamailo, and FreeSWITCH。

GB28181 PS流转发网关服务<Node 版>，以GB28181对接的方式将摄像机/硬盘录像机 的PS流（H264/H265）打包推送到RTMP服务器。

📞 Giggle Jingle library for XMPP, implementation of XEP-0166.

React SIP user agent

Webrtc proxy server built using drachtio (SIP Proxy) and rtpengine (RTP)

Multitenant PBX

Multiparty-meeting (mediasoup) SIP gateway using Kurento

SIP provisioning server / Auto configuration system (ACS)

Node.js client library for the Civic Secure Identity Platform (SIP).