Open-source VoIP videoconference

Multi-party SIP and WebRTC video conference on Asterisk 20 (ConfBridge SFU), nginx and SIP.js. Semestral project for Voice over IP I at VŠB-TU Ostrava, by Julian Lara (lar0053).

1. WebRTC StudioRegister an account, join room 1000 and see every participant. Works with a simulated camera, so three tabs are enough. 2. DocumentationArchitecture, configuration listings, SIP/SDP flows, bandwidth analysis and results. 3. PDF reportThe full report in English (22 pages).

Type	Accounts (user / password)
WebRTC	3041/3041ab, 3042/3042ab, 3043/3043ab, 3044/3044ab, Bolek/bolek123, Lolek/lolek123
Classic SIP	3001/3001ab, 3002/3002ab, 3003/3003ab
Extensions	1000 SFU room, 2000 follow-talker room, 9999 echo, 3xxx direct

Demo credentials on a public server. Signalling: wss://web-conference-lar0053.duckdns.org/ws