bytebuffer-voip
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call-p2p-backend
call-p2p-backend PublicWebRTC P2P Signaling Server viết bằng Rust (Axum + Tokio). Xử lý toàn bộ luồng tín hiệu cho cuộc gọi voice/video trực tiếp giữa 2 người dùng, bao gồm trao đổi SDP/ICE, quản lý trạng thái cuộc gọi t…
Rust 2
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janus-call-service
janus-call-service PublicJanus Call Service — A WebRTC signaling server built with Rust/Axum that integrates with Janus Gateway for peer-to-peer calls. Features WebSocket-based call signaling (SDP/ICE exchange), JWT authen…
Rust
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call-p2p-web
call-p2p-web PublicVue 3 + Vite web softphone, WebRTC Peer to Peer, UI built with Naive UI and Pinia.
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b2bua-web
b2bua-web PublicA Vue 3 reference implementation of a browser-based softphone for a B2BUA (Back-to-Back User Agent) call architecture. The client speaks WebRTC for media and a lightweight WebSocket protocol for ca…
Vue
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simulator-telco
simulator-telco PublicSIP/telco simulator lab using Kamailio 5.6 with MySQL-backed auth & user location, plus prefix-based dispatcher routing and failover. Dockerized for quick spin-up.
Repositories
- janus-call-service Public
Janus Call Service — A WebRTC signaling server built with Rust/Axum that integrates with Janus Gateway for peer-to-peer calls. Features WebSocket-based call signaling (SDP/ICE exchange), JWT authentication, MongoDB user management, and an actor-based call flow supervisor.
- simulator-telco Public
SIP/telco simulator lab using Kamailio 5.6 with MySQL-backed auth & user location, plus prefix-based dispatcher routing and failover. Dockerized for quick spin-up.
- b2bua-web Public
A Vue 3 reference implementation of a browser-based softphone for a B2BUA (Back-to-Back User Agent) call architecture. The client speaks WebRTC for media and a lightweight WebSocket protocol for call signaling, and is designed to plug into a B2BUA backend that bridges the browser leg to a PSTN/SIP trunk.
- call-p2p-web Public
Vue 3 + Vite web softphone, WebRTC Peer to Peer, UI built with Naive UI and Pinia.
- call-p2p-backend Public
WebRTC P2P Signaling Server viết bằng Rust (Axum + Tokio). Xử lý toàn bộ luồng tín hiệu cho cuộc gọi voice/video trực tiếp giữa 2 người dùng, bao gồm trao đổi SDP/ICE, quản lý trạng thái cuộc gọi theo mô hình Actor, xác thực JWT và hỗ trợ đa thiết bị qua WebSocket.
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