# ddssingsong/webrtc_android

webrtc VideoCall VideoConference  视频通话 视频会议

Repository: https://github.com/ddssingsong/webrtc_android
Canonical: https://ross.abutalabs.com/products/webrtc_android
Homepage: https://blog.csdn.net/u011077027/article/details/86225524
Language: Java
License: MIT
License Family: permissive
Topics: webrtc-android, webrtc, videochat, video, p2p, videocall, webrtc-demos
Last push: 2025-01-09T05:55:34+00:00

## Health v2 (maintenance only)
Score: 23/100 (v2, computed 2026-09-02T17:46:02.011165+00:00)
- activity 0, release rhythm 8, longevity 100
- inputs: {"age_days": 2800, "days_push": 601, "days_rel": null, "gap_med": null, "n_releases_24m": 0}
- flags: none
- formula: round(0.45*activity + 0.35*rhythm + 0.20*longevity); archived -> min(score, 10)

## Adoption (not part of the score)
Stars 1722, forks 547 (observed 2026-08-28T04:05:27.719653+00:00)

## What it is
An Android demo application built on WebRTC supporting one-to-one voice/video calls and many-to-many video conferencing, with a companion Java signaling server. It is primarily intended as a learning project to understand the full WebRTC call flow.

## Use cases
- make one-to-one video calls on android
- build a multi-party video conference app
- learn how webrtc signaling and calls work
- demo p2p voice and video chat on mobile
- integrate webrtc sdk into an android app
- set up stun/turn with a signaling server for calls

## When to choose
- you want a learning-oriented WebRTC Android example with full call flow
- you need a reference for one-to-one and multi-party calls on Android
- you want to experiment with configurable signaling and STUN/TURN servers

## When to avoid
- you need a production-ready, fully supported video conferencing product
- you require reconnection after disconnection or beauty filters out of the box
- you target platforms other than Android

## Facets
- artifact type: application
- maturity: active
- function: websocket, http-client, gui
- domain: mobile-development, web-development
- platform: jvm
- tags: webrtc, video-call, video-conference, p2p, signaling, android-demo, learning-project, real-time, android

## Member repositories
- ddssingsong/webrtc_android (main) score 23

## Provenance
- Observed fields: from GitHub, fetched 2026-08-28T04:05:27.719653+00:00.
- Health v2: computed from the inputs above; adoption is never an input.
- Inferred fields (summary, facets, guidance): AI-extracted, prompt v1, taxonomy v1, on 2026-08-30T03:33:59.462006+00:00, confidence not recorded.
  - readme: https://github.com/ddssingsong/webrtc_android (fetched 2026-08-28T04:05:27.719653+00:00, sha 411913018fdf)
- Data as of 2026-08-30T08:39:29.467469+00:00.