WebRTC is the real-time communication technology used for real-time media streaming. Used for things like Google Meet, Discord and Zoom on the web. With Handoff you create your WebRTC session in the browser, but then move it to a process you control. This lets you do a few interesting things.

• Record - Join the Zoom call via handoff and save media as it passes through.
• Send - Use FFmpeg or send an external source. Not limited by browser quality/capture code.
• Reverse Engineer - Capture ICE/DTLS and decrypted RTP/RTCP/SCTP traffic

sequenceDiagram
    participant Handoff
    participant Browser
    participant Website

    Browser->>Website: Authenticate and use the site normally
    Website->>Browser: WebRTC Signaling
    Note over Browser: WebRTC API is mocked
    Browser->>Handoff: Forward Signaling
    Handoff->>Website: Establish WebRTC session

See examples directory. examples/datachannel shows a normal page with an optional override, examples/media-save saves VP8 video on the backend while still showing it in the browser, examples/media-send forwards VP8 RTP from the backend into the browser, and examples/greasemonkey generates a userscript that overrides RTCPeerConnection automatically.

Typically you will install the greasemonkey script and then run one of the examples.

Below is an example of sending a users webcam to a WebRTC service, but replacing outgoing video with a ffmpeg testsrc. Handoff sits between the users so it can replace with any arbitrary video.

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MIT License - see LICENSE for full text