Show HN: Pion/handoff – Move WebRTC out of browser and into Go

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 viahandoff

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.

Pion has an active community on the Discord.

Follow the Pion Bluesky or Pion Twitter for project updates and important WebRTC news.

We are always looking to support your projects. Please reach out if you have something to build! If you need commercial support or don't want to use public methods you can contact us at [email protected]

Check out the contributing wiki to join the group of amazing people making this project possible

MIT License - see LICENSE for full text