Skip to content
KitploitKITPLOIT
ToolsBlog
Submit
ToolsBlog
Submit

Hacking, PenTest, and Cybersecurity Tools for Your Security Arsenal!

Kitploit is a directory of hacking, cybersecurity, and pentesting tools. Discover the latest project updates to find vulnerabilities, analyze systems, automate testing, and strengthen your security.

··Feeds·Contact·Privacy·© 2026 Kitploit

Tool Directory

Categories

View all categories
Loading categories
handoff — Start WebRTC in the browser—run it somewhere else | Kitploit
Tools/GitHubGitHub/pion/handoff
Packet Sniffing & AnalysisReverse EngineeringScripting & AutomationWeb SecurityUtilities & Frameworks
GitHubpion/handoff

handoff

Start WebRTC in the browser—run it somewhere else

View Repository
18854 months agoReviewed by Kitploit

Most Popular

View all →

Discover the most used tools by our community.

Explore all tools

Browse our collection of tools

View all tools →
Share


Pion Handoff

Create WebRTC session in the browser—run it somewhere else

join us on Discord Follow us on Bluesky Twitter Widget


Why

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
root@kitploit:~
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

Usage

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.

Example

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.

Example

Community

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]

Contributing

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

License

MIT License - see LICENSE for full text

Download Tool