chrome://webrtc-internals is a tool used in Chrome browser to diagnose WebRTC behavior, which can help troubleshoot problems such as video call delays or camera failures. After opening this page, multiple PeerConnection entries will be displayed, each containing ICE candidates, SDP negotiations and data flow statistics; click "Stats" to view real-time performance metrics, such as bytesSent/received, iceConnectionState, networkType, jitter and packetLoss; you can also export a complete data report in JSON format by clicking "Download the report" at the top, which is convenient for developers or support teams to further analyze; when using it, you should pay attention to the key fields, confirm that the correct PeerConnection is selected, and ensure that the website you are visiting does use WebRTC technology.
If you've ever wondered how WebRTC is being in your browser—like why a video call is lagging or why your camera isn't showing up—the chrome://webrtc-internals
page can give you a lot of useful insight. It's basically a diagnostic dashboard for all things WebRTC in Chrome.

What You'll See on chrome://webrtc-internals
When you open this page, it looks pretty technical and maybe a bit overwhelming at first. But once you get the hang of it, it becomes much more readable.
- PeerConnection sections show each connection your browser has made using WebRTC.
- Each section includes details like ICE candidates, SDP offers/answers, and stats about data flow.
- There's also a "Stats" tab that gives real-time performance metrics—like bandwidth usage, packet loss, and jitter.
If you're troubleshooting a call or a live stream, this is where you want to look.

How to Monitor an Active Call
Once you're in a WebRTC-based app (like Google Meet, Zoom, or any custom WebRTC app), just open a new tab and type chrome://webrtc-internals
. You should see one or more PeerConnection entries appear.
Here's what to do next:

- Click on the PeerConnection entry that corresponds to your current session.
- Look for the “Stats” button and click it to see live updates.
- Pay attention to key metrics:
-
bytesSent
/bytesReceived
: tells you if media is flowing both ways. -
iceConnectionState
: shows if ICE negotiation succeeded or failed. -
networkType
: lets you know if you're on Wi-Fi, Ethernet, etc. -
jitter
andpacketLoss
: important indicators of network quality.
-
This is especially helpful if someone's having trouble hearing or seeing you—it might be a network issue rather than a camera/mic problem.
When to Export Data for Analysis
One underused feature is the ability to export raw data from a session.
At the top of the page, there's a link labeled “Download the report” . This will save a JSON file containing all the collected stats and metadata for every PeerConnection.
Why would you want that?
- Share it with developers or support teams who can dig deeper into connectivity problems.
- Compare reports from different sessions to spot patterns (eg, always high jitter when on mobile data).
- Use tools like WebRTC Leak or other analyzers to visualize the data.
It's not something most users need daily, but it can be a lifesaver when debugging serious issues.
A Few Tips to Keep in Mind
- Don't panic if you don't understand everything—you only need to focus on a few key fields.
- Make sure you're looking at the right PeerConnection if multiple are listed.
- Some apps reuse connections, so refreshing the page or rejoining a call might help reset things.
- If nothing shows up, double-check that your site is actually using WebRTC (not all video apps do).
You don't need to be a developer to make good use of this tool, but a little familiarity with networking basics helps.
Basically that's it.
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