webrtchacks

How to use the AWS API Gateway WebSocket API functionality with Lamdba functions to implement a serverless WebRTC signaling architecture

The post How to Leverage the AWS WebSocket API for Serverless WebRTC signaling appeared first on webrtcHacks.

Pion seemingly came out of nowhere to become one of the biggest and most active WebRTC communities. Pion is a Go-based set of WebRTC projects. Golang is an interesting language, but it is not among the most popular programming languages out there, so what is so special about Pion? Why are there so many developers […]

The post How Go-based Pion attracted WebRTC Mass – Q&A with Sean Dubois appeared first on webrtcHacks.

Interview with WebRTC standards co-chair and author, Bernard Aboba. We cover the current status of WebRTC and where it is headed including WebRTC-NV, Simulcast, SVC, AV1, WebTransport, WebCodecs, ML and more.

The post WebRTC Today & Tomorrow: Interview with W3C WebRTC Chair Bernard Aboba appeared first on webrtcHacks.

Walkthrough and deep analysis of how Azure Communications Service makes use of WebRTC by Gustavo Garcia

The post How does the new Azure Communication Services implement WebRTC? (Gustavo Garcia) appeared first on webrtcHacks.

Chrome recently added the option of adding redundancy to audio streams using the RED format as defined in RFC 2198, and Fippo wrote about the process and implementation in a previous article. You should catch-up on that post, but to summarize quickly RED works by adding redundant payloads with different timestamps in the same packet. […]

The post Implementing REDundant audio on an SFU appeared first on webrtcHacks.

Back in April 2020 a Citizenlab reported on Zoom’s rather weak encryption and stated that Zoom uses the SILK codec for audio. Sadly, the article did not contain the raw data to validate that and let me look at it further. Thankfully Natalie Silvanovich from Googles Project Zero helped me out using the Frida tracing […]

The post RED: Improving Audio Quality with Redundancy appeared first on webrtcHacks.

I wanted to add local recording to my own Jitsi Meet instance. The feature wasn’t built in the way I wanted, so I set out on a hack to build something simple. That lead me down the road to  discovering that: getDisplayMedia for screen capture has many quirks, mediaRecorder for media recording has some of its […]

The post Using getDisplayMedia for local recording with audio on Jitsi appeared first on webrtcHacks.

Software as a Service, Infrastructure as a Service, Platform as a Service, Communications Platform as a Service, Video Conferencing as a Service, but what about Gaming as a Service? There have been a few attempts at Cloud Gaming, most notably Google’s recently launched Stadia. Stadia is no stranger to WebRTC, but can others leverage WebRTC […]

The post Open Source Cloud Gaming with WebRTC appeared first on webrtcHacks.

A couple of weeks ago, the Chrome team announced an interesting Intent to Experiment on the blink-dev list about an API to do some custom processing on top of WebRTC. The intent comes with an explainer document written by Harald Alvestrand which shows the basic API usage. As I mentioned in my last post, this is the […]

The post True End-to-End Encryption with WebRTC Insertable Streams appeared first on webrtcHacks.

WebRTC has made getting and sending real time video streams (mostly) easy. The next step is doing something with them, and machine learning lets us have some fun with those streams. Last month I showed how to run Computer Vision (CV) locally in the browser. As I mentioned there, local is nice, but sometimes more performance […]

The post Accelerated Computer Vision inside a WebRTC Media Server with Intel OWT appeared first on webrtcHacks.

Time for another opinionated post. This time on… end-to-end encryption (e2ee). Zoom apparently claims it supports e2ee while it can not satisfy that promise. Is WebRTC any better? Zoom does not have End to End Encryption Let’s get to the bottom of things fast: Boo Zoom! I reviewed how Zoom’s implements their web client last […]

The post Does your video call have End-to-End Encryption? Probably not.. appeared first on webrtcHacks.

Don’t touch your face! To prevent the spread of disease, health bodies recommend not touching your face with unwashed hands. This is easier said than done if you are sitting in front of a computer for hours.  I wondered, is this a problem that can be solved with a browser? We have a number of […]

The post Stop touching your face using a browser and TensorFlow.js appeared first on webrtcHacks.

When most people think of WebRTC they think of video communications. Similarly, home surveillance is usually associated with video streaming. That’s why I was surprised to hear about a home security project that leverages WebRTC not for video streaming, but for the DataChannel. WebRTC’s DataChannel might not demo as well as a video call, but […]

The post Private Home Surveillance with the WebRTC DataChannel (Ivelin Ivanov) appeared first on webrtcHacks.

SDP has been a frequent topic, both here on webrtcHacks as well as in the discussion about the standard itself. Modifying the SDP in arcane ways is referred to as SDP munging. This post gives an introduction into what SDP munging is, why its done and why it should not be done. This is not […]

The post Not a Guide to SDP Munging appeared first on webrtcHacks.

Continue reading Min Duration Series – Part 1: Perfect Negotiation at webrtcHacks.

Continue reading Does Chromium-based Edge’s WebRTC Look Like Chrome? at webrtcHacks.

Review of Chrome's migration to WebRTC's Unified Plan, how false metrics may have misguided this effort, and what that means moving forward.

Continue reading Is everyone switching to Unified Plan? at webrtcHacks.

An analysis of the most popular open-source WebRTC repos on GitHub with a review of how WebRTC itself is doing there.

Continue reading and the WebRTC Open Source Popularity Contest Winner is… at webrtcHacks.

As you may have heard, Whatsapp discovered a security issue in their client which was actively exploited in the wild. The exploit did not require the target to pick up the call which is really scary.
Since there are not many facts to go on, lets do some tea reading…

The security advisory issued by Facebook says

A buffer overflow vulnerability in WhatsApp VOIP stack allowed remote code execution via specially crafted series of SRTCP packets sent to a target phone number.

Continue reading The WhatsApp RTCP exploit – what might have happened? at webrtcHacks.

When running WebRTC at scale, you end up hitting issues and frequent regressions. Being able to quickly identify what exactly broke is key to either preventing a regression from landing in Chrome Stable or adapting your own code to avoid the problem. Chrome’s bisect-builds.py tool makes this process much easier than you would suspect. Arne from appear.in gives you an example of how he used this to workaround an issue that came up recently.
{“editor”, “Philipp Hancke“}

In this post I am going to provide a blow-by-blow account of how a change to Chrome triggered a bug in appear.in and how we went about determining exactly what that change was.

Continue reading Bisecting Browser Bugs (Arne Georg Gisnås Gleditsch) at webrtcHacks.