webrtcHacks

Guides and information for WebRTC developers

Post List

Guide

A playground for Simulcast without an SFU
Accelerated Computer Vision inside a WebRTC Media Server with Intel OWT
AIY Vision Kit Part 1: TensorFlow Computer Vision on a Raspberry Pi Zero
Am I behind a Symmetric NAT?
Anatomy of a WebRTC SDP (Antón Román)
Bisecting Browser Bugs (Arne Georg Gisnås Gleditsch)
Build your own phone company with WebRTC and a weekend
Calculating True End-to-End RTT (Balázs Kreith)
Computer Vision on the Web with WebRTC and TensorFlow
Data Nerding with WebRTC GitHub Data
Dealing with HTMLMediaElements and srcObjects in WebRTC applications
Debugging VP8 is more fun than it used to be
Developing mobile WebRTC hybrid applications
Does Amazon’s Mayday use WebRTC? A Wireshark analysis
Does Chromium-based Edge’s WebRTC Look Like Chrome?
First steps with ORTC
First steps with QUIC DataChannels
Fix Bad Lighting with JavaScript Webcam Exposure Controls (Sebastian Schmid)
Gaming with the WebRTC DataChannel – A Walkthrough with Arin Sime
getUserMedia – What happens when there’s missing media sources?
getUserMedia Mirrors and Frame Rates
getUserMedia resolutions III – constraints unleashed
Guide to WebRTC with Safari in the Wild (Chad Phillips)
How Janus Battled libFuzzer and Won (Alessandro Toppi)
How to add virtual background transparency in WebRTC
How to Build a Motion Detecting Baby Monitor with WebRTC
How to capture & replay WebRTC video streams with video_replay (Stian Selnes)
How to Figure Out WebRTC Camera Resolutions
How to Leverage the AWS WebSocket API for Serverless WebRTC signaling
How to limit WebRTC bandwidth by modifying the SDP
How to stop a leak – the WebRTC notifier
How to Train a Dog with JavaScript
How to use WebRTC and Chrome Extensions to Call a Browser When it is Not Open (Konstantin Goncharuk)
Introducing the WebRTC Developer Tool Vendor Directory
Kurento.org: WebRTC, Computer Vision, Augmented Reality, Awesome (Luis López Fernández)
Let’s Encrypt – how get to free SSL for WebRTC
Let’s get better at fuzzing in 2019 – here’s how
Making Zoom’s Smart Gallery on the Web with MediaPipe and BreakoutBox
Optimizing WebRTC Power Consumption (Markus Handell)
Part 2: Building a AIY Vision Kit Web Server with UV4L
Private Home Surveillance with the WebRTC DataChannel (Ivelin Ivanov)
Put in a Bug in Apple’s Apple – Alex Gouaillard’s Plan
Reacting to React Native for native WebRTC apps (Alexey Aylarov)
Real-Time Video Processing with WebCodecs and Streams: Processing Pipelines (Part 1)
Reeling in Safari on WebRTC – A Closer Look at What’s Supported
Sample code for a WebRTC feature? webrtc-experiment’s got that – Q&A with Muaz Khan
Sharpening the Edge – extended Q&A with Microsoft for RTC devs
Shut up! Monitoring audio volume in getUserMedia
Smile, You’re on WebRTC – Using ML Kit for Smile Detection
Stop touching your face using a browser and TensorFlow.js
Surviving Mandatory HTTPS in Chrome (Xander Dumaine)
The Big Churn – learning from real usage stats (Lasse Lumiaho and Varun Singh)
The Minimum Viable SDP
The new Android M App Permissions (Dag-Inge Aas)
The Ultimate Guide to Jitsi Meet and JaaS
The WebRTC Troubleshooter: test.webrtc.org
Troubleshooting Unwitting Browser Experiments (Al Brooks)
True End-to-End Encryption with WebRTC Insertable Streams
Update: Anatomy of a WebRTC SDP (Antón Román)
Using getDisplayMedia for local recording with audio on Jitsi
VR Video Calling with WebRTC and WebVR (Dan Jenkins)
Ways to save an image from your webcam in 2022
WebRTC Externals – the cross-browser WebRTC debug extension
WebRTC media servers in the Cloud: lessons learned (Luis López Fernández)
WebRTC Video Resolutions 2 – the Constraints Fight Back
Your Browser as a Audio Conference Server with WebRTC & Web Audio (Alexey Aylarov)

Other

and the WebRTC Open Source Popularity Contest Winner is…
Exciting Developer Track at Upcoming WebRTC Conference in Santa Clara
Help Typhoon Victims in the Philippines
How is WebRTC doing and who is driving usage? (Hint: Google Meet)
In case you missed it… webrtcHacks after 4 months
Introducing the WebRTC Event Directory
Looks-matter – the new webrtcHacks design
Post-Peak WebRTC Developer Trends: An Open Source Analysis
Strong WebRTC Signals at the IIT Real-Time Communications Conference
Survey results: and the WebRTC developers say…
The new webrtcHacks team member you already know!
WebRTC developers – we need 60 seconds
WebRTC Development Tools – Where to Begin
webrtcHacks Meetup at Mobile World Congress (MWC)
Welcome to webrtcHacks – a blog for WebRTC makers

Reverse-Engineering

All I want for Christmas is Hangouts to use WebRTC on Firefox
Apple’s not so private relay fails with WebRTC
Dear NY Times, if you’re going to hack people, at least do it cleanly!
Dear Slack: why is your WebRTC so weak?
Facebook Messenger likes WebRTC
Facetime doesn’t face WebRTC
FaceTime finally faces WebRTC – implementation deep dive
Finding the Warts in WebAssembly+WebRTC
Hello Chrome and Firefox, this is Edge calling
Hello, Hello – What’s your real story? A decode by Philipp Hancke
How Cloudflare Glares at WebRTC with WHIP and WHEP
How does Hangouts use WebRTC? webrtc-internals analysis
How does the new Azure Communication Services implement WebRTC? (Gustavo Garcia)
How Zoom’s web client avoids using WebRTC (DataChannel Update)
Is Slack’s WebRTC Really Slacking? (Yoshimasa Iwase)
Meet vs. Duo – 2 faces of Google’s WebRTC
Messenger was not forced to wiretap but…
RED: Improving Audio Quality with Redundancy
Reeling in Safari on WebRTC – A Closer Look at What’s Supported
Slack Does WebRTC Video – Here’s How (Gustavo Garcia)
The WhatsApp RTCP exploit – what might have happened?
What’s up with WhatsApp and WebRTC?
Wiresharking Wire
YouTube Does WebRTC – Here’s How

Standards

A Hitchhiker’s Guide to WebRTC standardization
Are we There Yet? WebRTC standards Q&A with Dan Burnett
Building Consensus on WebRTC – Q&A with W3C Editor Dan Burnett
Does your video call have End-to-End Encryption? Probably not..
Identifying Shared Tabs using Capture Handle (Elad Alon)
Implementing REDundant audio on an SFU
Is everyone switching to Unified Plan?
Not a Guide to SDP Munging
ORTC is not the “Other” RTC: Q&A with ORTC CG Chair Robin Raymond
Perfect Negotiation
Real-Time Video Processing with WebCodecs and Streams: Processing Pipelines (Part 1)
RED: Improving Audio Quality with Redundancy
SDP: The worst of all worlds or why compromise can be a bad idea (Tim Panton)
SDP: Your Fears Are Unleashed (Iñaki Baz Castillo)
The IMS approach to WebRTC
The Microsoft in the Room – IE and WebRTC (or ORTC?)
Trick or Treat? Cisco’s OpenH264.org & What it Means in the WebRTC Video Battle
Updated: Why the WebRTC API has it wrong – Interview with WebRTC Object API (ORTC) co-author Iñaki Baz
Video Frame Processing on the Web – WebAssembly, WebGPU, WebGL, WebCodecs, WebNN, and WebTransport
WebRTC mandatory video codec discussion: the final duel?
WebRTC MUST implement DTLS-SRTP but… MUST NOT implement SDES?
WebRTC standardization is more than codecs – Q&A with Dan Burnett
WebRTC Standards Update Webinar
WebRTC Today & Tomorrow: Interview with W3C WebRTC Chair Bernard Aboba
WebRTC Video Codec Debate Positions Infographic
WebRTC Video Codec Debate: Is There No End in Sight? (Chris Wendt)

Technology

An Intro to WebRTC’s NAT/Firewall Problem
Apple’s not so private relay fails with WebRTC
Autoplay restrictions and WebRTC (Dag-Inge Aas)
Avoiding Contact Center IVR Hell with WebRTC
Breaking Point: WebRTC SFU Load Testing (Alex Gouaillard)
Calculating True End-to-End RTT (Balázs Kreith)
Can an Open Source SFU Survive Acquisition? Q&A with Jitsi & Atlassian HipChat
Chrome Screensharing Blues – preparing for getDisplayMedia
Chrome’s WebRTC VP9 SVC Layer Cake: Sergio Garcia Murillo & Gustavo Garcia
coturn: No Time to Die – Q&A with new project leads
coTURN: the open-source multi-tenant TURN/STUN server you were looking for
Finding the Warts in WebAssembly+WebRTC
First steps with QUIC DataChannels
How does WebRTC End-to-End Encryption work? Matrix.org example (Dave Baker)
How Go-based Pion attracted WebRTC Mass – Q&A with Sean Dubois
ICE always tastes better when it trickles! (Emil Ivov)
Identifying Shared Tabs using Capture Handle (Elad Alon)
Implementing REDundant audio on an SFU
Improving Scale and Media Quality with Cascading SFUs (Boris Grozev)
Is Slack’s WebRTC Really Slacking? (Yoshimasa Iwase)
Making WebRTC source building not suck (Alex Gouaillard)
Microsoft’s ORTC Edge for WebRTC – Q&A with Bernard Aboba
New Windows into WebRTC with UWP: Q&A with Microsoft’s James Cadd
OMG WebRTC is tracking me! Or is it?
Open Source Cloud Gaming with WebRTC
Optimizing video quality using Simulcast (Oscar Divorra)
Optimizing WebRTC Power Consumption (Markus Handell)
orca.js: open real-time communications API
Plug-in free or free plug-in? Q&A with IE & Safari WebRTC plug-in maker Alex Gouaillard
Progressive Web Apps (PWA) for WebRTC (Trond Kjetil Bremnes)
Project WONDER: showing WebRTC NNI does not need SIP
Real-Time Video Processing with WebCodecs and Streams: Processing Pipelines (Part 1)
RED: Improving Audio Quality with Redundancy
Revealing mediasoup’s core ingredients: Q&A with Iñaki Baz Castillo
Signalling Options for WebRTC Applications
So your VPN is leaking because of Chrome’s WebRTC…
STUN the Network – How STUN helps WebRTC Traverse NATs
Suspending Simulcast Streams for Savvy Streamlining (Brian Baldino)
The Open Source rfc5766-turn-server Project – Interview with Oleg Moskalenko
The WebRTC Bitcode Soap Opera (Saúl Ibarra Corretgé)
The WhatsApp RTCP exploit – what might have happened?
Traffic Encryption
Trunking WebRTC: the user authentication challenge (Torrey Searle)
Video Frame Processing on the Web – WebAssembly, WebGPU, WebGL, WebCodecs, WebNN, and WebTransport
WebRTC and Man in the Middle Attacks
WebRTC beyond one-to-one communication (Gustavo Garcia Bernardo)
WebRTC Video Codec Decision is… NO DECISION
What I learned about H.264 for WebRTC video (Tim Panton)
What is a WebRTC Gateway anyway? (Lorenzo Miniero)
What’s in a WebRTC JavaScript Library?
Why put WebRTC in Webkit? Q&A with the webrtcinwebkit team
Wiresharking Wire

An Intro to WebRTC’s NAT/Firewall Problem

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Most folks that set out to write an application, or build an architecture, begin with nothing but features and functionality in mind.  Many might start out assuming they will be traversing flat, reliable, and secure networks.  Inevitably, reality sets in as one starts to demo or prototype much beyond the friendly confines of the lab, […]

Autoplay restrictions and WebRTC (Dag-Inge Aas)

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One of the great things about WebRTC is that it is built right into the web platform. The web platform is generally great for WebRTC, but occasionally it can cause huge headaches when specific WebRTC needs do not exactly align with more general browser usage requirements. The latest example of this is has to do […]

Avoiding Contact Center IVR Hell with WebRTC

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A couple of decades ago if you bought something of any reasonable complexity, odds are it came with a call center number you had to call in case something went wrong. Perhaps like the airline industry, economic pressures on contact centers shifted their modus operandi from customer delight to cost reduction. Unsurprisingly this has not done […]

Breaking Point: WebRTC SFU Load Testing (Alex Gouaillard)

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If you plan to have multiple participants in your WebRTC calls then you will probably end up using a Selective Forwarding Unit (SFU).  Capacity planning for SFU’s can be difficult – there are estimates to be made for where they should be placed, how much bandwidth they will consume, and what kind of servers you […]

Calculating True End-to-End RTT (Balázs Kreith)

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Balázs Kreith of the open-source WebRTC monitoring project, ObserveRTC shows how to calculate WebRTC latency – aka Round Trip Time (RTT) – in p2p scenarios and end-to-end across one or more with SFUs. WebRTC’s getStats provides relatively easy access to RTT values, bu using those values in a real-world environment for accurate results is more difficult. He provides a step-by-step guide using some simple Docke examples that compute end-to-end RTT with a single SFU and in cascaded SFU environments.

Can an Open Source SFU Survive Acquisition? Q&A with Jitsi & Atlassian HipChat

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Atlassian’s HipChat acquired BlueJimp, the company behind the Jitsi open source project. Other than for positive motivation, why should WebRTC developers care? Well, Jitsi had its Jitsi Video Bridge (JVB) which was one of the few open source Selective Forwarding Units (SFU) projects out there. Jitsi’s founder and past webrtcHacks guest author, Emil Ivov, was a […]

Chrome Screensharing Blues – preparing for getDisplayMedia

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The Chrome Webstore has decided to stop allowing inline installation for Chrome extensions. This has quite an impact on WebRTC applications since screensharing in Chrome currently requires an extension. Will the getDisplayMedia API come to the rescue? Screensharing in Chrome When screensharing was introduced in Chrome 33, it required implementation via an extension as a way to […]

Chrome’s WebRTC VP9 SVC Layer Cake: Sergio Garcia Murillo & Gustavo Garcia

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Multi-party calling architectures are a common topic here at webrtcHacks, largely because group calling is widely needed but difficult to implement and understand. Most would agree Scalable Video Coding (SVC) is the most advanced, but the most complex multi-party calling architecture. To help explain how it works we have brought in not one, but two WebRTC video architecture experts. […]

Finding the Warts in WebAssembly+WebRTC

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A while ago we looked at how Zoom was avoiding WebRTC by using WebAssembly to ship their own audio and video codecs instead of using the ones built into the browser’s WebRTC.  I found an interesting branch in Google’s main (and sadly mostly abandoned) WebRTC sample application apprtc this past January. The branch is named […]

First steps with QUIC DataChannels

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Note: as of March 2021 both experiments no longer work in Chrome. QUIC-based DataChannels are being considered as an alternative to the current SCTP-based transport. The WebRTC folks at Google are experimenting  with it: Looking for feedback: QUIC based RTCQuicTransport and RTCIceTransport API's are available as origin trial in Chrome 73 for experimentation.https://t.co/KVVEVmggms — WebRTC […]

How does WebRTC End-to-End Encryption work? Matrix.org example (Dave Baker)

· · Leave a Comment

One of WebRTC’s great features is its mandated strong encryption.  Encryption mechanisms are built-in, meaning developers don’t (often) need to deal with the details. However, these easy, built-in encryption mechanisms assume you have: 1) media is communicated peer-to-peer and 2) a secure signaling channel setup. Most group-calling services make use of a media server device, […]

How Go-based Pion attracted WebRTC Mass – Q&A with Sean Dubois

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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 […]

ICE always tastes better when it trickles! (Emil Ivov)

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For the last year and a half I’ve been working with a number of customers helping them to understand what WebRTC is about, supporting them in the definition of new products, services, and in some cases even developing WebRTC prototypes/labs for them. I’ve spent time with Service Providers, Enterprise and OTT customers and the very […]

Identifying Shared Tabs using Capture Handle (Elad Alon)

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Introduction to capture handle – a new Chrome Origin Trial that lets a WebRTC screen sharing application communicate with the tab it is capturing. Examples use case discussed include detecting self-capture, improving the use of collaboration apps that are screen shared, and optimizing stream parameters of the captured content.

Implementing REDundant audio on an SFU

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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. […]

Improving Scale and Media Quality with Cascading SFUs (Boris Grozev)

· · 1 Comment

Deploying media servers for WebRTC has two major challenges, scaling beyond a single server as well as optimizing the media latency for all users in the conference. While simple sharding approaches like “send all users in conference X to server Y” are easy to scale horizontally, they are far from optimal in terms of the […]

Is Slack’s WebRTC Really Slacking? (Yoshimasa Iwase)

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Earlier this month Fippo published a post analyzing Slack’s new WebRTC implementation. He did not have direct access or a team account to do a thorough deep dive – not to mention he is supposed to be taking some off this month. That left many with some open questions? Is there more to the TURN network? […]

Making WebRTC source building not suck (Alex Gouaillard)

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One of WebRTC’s benefits is that the source to it is all open source. Building WebRTC from source provides you the ultimate flexibility to do what you want with the code, but it is also crazy difficult for all but the small few VoIP stack developers who have been dedicated to doing this for years. […]

Microsoft’s ORTC Edge for WebRTC – Q&A with Bernard Aboba

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We have been waiting a long time for Microsoft to add WebRTC to its browser portfolio. That day finally came last month when Microsoft announced its new Windows 10 Edge browser had ORTC. This certainly does not immediately address the Internet Explorer population and ORTC is still new to many (which is why we cover it […]

OMG WebRTC is tracking me! Or is it?

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There has been more noise about WebRTC making it possible to track users. We have covered some of the nefarious uses of WebRTC and look out for it before. After reading a blog post on this topic covering some allegedly new unaddressed issues a week ago I decided to ignore it after some discussion on the mozilla […]

Open Source Cloud Gaming with WebRTC

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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 […]

Optimizing video quality using Simulcast (Oscar Divorra)

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Dealing with multi-party video infrastructure can be pretty daunting. The good news is the technology, products, and standards to enable economical multiparty video in WebRTC has matured quite a bit in the past few years. One of the key underlying technologies enabling some of this change is called simulcast. Simulcast has been an occasional sub-topic […]

Optimizing WebRTC Power Consumption (Markus Handell)

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The performance of WebRTC in Chrome as well as other RTC applications needed to be improved a lot during the pandemic when more people with a more diverse set of machines and network connections started to rely on video conferencing. Markus Handell is a team lead at Google who cares a lot about performance of […]

orca.js: open real-time communications API

· · Leave a Comment

WebRTC promises to greatly simplify the development of multimedia realtime communications, without the need to install an application or browser plug-in. It enables this by exposing a media engine and the network stack through a set of specialised APIs. Application developers can use these APIs to easily add realtime communication to web applications. The defined […]

Progressive Web Apps (PWA) for WebRTC (Trond Kjetil Bremnes)

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One of WebRTC’s biggest challenges has been providing consistent, reliable support across platforms. For most apps, especially those that started on the web, this generally means developing a native or hybrid mobile app in addition to supporting the web app.  Progressive Web Apps (PWA) is a new concept that promises to unify the web for […]

Project WONDER: showing WebRTC NNI does not need SIP

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As discussed in previous posts, WebRTC standards do not specify a signaling protocol. In general this decision is positive by giving developers the freedom to select (or invent) the protocol that best suits the particular WebRTC application’s needs. This can also reduce the time to market since standards compliance-related tasks are minimized. WebRTC media and […]

RED: Improving Audio Quality with Redundancy

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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 […]

Revealing mediasoup’s core ingredients: Q&A with Iñaki Baz Castillo

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I interviewed mediasoup’s co-founder, Iñaki Baz Castillo, about how the project got started, what makes it different, their recent Rust support, and how he maintains a developer community there despite the project’s relative unapproachability. mediasoup was one of the second-generation Selective Forwarding Units (SFUs). This second generation emerged to incorporate different approaches or address different use cases a few years after the first generation of SFUs came to market. mediasoup was and is different. It is node.js-based, built as a library to be part of a serve app, and incorporated the Object-oriented approaches used by ORTC – the alternative spec to WebRTC at the time. Today, mediasoup is a popular SFU choice among skilled WebRTC developers. mediasoup’s low-level native means this skill is required.

Signalling Options for WebRTC Applications

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As I described in the standardization post, the model used in WebRTC for real-time, browser-based applications does not envision that the browser will contain all the functions needed to function as a telephone or video conferencing unit. Instead, is specifies the browser will contain the functions that are needed to run a Web application which would […]

So your VPN is leaking because of Chrome’s WebRTC…

· · 2 Comments

We have covered the “WebRTC is leaking your IP address” topic a few times, like when I reported what the NY Times was doing and in my WebRTC-Notifier. Periodically this topic comes up now and again in the blogosphere, generally with great shock and horror. This happened again recently, so here is an updated look into […]

STUN the Network – How STUN helps WebRTC Traverse NATs

· · 5 Comments

In my last post (a long time ago) I introduced the issue of NATs and Firewalls, and the tools WebRTC uses to overcome them.  First off, my apologies for the lengthy hiatus after promising to continue the discussion of NAT/Firewall traversal.  Since that entry, I became a Dad for the 2nd time, and lets just […]

The WhatsApp RTCP exploit – what might have happened?

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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 […]

Traffic Encryption

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So I talked about Skype and Viber at KrankyGeek two weeks ago. Watch the video on youtube or take a look at the slides. No “reports” or packet dumps to publish this time, mostly because it is very hard to draw conclusions from the results. The VoIP services we have looked at so far which […]

Video Frame Processing on the Web – WebAssembly, WebGPU, WebGL, WebCodecs, WebNN, and WebTransport

· · Leave a Comment

There are a lot of options for reading and changing the pixels inside a video frame. In this post, W3C specialists François Daoust and Dominique Hazaël-Massieux (Dom) review every web-based option for processing video frames on the web available today – JavaScript, WebAssembly (wasm), WebGPU, WebGL, WebCodecs, Web Neural Networks (WebNN), and WebTransport.

WebRTC and Man in the Middle Attacks

· · 6 Comments

WebRTC is supposed to be secure. A lot more than previous VoIP standards. It isn’t because it uses any special new mechanism, but rather because it takes it seriously and mandates it for all sessions. Alan Johnston decided to take WebRTC for a MitM spin – checking how easy is it to devise a man-in-the-middle […]

WebRTC beyond one-to-one communication (Gustavo Garcia Bernardo)

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WebRTC and its peer-to-peer capabilities are great for one-to-one communications. However, when I discuss with customers use cases and services that go beyond one-to-one, namely one-to-many or many-to-many, the question arises: “OK, but what architecture shall I use for this?”. Some service providers want to reuse the multicast support they have in their networks (we […]

WebRTC Video Codec Decision is… NO DECISION

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As we discussed in previous posts, the IETF is meeting this week in Vancouver. Lots of interesting discussing including two sessions for the RTCWeb WG; the agenda for the two sessions can be found here. The first session, which was held on Monday, was mainly about updates on the JSEP (Javascript Session Establishment Protocol) specification, […]

What I learned about H.264 for WebRTC video (Tim Panton)

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It has been a few years since the WebRTC codec wars ended in a detente. H.264 has been around for more than 15 years so it is easy to gloss over the the many intricacies that make it work. Reknown hackathon star, live-coder, and |pipe| CTO Tim Panton was working on a drone project where he needed […]

What is a WebRTC Gateway anyway? (Lorenzo Miniero)

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As I mentioned in my ‘WebRTC meets telecom’ article a couple of weeks ago, at Quobis we’re currently involved in 30+ WebRTC field trials/POCs which involve in one way or another a telco network. In most cases service providers are trying to provide WebRTC-based access to their existing/legacy infrastructure and services (fortunately, in some cases it’s […]

What’s in a WebRTC JavaScript Library?

· · 9 Comments

Looking over the past few years of WebRTC growth, and the landscape of emerging WebRTC solutions, we see quite a number of  WebRTC-centric JavaScript (JS) libraries on the scene. Indeed, not long after browser vendors began shipping WebRTC implementations, a bouquet of WebRTC signaling libraries bloomed. Each delivers a JavaScript API surface in the browser, which provides […]

Why put WebRTC in Webkit? Q&A with the webrtcinwebkit team

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The world of browsers and how they work is both complex and fascinating. For those that are new to the browser engine landscape, Google, Apple, and many others collaborated on an open source web rendering engine for many years known as WebKit.  WebKit has active community with many less well known browsers that use it, so the […]

Wiresharking Wire

· · Leave a Comment

This is the next decode and analysis in Philipp Hancke’s Blackbox Exploration series conducted by &yet in collaboration with Google. Please see our previous posts covering WhatsApp, Facebook Messenger and FaceTime for more details on these services and this series. {“editor”: “chad hart“} Wire is an attempt to reimagine communications for the mobile age. It is a messaging app […]

guides and information for WebRTC developers