Tag Archives: webrtc

WebRTC predictions for 2016

I wrote these predictions in the first week of January and meant to publish them as encouragement to think about where WebRTC still needs some work. I’d like to be able to compare the state of WebRTC in the browser a year from now. Therefore, without further ado, here are my thoughts.

WebRTC Browser support

I’m quite optimistic when it comes to browser support for WebRTC. We have seen Edge bring in initial support last year and Apple looking to hire engineers to implement WebRTC. My prediction is that we will see the following developments in 2016:

  • Edge will become interoperable with Chrome and Firefox, i.e. it will publish VP8/VP9 and H.264/H.265 support
  • Firefox of course continues to support both VP8/VP9 and H.264/H.265
  • Chrome will follow the spec and implement H.264/H.265 support (to add to their already existing VP8/VP9 support)
  • Safari will enter the WebRTC space but only with H.264/H.265 support

Codec Observations

With Edge and Safari entering the WebRTC space, there will be a larger focus on H.264/H.265. It will help with creating interoperability between the browsers.

However, since there are so many flavours of H.264/H.265, I expect that when different browsers are used at different endpoints, we will get poor quality video calls because of having to negotiate a common denominator. Certainly, baseline will work interoperably, but better encoding quality and lower bandwidth will only be achieved if all endpoints use the same browser.

Thus, we will get to the funny situation where we buy ourselves interoperability at the cost of video quality and bandwidth. I’d call that a “degree of interoperability” and not the best possible outcome.

I’m going to go out on a limb and say that at this stage, Google is going to consider strongly to improve the case of VP8/VP9 by improving its bandwidth adaptability: I think they will buy themselves some SVC capability and make VP9 the best quality codec for live video conferencing. Thus, when Safari eventually follows the standard and also implements VP8/VP9 support, the interoperability win of H.264/H.265 will become only temporary overshadowed by a vastly better video quality when using VP9.

The Enterprise Boundary

Like all video conferencing technology, WebRTC is having a hard time dealing with the corporate boundary: firewalls and proxies get in the way of setting up video connections from within an enterprise to people outside.

The telco world has come up with the concept of SBCs (session border controller). SBCs come packed with functionality to deal with security, signalling protocol translation, Quality of Service policing, regulatory requirements, statistics, billing, and even media service like transcoding.

SBCs are a total overkill for a world where a large number of Web applications simply want to add a WebRTC feature – probably mostly to provide a video or audio customer support service, but it could be a live training session with call-in, or an interest group conference all.

We cannot install a custom SBC solution for every WebRTC service provider in every enterprise. That’s like saying we need a custom Web proxy for every Web server. It doesn’t scale.

Cloud services thrive on their ability to sell directly to an individual in an organisation on their credit card without that individual having to ask their IT department to put special rules in place. WebRTC will not make progress in the corporate environment unless this is fixed.

We need a solution that allows all WebRTC services to get through an enterprise firewall and enterprise proxy. I think the WebRTC standards have done pretty well with firewalls and connecting to a TURN server on port 443 will do the trick most of the time. But enterprise proxies are the next frontier.

What it takes is some kind of media packet forwarding service that sits on the firewall or in a proxy and allows WebRTC media packets through – maybe with some configuration that is necessary in the browsers or the Web app to add this service as another type of TURN server.

I don’t have a full understanding of the problems involved, but I think such a solution is vital before WebRTC can go mainstream. I expect that this year we will see some clever people coming up with a solution for this and a new type of product will be born and rolled out to enterprises around the world.


So these are my predictions. In summary, they address the key areas where I think WebRTC still has to make progress: interoperability between browsers, video quality at low bitrates, and the enterprise boundary. I’m really curious to see where we stand with these a year from now.

It’s worth mentioning Philipp Hancke’s tweet reply to my post:

— we saw some clever people come up with a solution already. Now it needs to be implemented 🙂

Progress with rtc.io

At the end of July, I gave a presentation about WebRTC and rtc.io at the WDCNZ Web Dev Conference in beautiful Wellington, NZ.


Putting that talk together reminded me about how far we have come in the last year both with the progress of WebRTC, its standards and browser implementations, as well as with our own small team at NICTA and our rtc.io WebRTC toolbox.

WDCNZ presentation page5

One of the most exciting opportunities is still under-exploited: the data channel. When I talked about the above slide and pointed out Bananabread, PeerCDN, Copay, PubNub and also later WebTorrent, that’s where I really started to get Web Developers excited about WebRTC. They can totally see the shift in paradigm to peer-to-peer applications away from the Server-based architecture of the current Web.

Many were also excited to learn more about rtc.io, our own npm nodules based approach to a JavaScript API for WebRTC.


We believe that the World of JavaScript has reached a critical stage where we can no longer code by copy-and-paste of JavaScript snippets from all over the Web universe. We need a more structured module reuse approach to JavaScript. Node with JavaScript on the back end really only motivated this development. However, we’ve needed it for a long time on the front end, too. One big library (jquery anyone?) that does everything that anyone could ever need on the front-end isn’t going to work any longer with the amount of functionality that we now expect Web applications to support. Just look at the insane growth of npm compared to other module collections:

Packages per day across popular platforms (Shamelessly copied from: http://blog.nodejitsu.com/npm-innovation-through-modularity/)

For those that – like myself – found it difficult to understand how to tap into the sheer power of npm modules as a font end developer, simply use browserify. npm modules are prepared following the CommonJS module definition spec. Browserify works natively with that and “compiles” all the dependencies of a npm modules into a single bundle.js file that you can use on the front end through a script tag as you would in plain HTML. You can learn more about browserify and module definitions and how to use browserify.

For those of you not quite ready to dive in with browserify we have prepared prepared the rtc module, which exposes the most commonly used packages of rtc.io through an “RTC” object from a browserified JavaScript file. You can also directly download the JavaScript file from GitHub.

Using rtc.io rtc JS library
Using rtc.io rtc JS library

So, I hope you enjoy rtc.io and I hope you enjoy my slides and large collection of interesting links inside the deck, and of course: enjoy WebRTC! Thanks to Damon, JEeff, Cathy, Pete and Nathan – you’re an awesome team!

On a side note, I was really excited to meet the author of browserify, James Halliday (@substack) at WDCNZ, whose talk on “building your own tools” seemed to take me back to the times where everything was done on the command-line. I think James is using Node and the Web in a way that would appeal to a Linux Kernel developer. Fascinating!!

AppRTC : Google’s WebRTC test app and its parameters

If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC: AppRTC.

When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

Here are my favourite parameters:

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

For example, here’s how a stereo, HD loopback test would look like: https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery:

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

Have fun playing with the main and always up-to-date WebRTC application: AppRTC.

UPDATE 12 May 2014

AppRTC now also supports the following bitrate controls:

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

Example usage: https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true

Use deck.js as a remote presentation tool

deck.js is one of the new HTML5-based presentation tools. It’s simple to use, in particular for your basic, every-day presentation needs. You can also create more complex slides with animations etc. if you know your HTML and CSS.

Yesterday at linux.conf.au (LCA), I gave a presentation using deck.js. But I didn’t give it from the lectern in the room in Perth where LCA is being held – instead I gave it from the comfort of my home office at the other end of the country.

I used my laptop with in-built webcam and my Chrome browser to give this presentation. Beforehand, I had uploaded the presentation to a Web server and shared the link with the organiser of my speaker track, who was on site in Perth and had set up his laptop in the same fashion as myself. His screen was projecting the Chrome tab in which my slides were loaded and he had hooked up the audio output of his laptop to the room speaker system. His camera was pointed at the audience so I could see their reaction.

I loaded a slide master URL:
and the room loaded the URL without query string:

Then I gave my talk exactly as I would if I was in the same room. Yes, it felt exactly as though I was there, including nervousness and audience feedback.

How did we do that? WebRTC (Web Real-time Communication) to the rescue, of course!

We used one of the modules of the rtc.io project called rtc-glue to add the video conferencing functionality and the slide navigation to deck.js. It was actually really really simple!

Here are the few things we added to deck.js to make it work:

  • Code added to index.html to make the video connection work:
    <meta name="rtc-signalhost" content="http://rtc.io/switchboard/">
    <meta name="rtc-room" content="lca2014">
    <video id="localV" rtc-capture="camera" muted></video>
    <video id="peerV" rtc-peer rtc-stream="localV"></video>
    <script src="glue.js"></script>
    glue.config.iceServers = [{ url: 'stun:stun.l.google.com:19302' }];

    The iceServers config is required to punch through firewalls – you may also need a TURN server. Note that you need a signalling server – in our case we used http://rtc.io/switchboard/, which runs the code from rtc-switchboard.

  • Added glue.js library to deck.js:

    Downloaded from https://raw.github.com/rtc-io/rtc-glue/master/dist/glue.js into the source directory of deck.js.

  • Code added to index.html to synchronize slide navigation:
    glue.events.once('connected', function(signaller) {
      if (location.search.slice(1) !== '') {
        $(document).bind('deck.change', function(evt, from, to) {
          signaller.send('/slide', {
            idx: to,
            sender: signaller.id
      signaller.on('slide', function(data) {
        console.log('received notification to change to slide: ', data.idx);
        $.deck('go', data.idx);

    This simply registers a callback on the slide master end to send a slide position message to the room end, and a callback on the room end that initiates the slide navigation.

And that’s it!

You can find my slide deck on GitHub.

Feel free to write your own slides in this manner – I would love to have more users of this approach. It should also be fairly simple to extend this to share pointer positions, so you can actually use the mouse pointer to point to things on your slides remotely. Would love to hear your experiences!

Note that the slides are actually a talk about the rtc.io project, so if you want to find out more about these modules and what other things you can do, read the slide deck or watch the talk when it has been published by LCA.

Many thanks to Damon Oehlman for his help in getting this working.

BTW: somebody should really fix that print style sheet for deck.js – I’m only ever getting the one slide that is currently showing. 😉

WebRTC books – a brief review

I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

Rob’s focus is very much on the features required in a typical Web application:

  • video calls
  • audio calls
  • text chats
  • file sharing

In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

Rob then also extends his introduction to WebRTC to two key application areas: education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.

Video Conferencing in HTML5: WebRTC via Socket.io

Six months ago I experimented with Web sockets for WebRTC and the early implementations of PeerConnection in Chrome. Last week I gave a presentation about WebRTC at Linux.conf.au, so it was time to update that codebase.

I decided to use socket.io for the signalling following the idea of Luc, which made the server code even smaller and reduced it to a mere reflector:

 var app = require('http').createServer().listen(1337);
 var io = require('socket.io').listen(app);

 io.sockets.on('connection', function(socket) {
         socket.on('message', function(message) {
         socket.broadcast.emit('message', message);

Then I turned to the client code. I was surprised to see the massive changes that PeerConnection has gone through. Check out my slide deck to see the different components that are now necessary to create a PeerConnection.

I was particularly surprised to see the SDP object now fully exposed to JavaScript and thus the ability to manipulate it directly rather than through some API. This allows Web developers to manipulate the type of session that they are asking the browsers to set up. I can imaging e.g. if they have support for a video codec in JavaScript that the browser does not provide built-in, they can add that codec to the set of choices to be offered to the peer. While it is flexible, I am concerned if this might create more problems than it solves. I guess we’ll have to wait and see.

I was also surprised by the need to use ICE, even though in my experiment I got away with an empty list of ICE servers – the ICE messages just got exchanged through the socket.io server. I am not sure whether this is a bug, but I was very happy about it because it meant I could run the whole demo on a completely separate network from the Internet.

The most exciting news since my talk is that Mozilla and Google have managed to get a PeerConnection working between Firefox and Chrome – this is the first cross-browser video conference call without a plugin! The code differences are minor.

Since the specification of the WebRTC API and of the MediaStream API are now official Working Drafts at the W3C, I expect other browsers will follow. I am also looking forward to the possibilities of:

The best places to learn about the latest possibilities of WebRTC are webrtc.org and the W3C WebRTC WG. code.google.com has open source code that continues to be updated to the latest released and interoperable features in browsers.

The video of my talk is in the process of being published. There is a MP4 version on the Linux Australia mirror server, but I expect it will be published properly soon. I will update the blog post when that happens.