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Guides and information for WebRTC developers
Updated: 36 min 11 sec ago

New Windows into WebRTC with UWP: Q&A with Microsoft’s James Cadd

Wed, 02/22/2017 - 13:30

While Windows may no longer be the default platform it was a decade ago it still has a huge and active community. More than 400 million devices support Windows 10 and there are many millions of .NET and Visual Studio users out there. In fact, I made my first WebRTC application in .NET using XSockets years ago. In […]

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

Wed, 02/15/2017 - 01:29

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. […]

The post Chrome’s WebRTC VP9 SVC Layer Cake: Sergio Garcia Murillo & Gustavo Garcia appeared first on webrtcHacks.

Slack Does WebRTC Video – Here’s How (Gustavo Garcia)

Thu, 12/15/2016 - 21:53

Slack is an über popular and fast growing communications tool that has a ton of integrations with various WebRTC services. It also acquired a WebRTC company a year ago and launched its own audio conferencing service earlier this year which we analyzed here and here. Earlier this week they launched video. Does this work the same? Are […]

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How to limit WebRTC bandwidth by modifying the SDP

Wed, 10/26/2016 - 17:12

WebRTC 1.0 uses SDP for negotiating capabilities between parties.  While there are a growing number of objects coming to WebRTC to avoid this protocol from the 90’s , the reality is SDP will be with us for some time. If you want to do things like change codecs or adjust bandwidth limits, then you’re going to need to “munge” […]

The post How to limit WebRTC bandwidth by modifying the SDP appeared first on webrtcHacks.

WebRTC media servers in the Cloud: lessons learned (Luis López Fernández)

Fri, 09/09/2016 - 13:32

Media servers, server-side media handling devices, continue to be a popular topic of discussion in WebRTC. One reason for this because they are the most complex elements in a VoIP architecture and that lends itself to differing approaches and misunderstandings. Putting WebRTC media servers in the cloud and reliably scaling them is  even harder. Fortunately there are […]

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Let’s Encrypt – how get to free SSL for WebRTC

Mon, 08/01/2016 - 21:16

Way back in 47 (version that is), Chrome started to mandate the use of HTTPS in conjunction with getUserMedia. To use HTTPS you need a SSL/TLS certificate.  Xander Dumaine covered this a bit for us before, but I still see a lot of people out there struggle with it. As it so happens, the certificate for my […]

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Optimizing video quality using Simulcast (Oscar Divorra)

Thu, 06/16/2016 - 21:37

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 […]

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Your Browser as a Audio Conference Server with WebRTC & Web Audio (Alexey Aylarov)

Fri, 05/20/2016 - 11:04

Conference calling is a multi-billion dollar industry that is mostly powered by expensive, high-powered conferencing servers. Now you can replicate much of this functionality for free with a modern browser using the combination of WebRTC and WebAudio. Like with video, multi-party audio can utilize a few architectures: Full mesh – each client sends their audio […]

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Update: Anatomy of a WebRTC SDP (Antón Román)

Thu, 05/05/2016 - 14:33

Session Description Protocol (SDP) is a fundamental, but very unintuitive concept behind how WebRTC works today. Its no wonder that the Anatomy of a WebRTC SDP post and the interactive SDP guide by Quobis CTO, Antón Román has been so popular here on webrtcHacks. With all things WebRTC, things have changed and we were due for an […]

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Sharpening the Edge – extended Q&A with Microsoft for RTC devs

Thu, 04/21/2016 - 17:22

Two weeks ago Microsoft’s Bernard Aboba (and former webrtcHack’s interviewee) gave an update on Edge’s ORTC and WebRTC at the Microsoft Build conference. He covered some big topics including VP8 and WebRTC 1.0 support. You can see the update video here or read the follow-up post for details. Then last week Microsoft announced plug-in free Skype […]

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The Big Churn – learning from real usage stats (Lasse Lumiaho and Varun Singh)

Fri, 04/08/2016 - 15:39

Losing customers because of issues with your network service is a bad thing. Sure you can gather data and try to prevent, but isn’t it better to prevent issues in the first place? What are the most common pitfalls to look out for? What’s a good benchmark? What WebRTC-specific user experience elements should you spend […]

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

Thu, 03/24/2016 - 13:25

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? […]

The post Is Slack’s WebRTC Really Slacking? (Yoshimasa Iwase) appeared first on webrtcHacks.

Dear Slack: why is your WebRTC so weak?

Thu, 03/03/2016 - 20:41

  Dear Slack, There has been quite some buzz this week about you and WebRTC. WebRTC… kind of. Because actually you only do stuff in Chrome and your native apps: I’ve been there. Launching stuff only for Chrome. That was is late 2012. In 2016, you need to have a very good excuse to launch something […]

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getUserMedia resolutions III – constraints unleashed

Mon, 02/08/2016 - 14:03

Back in October 2013,  the relative early days of WebRTC, I set out to get a better understanding of the getUserMedia API and camera constraints in one of my first and most popular posts. I discovered that working with getUserMedia constraints was not all that straight forward. A year later I gave an update after the […]

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Surviving Mandatory HTTPS in Chrome (Xander Dumaine)

Thu, 12/17/2015 - 13:11

Xander Dumaine provides some strategies and code for dealing with the new secure origin only policy in Chrome 47+ that forces the use of HTTPS.

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Shut up! Monitoring audio volume in getUserMedia

Thu, 12/10/2015 - 13:14

A few days back my old friend Chris Koehnke, better known as “Kranky” asked me how hard it would be to implement a wild idea he had to monitor what percentage of the time you spent talking instead of listening on a call when using WebRTC. When I said “one day” that made him wonder whether he could offshore it to save money. Well… good luck!

A week later Kranky showed me some code. Wait, he is writing code? It was not bad – it was using the WebAudio API so going in the right direction. It was enough to prod me to finish writing the app for him.

The audio stream volume sample application from Google calculates the root mean square (RMS) of the audio signal which is extracted from the input stream using a script processor every 200ms. There is a lot of tuning options here of course.

Instead of starting from scratch, I decided to use hark, a small open source module for this task that my coworker Philip Roberts had built in mid-2013 when the WebAudio API became first available.

Instead of the RMS, hark uses the Fast Fourier Transformation to obtain a frequency domain representation of the input signal. Then, hark picks the maximum amplitude as an indication for the volume of the signal. Let’s try this (full code here):

var hark = require('../hark.js') var getUserMedia = require('getusermedia') getUserMedia(function(err, stream) { if (err) throw err var options = {}; var speechEvents = hark(stream, options); speechEvents.on('volume_change', function(volume) { console.log('current volume', volume); }); });

On top of this, hark uses a simple speech detection algorithm that considers speech to be started when the maximum amplitude stays above a threshold for a number of milliseconds. Much less complicated than typical voice activity detection algorithms but pretty effective. And easy to use as well, just subscribe to two additional events:

speechEvents.on('speaking', function() { console.log('speaking'); }); speechEvents.on('stopped_speaking', function() { console.log('stopped_speaking'); });

Tuning the threshold for accurate speech detection is pretty tricky. So I needed visualization (and just requiring hark only took five minutes so I had plenty of time). Using the awesome Highcharts graph library I quickly added plot bands to the graph I was generating:

With the visualization I could easily see that the speech detection events happened a bit later than I expected since hark requires a certain history over the threshold for the trigger to work (say: 400ms).  To adjust for this in the graph had to substract this speech starting to trigger time from my x-axis (now()– 400ms for example).

That graph is still visibile on the more techie variant of the website so if you think the results are not accurate… it might help you figure out what is going on. I am happy with the current behavior.

The percentage of speech then calculated as the sum of the intervals that speech is detected divided by the duration of the call. As a display, a gauge chart is used with three different colors:

  • up to 65% speech time: green
  • up to 79%: yellow
  • more than 80%: red

Adding remote audio to this would be awesome. However, while the WebAudio API is supported for local media streams in Chrome, Firefox and Edge, it is only supported for remote streams in Firefox. Hooking this up with the getStats API (in Chrome) to get the audio level would certainly be possible, but would require calling getStats at a very high frequency to get proper averages.

Check out the app in action at talklessnow and let us know what you think.

{“author”: “Philipp Hancke“}

Want to keep up on our latest posts? Please click here to subscribe to our mailing list if you have not already. We only email post updates. You can also follow us on twitter at @webrtcHacks for blog updates and news of technical WebRTC topics or our individual feeds @chadwallacehart, @victorpascual and @tsahil.

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OMG WebRTC is tracking me! Or is it?

Thu, 11/05/2015 - 15:23

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 IRC channel. But this has some up on a the twitter-sphere again and Tsahi said ‘ouch’, here are my thoughts.


The blog post (available here) makes a number of claims about how certain Chrome behavior makes fingerprinting easier:

  • Chrome started caching certificates for 30 days recently, creating a cookie-like attack surface for privacy
  • this allows cross-origin tracking of users
  • the incognito mode behavior is inconsistent with respect to this

Caching certificates

First, there is a claim that the way Chrome caches certificates changed recently:

In the past, Google Chrome used to generate a new self-signed certificate for every WebRTC PeerConnection. But now (using Chrome 46, or maybe earlier as i did not check) it generates a self-signed certificate which is valid for one month and uses it for all PeerConnections of a particular domain.

The code used to demonstrate this behaviour is rather odd, too. It uses the getStats API to the query the fingerprint, which is also available more easily in the SDP.

Chrome has cached certificates in this way for about two years, this is not real news. One of the reasons for this is that it is rather expensive to generate the current private keys for DTLS, especially on mobile devices. In the future, there will be more control over this behaviour. Neither Firefox nor Edge currently cache certificates.

To be fair, the WebRTC team made a serious blunder here. Until Chrome 45, the certificate was not cleared when cookies were cleared, only when all data was cleared. The bugfix for this only appeared in the Chrome 47 release notes:

Issue 510850 DTLS cert should be cleared when cookies are cleared

Cross-Origin Tracking

So this part is not really news. The second claim made in the blog post is that this enables cross-origin tracking:

To test this go to http://www.kapejod.org/tracking/test.html and to http://kapejod.org/tracking/test.html. Open the network tab of Chrome’s developer console and compare the urls of the requested “tracking.png”. They should contain the same fingerprint, now!

They do. Now, let’s look at this test page:

// make up some random id var transactionId = 'xxxxxxxx-xxxx-4xxx-yxxx-xxxxxxxxxxxx'.replace(/[xy]/g, function(c) {var r = Math.random()*16|0,v=c=='x'?r:r&0x3|0x8;return v.toString(16);}); var fragment = document.createDocumentFragment(); var div = document.createElement("DIV"); div.innerHTML = '<iframe src="http://kapejod.org/tracking/identify.html?'+transactionId+'" width="1" height="1" style="display:none;"/>'; fragment.appendChild(div); document.body.insertBefore(fragment, document.body.childNodes[document.body.childNodes.length - 1]);

It includes the URL http://kapejod.org/tracking/identify.html. Let’s also look at the code there as well. It executes the code shown above and logs the fingerprint to the console:

console.log('your fingerprint is: ' + fingerprint);

Now why is the fingerprint the same? Well, the iframe is always included from kapejod.org. Which means the Javascript is executed within the context of this origin.
So Chrome can use the persisted fingerprint. As well as any cookies and localStorage data. The attack surface here is no worse than setting a cookie.

Another thing related to this (and I am surprised this has not yet been mentioned) are the deviceIds returned by navigator.mediaDevices.enumerateDevices. Those are also persisted with the same lifetime as cookies. The W3C mediacapture specification has a paragraph about security and privacy considerations on this:

The identifiers for the devices are designed to not be useful for a fingerprint that can track the user between origins, but the number of devices adds to the fingerprint surface. It recommends to treat the per-origin persistent identifier deviceId as other persistent storages (e.g. cookies) are treated.

Again, WebRTC and other HTML5 techniques increase the fingerprint surface. But by design, this is not worse than cookies or equivalent techniques like localStorage.

Incognito Mode

Last but not least the blog post makes claims about the incognito mode:

But to make it generate a new one you have to close ALL incognito tabs. Otherwise you can be tracked across multiple domains.

Again, this behaviour is consistent with the incognito mode behaviour for things like localStorage. In both Chrome and Firefox. In incognito mode, open a site, set something in localStorage. Open another tab. Close first tab. Navigate to same site. Check localStorage. Boo!


There is no real news here. In Germany, we call this ‘olle kamellen’.

{“author”: “Philipp Hancke“}

Want to keep up on our latest posts? Please click here to subscribe to our mailing list if you have not already. We only email post updates. You can also follow us on twitter at @webrtcHacks for blog updates and news of technical WebRTC topics or our individual feeds @chadwallacehart, @victorpascual and @tsahil.

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Are we There Yet? WebRTC standards Q&A with Dan Burnett

Wed, 10/21/2015 - 12:58

If you are new to WebRTC then you have missed out on years of drama in the standards bodies over various issues like SDP and codecs. These standards dictate what vendors must implement so they ultimately dictate the industry roadmap.  To get a deep perspective and appreciation of the issues, we like to ask Dan Burnett, W3C editor to comment on where we are at with the standardization process. I caught up with Dan at this year’s IIT Real Time Communications Conference and had the more detailed Q&A with him shortly thereafter.

We asked Dan to comment on recent spec changes, ORTC, the next version of WebRTC, codecs, Apple, when the 1.0 spec might ever be finalized, and a whole lot more.

{“editor”, “chad hart“}

New Governance

webrtcHacks: Hi Dan. Can you describe some of the recent changes to the W3C WebRTC governance?

Dan: Yes. There was a long-running but productive discussion among the members of the WebRTC Working Group (WG), ORTC Community Group (CG), and the some of the members of the W3C advisory committee – which is the group that officially determines group charters.

As part of the Charter renewal process, we decided that there would be one additional Chair of the WebRTC Working Group –  Eric Lagerway of Hookflash who was one of the initiators of ORTC. Also the decision was that the WebRTC WG is the official group where all future standardization work in WebRTC will happen, meaning the ORTC work will gradually fold into that group.

Additionally, the group was chartered to work on another version beyond 1.0 – WebRTC Next Version or WebRTC-NV.

There are 2 requirements on that version:

  1. There is no requirement that new features introduced in the specification have an SDP equivalent
  2. WebRTC NV is not a replacement for WebRTC 1.0 – it is an extension. It is expected that all browsers that support WebRTC NV will support 1.0 functionality as well.

One other thing has happened that is not official, but is probably good is that Bernard Aboba from Microsoft has joined the WebRTC 1.0 editing team.

The Next Version

webrtcHacks: yeah, Bernard mentioned that in the interview I did with him last week. Can you explain WebRTC NV? Why didn’t you just call it 2.0, or 1.1, or whatever?

Dan: I have been working on standards for a long time. I have seen groups spend ridiculous amounts of time deciding on a name for a specification.  In this particular case a “1.1” sounds like a minor change from “1.0” while “2.0” sounds like a major change. Some people want a minor change. Some people want a major change. If enough people want different minor changes it will end up being a 2.0 anyway because of the number of changes. The goal was to avoid that disagreement now so that we can move forward,.

webrtcHacks: So what is WebRTC NV then, beyond what you stated earlier about no SDP?

Dan: Nothing is officially decided but I expect that there will continue to be more low-level controls as in ORTC. This is complicated by the fact that new feature proposals are continuing to come in for 1.0. Many of these features are from ORTC.

In the Sapporo meeting coming up, Google will be sharing their idea for what should go into WebRTC-NV when we finally start working on it.

Dan at the IIT-RTC Conference

webrtcHacks: How do you see ORTC influencing the WebRTC spec? Is WebRTC-NV really just ORTC?

Dan: If it had to summarize WebRTC-NV I would say that it is the combination of WebRTC 1.0 and ORTC. It is a requirement that 1.0 applications continue to work in WebRTC-NV implementations. It is not required that ORTC applications work directly in WebRTC-NV.

I believe the ORTC community intends to modify ORTC as necessary to remain consistent with WebRTC as it evolves.

webrtcHacks: Is there an end-date to ORTC-then? When it is mostly merged with WebRTC-NV will it cease to exist?

Dan: I can’t speak for the ORTC group. I have not heard of an end date. You’ll have to ask one of the primary ORTC contributors.

Spec Changes

webrtcHacks: What are some of changes made to the specs recently. Particularly those that impact the developers out there?

Dan: First I would like to give a little plug for my webrtcstandards.info site where I have been putting exactly that sort of information over the past few months. I will mention some things here, but you can get more details on that site.

webrtcHacks: ok, we’ll give you one plug (laughs)

Dan: One of the biggest and most relevant changes on what we were just talking about is the introduction of the RTCsenders and RTCreceivers. These are objects that allow for both information and more direct control over how tracks are sent over a PeerConnection. Notice as part of this that we have moved from a stream based API to a track based API.

webrtcHacks: And what advantage does the track approach provide?

Dan: It turns out developers want to have more control over exactly how tracks are sent and received. For example being able to specify which codecs are to be used and the parameters used to configure those codecs. They should be able to configure some transport properties as well on a per track basis such as FEC, retransmission, and bandwidth. Because of this it really didn’t make sense to talk about streams as the primary primitive being sent over a PeerConnection since they are really just a collection of tracks.

One of Peter Thatcher ORTC update slide’s showing the differences between the WebRTC and ORTC API. source: IIT-RTC 2015

webrtcHacks: So the others?

Dan: First, on the one we just mentioned – that was a foundational change where we are going to be seeing many other changes later on. Now I’ll talk about the others that are not related to that.

One big change was the API’s have been converted to use ECMAScript Promises. I think I mentioned this last year.

webrtchacks: You did.

Dan: It has happened. It is now in the specifications.

Promises are now the recommended mechanism for WebRTC specifications and for web specifications in general for dealing with asynchronous function calls. Not so much for things that generate multiple events, but definitely for any single asynchronous function call.

This is part of the move of ECMAscript toward truly asynchronous function calls as you can see if you look at some of the thoughts or future versions of ECMAscript.

The original callback based API’s currently still exist but will eventually be deprecated. Developers should start using the Promise versions.

webrtcHacks: I know media capture from the DOM is another one.

Dan: There has been good progress on capturing media directly from media elements such as audio, video and canvas. Developers have had to use hacks up to this point to be able to capture a canvas for example. Maybe they would take snapshots, but that is not the same as a realtime media stream as you would get from a getUserMedia call.

The major changes going into the specification soon  are to try to reproduce the resulting media stream as faithfully as possible to what a user would experience from that element. For example, if the user is playing a video and pauses it and then resumes, the resulting stream should show the paused video for the amount of time it was paused and then resume again.

This seems to be what developers are most interested in.

webrtcHacks: can you talk about some of the use cases that are being referenced around this feature?

Dan: Shared whiteboard is probably the best example, but there maybe some instances for training purposes where you want to capture how the user has interacted with existing elements – video or audio.

webrtcHacks: What about screensharing?

Dan: There is good progress happening there as well on the specification. It still has some tricky issues in terms of what apps should be able to request to be shared and what users should have control over. An example of this is Microsoft Powerpoint – if a user has 3 powerpoint documents up – say different presentations for different clients; they are likely to only want to share one one of those presentations – one window of that application. That works great until they go into presentation mode, which is far as the computer is concerned is a different window. So is this a case where the user should decide or is this a case where the application should decide what is shared?

In general the WG believes that the user should have the control, but browsers may have to make special cases for known applications such as Powerpoint so that it just works.

webrtcHacks: How about simulcast?

Dan: At the Seattle meeting there were some strong opinions on how simulcast should work and some proposals. Each time we get to the details the discussions diverge rather than converge. We all want it but we do not agree on how it should be signaled.


webrtcHacks: Now for an easier one. When will 1.0 be done?


Dan: I am tempted to give a similar answer as last year.

There are 2 primary specifications. The media capture specification is right now finishing up addressing the comments from its Last Call review which is the wide range review that is required in order to go forward. There aren’t any new features being requested by group members – it’s just cleaning up and fixing.

It probably will be stable within another 6 months.

webrtcHacks: Stable meaning not changing any more?

Dan: Yes – meaning no contentful changes. Only editorial fixes.

Now the WebRTC specification has the problem that new features keep coming in.

werbrtcHacks: Just to clarify – the Media Capture group is the getUserMedia API and when you WebRTC, that means the RTCPeerConnection and DataChannel related API’s?

Dan: Yes.

These are features that have come from ORTC. At each meeting we have tried to finalize the list, but new proposals continue to creep in. Within 6 months we will know whether the chairs have been able to hold the line on the most recent list agreed to in Seattle.

webrtcHacks: So is this why it is taking so long?

Dan: Yes.. The good news about it is that the features that are going in are the most requested ones from ORTC.

IP Leakage

webrtcHacks: The IP leakage issue was a hot topic on webrtcHacks and elsewhere? Many have labeled it as a flaw; other say this behaviour was by design? Can you share the “standards” perspective on this topic and the considerations that were discussed?

Dan: The summary is this – there are 2 problems with IP leakage:

One kind is the leakage of public addresses that the user doesn’t want leaked. This can happen when a user is using a VPN and not all of the traffic is sent over the VPN – a so called split tunnel VPN.  This is an issue if the user doesn’t want their non-VPN public address to be revealed. This is not a WebRTC problem; this is a split tunnel VPN problem. That doesn’t mean that people don’t blame the browser vendors even though it’s not their fault (laughs}

Technically any application running on your machine could do the same thing if you’re running a split tunnel VPN. There are extensions to turn off WebRTC for people who are very concerned about this.

The other kind of leakage is leakage of your local IP address. the reason this concerns some people is that it can be used to map the topology of your local network, say within an enterprise. However it turns out that applications can use an XmlHttpRequest to do the same thing.  Despite that, the browser vendors are working on ways to turn off the reporting of these local addresses.

There will be more details coming up in an upcoming post on my site.

Dan talking to webrtcHacks guest author Alan Jonhston at the IIT-RTC show

What’s Apple Doing?

webrtcHacks: Now the only major browser vendor left is Apple. Can you comment on public participation by Apple?

Dan: It is clear that people from Apple are continue to follow the work, but they still don’t contribute.

webrtcHacks: Do you know if they contribute to other WG more actively.

Dan: Yes, Apple does contribute more actively in other WG within W3C.


webrtcHacks: Anything new with video codecs now that the market has had some time to react to the decision to include both VP8 & H.264 for browsers? How is the VP9 vs. H.265 and Alliance for Open Media (AOM) discussion changed the discussion?

Dan: The gauntlet has been thrown for the creation of free and open source video codecs. MPEG-LA needs to take notice that the media producers and distributors are serious about coming up with lower cost alternatives. This pressure just continually increases. The AOM is a prime example of that.

webrtcHacks: Has the Alliance for Open Media come up in standards discussion? In the past I know there was discussion of just allowing software codecs that could defined on the fly.

Dan: Codecs still need to be created.  The discussions of VP8 vs H.265 and VP9 vs. H.265 are not really technical discussions. They are all about intellectual property because of the cost of licensing the codecs. The issue is not being able to select a codec – the issue is having a codec that you want to choose.

One API change that is just gone in is being able to choose which codec of the browser supported ones to use.


webrtcHacks: Anything else to add?

Dan: I think we’re finally on a good track in respect to a path forward for ORTC and WebRTC and thus the inclusion of Microsoft as a true and complete WebRTC vendor eventually. We just need the feature inflow from ORTC to stop right now to be able to declare victory and move on.

I think this is evidence that the industry really does want this to happen.

I spoke with a number of people who talk to HTML developer groups and they all agree that even today no more than 50% of the developers have heard of WebRTC – still! It is likely that one reason for that is for many developers a technology isn’t real until it is in Internet Explorer or its successor – Edge.

So having Microsoft fully engaged on a plan that we can all agree on now is a good thing for everyone.


    “interviewer”:“chad hart“,
    “interviewee”:“Dan Burnett

Want to keep up on our latest posts? Please click here to subscribe to our mailing list if you have not already. We only email post updates. You can also follow us on twitter at @webrtcHacks for blog updates and news of technical WebRTC topics or our individual feeds @chadwallacehart, @victorpascual and @tsahil.

The post Are we There Yet? WebRTC standards Q&A with Dan Burnett appeared first on webrtcHacks.

Hello Chrome and Firefox, this is Edge calling

Thu, 10/15/2015 - 14:15

Chrome, Firefox, and Edge are all on the same party line. Image from Pillow Talk (1959)

For the first time, Chrome, Firefox and Edge can “talk” to each other via WebRTC and ORTC. Check the demo on Microsoft’s modern.ie testdrive.

tl;dr: don’t worry, audio works. codec interop issue…

Feature Interoperability Notes ICE yes Edge requires end-of-candidate signaling DTLS yes audio yes using G.722, Opus or G.711 codecs video no standard H.264 is not supported in Edge yet DataChannels no Edge does not support dataChannels

As a reader of this blog, you probably know what WebRTC is but let me quote this:

WebRTC is a new set of technologies that brings clear crisp voice, sharp high-definition (HD) video and low-delay communication to the web browser.

In order to succeed, a web-based communications platform needs to work across browsers. Thanks to the work and participation of the W3C and IETF communities in developing the platform, Chrome and Firefox can now communicate by using standard technologies such as the Opus and VP8 codecs for audio and video, DTLS-SRTP for encryption, and ICE for networking.

This description is taken from the early-2013 Chromium blog post that announced interoperability between Chrome and Firefox. And now Edge?


So we have interoperability – for audio calls.  It is just audio. No video interoperability yet. Now this is just an issue of all vendors implementing at least one common video codec:

  • Edge currently implements a Microsoft variant of H264 called H264UC which adds some features like SVC
    • Adding H264 is work in progress
    • While there is a VP9 decoder for playing videos, that is not usable for ORTC so don’t get too excited
    • See Bernard’s comments for more information
  • Chrome implements VP8; H264 is work in progress
  • Firefox implements VP8 and H264

Audio interoperability is currently using G.722 instead of Opus because Edge still prefers Silk and G.722 over Opus.


But wait, how can those browsers talk if they do not agree on APIs?

Well, I implemented the PeerConnection API on top of ORTC. The gory details can be found here as part of a pull request for adapter.js. It has undergone a quite critical review and improved as a result of that. This process also showed some issues in the ORTC specification. While there has always been the assumption that it would be possible to implement the PeerConnection API using the lower-level ORTC API, nobody had actually done it.

The functionality provided is limited. More than a single audio and video track has not been tested and, since this is using an SDP similar to what is specified in the Unified Plan draft would likely not be interoperable with Chrome. But this is sufficient for quite a number of applications that are simple enough not to benefit from ORTC natively.


Using this Javascript implementation, Edge will generate something that is close enough to the SDP used by the PeerConnection API:

v=0 o=thisisadapterortc 8169639915646943137 2 IN IP4 s=- t=0 0 m=audio 9 UDP/TLS/RTP/SAVPF 104 9 106 0 103 8 97 13 118 101 c=IN IP4 a=rtcp:9 IN IP4 a=rtpmap:104 SILK/16000 a=rtcp-fb:104 x-message app send:dsh recv:dsh a=rtpmap:9 G722/8000 a=rtcp-fb:9 x-message app send:dsh recv:dsh a=rtpmap:106 OPUS/48000/2 a=rtcp-fb:106 x-message app send:dsh recv:dsh a=rtpmap:0 PCMU/8000 a=rtcp-fb:0 x-message app send:dsh recv:dsh a=rtpmap:103 SILK/8000 a=rtcp-fb:103 x-message app send:dsh recv:dsh a=rtpmap:8 PCMA/8000 a=rtcp-fb:8 x-message app send:dsh recv:dsh a=rtpmap:97 RED/8000 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=rtpmap:101 telephone-event/8000 a=rtcp-mux a=ice-ufrag:lMRF a=ice-pwd:NR15fT4U6wHaOKa0ivn64MtQ a=setup:actpass a=fingerprint:sha-256 6A:D8:7D:05:1A:ED:DB:BD:6A:60:1A:BC:15:70:D1:6C:A1:D9:00:79:E5:5C:56:15:73:80:E2:82:9D:B9:FB:69 a=mid:nbiwo5l60z a=sendrecv a=msid:7E4272C7-2B6C-49BD-BF7A-A3E7B8DD44F5 D2945771-D7B4-4915-AC29-CEA9EC51EC9E a=ssrc:1001 msid:7E4272C7-2B6C-49BD-BF7A-A3E7B8DD44F5 D2945771-D7B4-4915-AC29-CEA9EC51EC9E a=ssrc:1001 cname:3s6hzpz1jj

Check the anatomy of a WebRTC SDP post to find out what each of these lines mean.

This allows quite a number of the WebRTC PeerConnection samples to work in Edge, just like many of the getUserMedia samples already work.

With that working, the next big challenge was browser interoperability. Would this underspecified blob of text be good enough to be accepted by Chrome and Firefox?

It turned out to be good enough. After adding ICE candidates on both sides the ice connection and DTLS states soon changed to completed and connected. Yay. In Chrome at least.
Firefox did not work because of trivial mistakes that took a while to figure out. But then, it just worked as well.

As far as I am concerned this shows the hard part, making ICE and DTLS interoperable, is solved. The rest is something for codec folks to work out. Not my area of interest

{“author”: “Philipp Hancke“}

Want to keep up on our latest posts? Please click here to subscribe to our mailing list if you have not already. We only email post updates. You can also follow us on twitter at @webrtcHacks for blog updates and news of technical WebRTC topics or our individual feeds @chadwallacehart, @victorpascual and @tsahil.

The post Hello Chrome and Firefox, this is Edge calling appeared first on webrtcHacks.

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

Mon, 10/12/2015 - 17:56

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 often). On the positive side, interoperability between Edge, Chrome, and Firefox on the audio side was proven within days by multiple parties. Much of ORTC is finding its way into the WebRTC 1.0 specification and browser implementations.

I was with Bernard Aboba, Microsoft’s WebRTC lead at the IIT Real Time Communications Conference (IIT-RTC) and asked him for an interview to cover the Edge implementation and where Microsoft is headed. The conversation below has been edited for readability and technical accuracy. The full, unedited audio recording is also available below if you would rather listen than read. Warning – we recorded our casual conversation in an open room off my notebook microphone, so please do not expect high production value.


We cover what exactly is in Edge ORTC implementation, why ORTC in the first place, the roadmap, and much more.

You can view the IIT-RTC ORTC Update presentation slides given by Bernard, Robin Raymond of Hookflash, and Peter Thatcher of Google here.

{“editor”, “chad hart“}

Micosoft’s Edge is hungry for WebRTC

Intro to Bernard

webrtcHacks: Hi Bernard. To start out, can you please describe your role at Microsoft and the projects you’ve been working on? Can you give a little bit of background about your long time involvement in WebRTC Standards, ORTC, and also your new W3C responsibilities?

Bernard: I’m a Principal Architect at Skype within Microsoft, and I work on the Edge ORTC project primarily, but also help out other groups within the company that are interested in WebRTC. I have been involved in ORTC since the very beginning as one of the co-authors of ORTC, and very recently, signed up as an Editor of WebRTC 1.0.

webrtcHacks:  That’s concurrent with some of the agreement around merging more of ORTC into WebRTC going forward. Is that accurate?

Bernard: One of the reasons I signed up was that I found that I was having to file WebRTC 1.0 API issues and follow them. Because many of the remaining bugs in ORTC related to WebRTC 1.0, and of course we wanted the object models to be synced between WebRTC 1.0 and ORTC, I had to review pull requests for WebRTC 1.0  anyway, and reflect the changes within ORTC.  Since I had to be aware of WebRTC 1.0 Issues and Pull Requests to manage the ORTC and Pull Requests, I might as well be an editor of WebRTC 1.0.

Bernard Aboba of Microsoft and Robin Raymond of Hookflash discussing ORTC at the IIT Real Time Communications Conference (IIT-RTC)

What’s in Edge

webrtcHacks:  Then I guess we’ll move on to Edge then. Edge and Edge Preview are out there with varying forms of WebRTC. Can you walk through a little bit of that?

Bernard: Just also to clarify for people, Edge ORTC is in what’s called Windows Insider Preview.  Windows Insider Preview builds are only available to people who specifically sign up to receive them.  If you sign up for the Windows Insider Preview program and install the most recent build 10547, then you will have access to the ORTC API in Edge. In terms of what is in it, the audio is relatively complete. We have:

  • G.711,
  • G.722,
  • Opus,
  • Comfort Noise,
  • DTMF, as well as the
  • SILK codec.

Then on the video side, we have an implementation of H.264/SVC, which does both simulcast and scalable video coding, and as well as forward error correction (FEC), known as H.264UC. I should also mention, we support RED and forward error correction for audio as well. 

That’s what’s you will find in the Edge ORTC API within Windows Insider Preview, as well as support for “half-trickle” ICE, DTLS 1.0, etc.

webrtcHacks: I’ll include the slide from your presentation for everyone to reference because there’s a lot of stuff to go through. I do have a couple of questions on a few things for follow up. One was support on the video side of things for. I think you mentioned external FEC and also talked about other aspects of robustness, such as retransmission?

Bernard’s slide from IIT-RTC 2015 showing Edge’s ORTC coverage

Bernard: Currently in Edge ORTC Insider Preview, we do not support generic NACK or re-transmission.  We do support external forward error correction (FEC), both for audio and video.   Within Opus as well as SILK we do not support internal FEC, but you can configure RED with FEC externally.  Also, we do not support internal Discontinuous Operation (DTX) within Opus or SILK, but you can configure Comfort Noise (CN) for use with audio codec, including Opus and SILK.

Video interoperability

webrtcHacks: Then could you explain H.264 UC? The majority of the people out there that aren’t familiar with the old Lync or Skype for Business as it is now called.

Bernard: Basically, H.264 UC supports spatial simulcast along with temporal scalability in H.264/SVC, handled automatically “under the covers”.  These are basically the same technologies that are in Hangouts with VP8.   While the ORTC API offers detailed control of things like simulcast and SVC, in many cases, the developer just basically wants the stack to do the right thing, such as figuring out how many layers it can send. That’s what H.264UC does.  It can adapt to network conditions by dropping or adding simulcast streams or temporal layers, based on the bandwidth it feels is available. Currently, the H.264UC codec is only supported by Edge.

webrtcHacks:  Is the base layer H.264?

Bernard: Yes, the base layer is H.264 but RFC 6190 specifies additional NAL Unit types for SVC, so that an implementation that only understands the base layer would not be able to understand extension layers.  Also, our implementation of RFC 6190 sends layers using distinct SSRCs, which is known as Multiple RTP stream Single Transport (MRST).  In contrast, VP8 uses Single RTP stream Single Transport (SRST).

We are going to work on an implementation of H.264/AVC in order to interoperate.  As specified in RFC 6184 and RFC 6190, H.264/AVC and H.264/SVC have different codec names.

webrtcHacks:  For Skype, at least, in the architecture that was published, they showed a gateway. Would you expect other people to do similar gateways?

Bernard: Once we support H.264/AVC, developers should be able to configure that codec, and use it to communicate with other browsers supporting H.264/AVC.  That would be the preferred way to interoperate peer-to-peer.  There might be some conferencing scenarios where it might make sense to configure H.264UC and have the SFU or mixer strip off layers to speak to H.264/AVC-only browsers, but that would require a centralized conferencing server or media relay that could handle that. 


webrtcHacks:  What can you can you say about the future roadmap? Is it basically what’s on the dev.modern.ie page?

Bernard: In general, people should look at the dev.modern.ie web page for status, because that has the most up to date. In fact, I often learn about things from the page. As I mentioned, the Screen Sharing and Media Recorder specifications are now under consideration, along with features that are in preview or are under development.  The website breaks down each feature.  If the feature is in Preview, then you can get access to it via the Windows Insider Preview.  If it is under development, this means that it is not yet in Preview.  Features that are supported have already been released, so if you have Windows 10, you should already have access to them. 

Slide from Bernard’s IIT-RTC 2015 presentation covering What’s in Edge

In terms of our roadmap, we made a roadmap announcement in October 2014 and are still executing on things such as H.264, which we have not delivered yet.  Supporting interoperable H.264 is about more than just providing an encoder/decoder, which we have already delivered as part of H.264UC.  The IETF RTCWEB Video specification provides guidance on what is needed to provide interoperable H.264/AVC, but that is not all that a developer needs to implement – there are aspects that are not yet specified, such as bandwidth estimation and congestion control.

Beyond the codec bitstream, RTP transport and congestion control there are other aspects as well.  For example, I mentioned robustness features such as Forward Error Correction and Retransmission.   A Flexible FEC draft is under development in IETF which will handle burst loss (distances greater than one).  That is important for robust operation on wireless networks, for both audio and video.  Today we have internal FEC within Opus, but that does not handle burst loss well.

webrtcHacks: Do you see Edge pushing the boundaries in this area? 

Bernard: One of the areas where Edge ORTC has advanced the state of the art is in external forward error (FEC) correction as well as in statistics.  Enabling external FEC to handle burst loss, provides additional robustness for both audio and video.  We also support additional statistics which provide information on burst loss and FEC operation.  What we have found is that burst loss is a fact of life on wireless networks, so that being able to measure this and to address it is important. The end result of this work is that Edge should be more robust than existing implementations with respect to burst loss (at least with larger RTTs where retransmission would not be available).  We can also provide burst loss metrics, which other implementations cannot currently do.  I should also mention that there are metrics have been developed in the XRBLOCK WG to address issues of burst loss, concealment, error correction, etc.


webrtcHacks:  You have been a long time advocate for ORTC. Maybe you can summarize why ORTC was a good fit for Edge? Why did you start with that spec versus something else? What does it enable you to do now as a result?

Bernard: Some of the advantages of ORTC were indeed advantages, but in implementation we found there were also other advantages we didn’t think of at the time.


Bernard: ORTC doesn’t have SDP [like WebRTC 1.0]; the irony is ORTC allowed us to get to WebRTC 1.0 compatibility and interoperability faster than we would have otherwise. If you look at the adapter.js, it’s actually interesting to read that code- the actual code for Edge is actually smaller than for some of the other browsers. One might think that’s weird – why would it take less adaptation for Edge than for anything else? Are we really more 1.0 compatible than 1.0? The answer is, to some respects, we are, because we don’t generate SDP than somebody needs to parse and reformat. It certainly saves a lot of development to not have to write that code and have control in JavaScript, and also be easy to modify in case people find bugs in it.

The irony is ORTC allowed us to get to WebRTC 1.0 compatibility and interoperability faster than we would have otherwise

Connection State Details

The other thing we found about ORTC that we didn’t quite understand early on was it gives you detailed status of each of the transports- each of your ICE transports. Particularly when you’re dealing with situations like multiple interfaces, you actually get information about failure conditions that you don’t get out of WebRTC 1.0. 

It’s interesting to look at 1.0 – one of the reasons that I think people will find the objects interesting in 1.0 is because you actually need that kind of diagnostic information. The current connection state [in the current WebRTC] is not really enough – it’s not even clear what it means. It says in the spec that it’s about ICE, but it really combines ICE and DTLS. With the object model, you know exactly what ICE transport went down or if DTLS is in some weird state. Actually for diagnostics, details of the connection state is actually pretty important. It’s one of the most frequently requested statistical things. That was a benefit we didn’t anticipate, that we found is pretty valuable and will be coming into 1.0.

Many simple scenarios

Bernard: Then there were the simple scenarios. Everyone said, “I don’t need ORTC because I don’t do scalable video coding and simulcast” Do you ever do hold? Do you ever do changing owners of codecs? All illustrations that Peter [Thatcher] showed in his WebRTC 1.0 presentation. The answer is, a lot of those things are, in fact, common, and were not possible in 1.0. There is a lot of fairly basic benefits that you get as well. 

How is Edge’s Media Engine built

webrtcHacks:  In building and putting this in the Edge, you had a few different media engines you could choose from. You had the Skype media engine and a Lync media – you combine them or go and build a new one. Can you reveal the Edge media architecture and how you put that together?

Bernard: What we chose to do in Skype is move to a unified media engine. What we’ve done is, we’ve added WebRTC capabilities into that media engine. That’s a good thing because, for example, things like RTCP MUX and things like BUNDLE are now part of the Skype media engine so we can use them. The idea was to produce something that was unified and would have all the capabilities in one. It took a little bit longer to do it that way, but the benefit is that we get to produce a standardized compliant browser and we also get to use those technologies internally. Now we do not have 3 or 4 different stacks that we would have to rationalize later.

right now, our focus is very much on video, and trying to get that more solid, and more interoperable

Also, I should mention that one thing that is interesting about the way we work is we produce stacks that are both client and server capable. We don’t just produce pure client code that wouldn’t, for example, be able to handle load. Some of those things can go into back-end components as well. That is also true for DTLS and all that. Whether or not we use all those things in Skype is another issue, but it is part of the repertoire for apps. 

More than Edge

webrtcHacks: Is there anything else that’s not on dev.modern.ie that is exposed that a developer would care about? Any NuGet packages with these API’s for example?

Bernard: There is a couple of things. dev.modern.ie does not cover non-browser things in Windows platform. For example, currently we support DTLS 1.0. We do want to support 1.2, because there’s additional cipher  suites that are important. For example, the Elliptic Curve stuff we’re seeing going into all the browsers. I think Mozilla already has it, or Chrome has it, or if they don’t, they will very soon. That is actually very important. Elliptic Curve turned out to be more than just a cipher suite issue – the time and effort it takes to generate more secure certificates is large. For RSA-2048 you can actually block the UI thread if you thread the object. Anyway, those are very important things that we don’t cover on dev.modern.ie, but those are the things we obviously have to do. 

There’s a lot of work and a lot of thinking that’s been going on in the IETF if relating to ICE and how to be better for mobile scenarios. Some of that I don’t think is converged yet, but there’s a new ICE working group. Some of that is in the ortc-lib implementation yet. Robin [Raymond] likes to be on the cutting edge so he has done basically the first implementation of a lot of those new technologies. That’s something, I think is of general interest – particularly as ORTC moves to mobile.

I should mention, by the way, that the Edge Insider Preview was only for desktop. It does not run on Windows Phone just to clarify that. 

webrtcHacks:  Any plans for embedding the Edge ORTC engine as a IE plugin?

Bernard: An external plugin or something?

webrtcHacks:  Yeah, or a Microsoft plugin for IE that would implement ORTC. 

Bernard: Basically at this point, IE is frozen technology. All the new features, if you look on the website, they all go into Edge. That’s what we’ve been developing for. I never say Microsoft will never do anything, but currently that’s not the thinking. Windows 10 for consumers is a free upgrade. Hopefully, people will take advantage of that and get all the new stuff, including Edge.

Is there an @MSEdgeDev post on the relationship between this and InPrivate? pic.twitter.com/bbu0Mdz0Yd

— Eric Lawrence (@ericlaw) September 22, 2015

A setting discovered in Internet Explorer that appears to address the IP Address Leakage issue. Validating ORTC

webrtcHacks:  Is there anything you want to share?

Bernard: I do want to clarify a little bit, I think adapter.js is a very important thing because it validates our original idea that essentially WebRTC 1.0 could be built into the JavaScript layer with ORTC. 

webrtcHacks:  And that happened pretty quick – with Fippo‘s help. Really quick. 

Bernard: Fippo has written all the pull requests. We’re paying a lot of attention to the bugs he’s finding. Obviously, he’s finding bugs in Edge, which hopefully we’ll fix, but he’s also finding spec bugs. It really helps make sure that this compatibility that we’ve promised is actually real. It’s a very interesting process to actually reduce that to code so that it’s not just a vague promise. It has to be demonstrated in software.

Of course what we’ve done is currently with audio. We know that video is more complicated, particularly as you start adding lots and lots of codecs to get that level of compatibility. I wouldn’t say that when Fippo is down with audio that it will be the last word. I think we’ll have to  pay even more attention to interoperability stuff in the video cases. It will be interesting because video is a lot more complicated. 

adapter.js is a very important thing because it validates our original idea that essentially WebRTC 1.0 could be built into the JavaScript layer with ORTC.

What does the Microsoft WebRTC team look like

webrtcHacks:  Can you comment on how big the time is that’s working on ORTC in Edge? You have a lot of moving pieces in different aspects … 

Bernard: There’s the people in Edge. There’s the people in Skype. In the Windows system there’s the people on the S-channel team that worked on the DTLS. There’s people all over – for example, the VP9 work that we talked about, was not done by either Skype or the conventional Edge people. It’s the whole Windows Media team. I don’t really know how to get my hands around this, because if you look at all the code we’re using, it’s written by probably, I don’t know, hundreds and hundreds of people. 

webrtcHacks:  And you need to pull it together for purposes of WebRTC/ORTC, is that right?

Bernard: Yeah. We have to pull it together, but there’s a lot there. There’s a lot of teams. There will probably be more teams going forward. People say, “Why don’t you have the datachannel”? The dataChannel isn’t something that would be in Skype’s specific area of expertise. That’s a transfer protocol, it should be really written by people who are experts in transfer protocol, which isn’t either Edge or Skype. It’s not some decision that was made by either of our groups not to do it. We have to find somebody who proves that they can do that work, to take ownership of that. 

Feedback please

webrtcHacks:  Any final comments?

Bernard: No. I just encourage people to download the preview, run it, file bugs, and let us know what you think. You can actually can vote on the website for new features, which is cool. 

We do listen to the input. WebRTC is an expanding thing. There’s a ton of things you can do – there’s all that stuff on dev.modern.ie site and then there’s internal improvement. Getting a sense of priority is what’s most important to people, is not that easy, because there’s so much that you could possibly focus on. I’d say right now, our focus is very much on video, and trying to get that more solid, and more interoperable, at least for the moment. We can walk and chew gum at the same time. We can do more than just one thing. Conceivably, especially when you look at IE and other teams. 

webrtcHacks:  This is great and very insightful. I think it will be a big help to all the developers out there. Thanks!

    “interviewer”:“chad hart“,
    “interviewee”:“Bernard Aboba

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