Technology

Imaged modified from Window in Seattle airport by Flickr user Robert Scoble (CC BY-2.0)

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 addition to the couple 3rd party WebRTC libraries for WebRTC, Edge & Skype support for WebRTC/ORTC, Microsoft’s has had a few other less known and non-public WebRTC projects in the works.  Last week they publicly launched WebRTC for Universal Windows Platform (UWP), providing WebRTC support for another huge chunk of the world’s developers. I asked, James Cadd, Microsoft’s Program Manager in the Windows Developer Platform Group in charge of the project to share some details.

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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. Sergio Garcia Murillo is a long time media server developer and founder of Medooze. Most recently, and most relevant for this post, he has been working on an open source SFU that leverages VP9 and SVC (the first open source project to do this that I am aware of). In addition, frequent webrtcHacks guest author and renown video expert Gustavo Garcia Bernando joins him. ...

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Slack is an über popular and fast growing communications tool that has a ton of integrations with various WebRTC services. Slack 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 there any tricks we can learn from their implementation? Long time WebRTC expert and webrtcHacks guest author Gustavo Garica takes a deeper dive into Slack’s new video conferencing feature below to see what’s going on under the hood. ...

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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 here at webrtcHacks in the past and it is time we gave it more dedicated attention.

To do that we asked Oscar Divorra Escoda, Tokbox’s Senior Media Scientist and Media Cloud Engineering Lead to walk us through it. Tokbox was one of the first to market with a SFU and Oscar shares some of his learnings below. ...

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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? How does multi-party calling work? How exactly is Slack using the Janus gateway? Fortunately WebRTC has an awesomely active and capable community that quickly picked up the slack (pun intended). ...

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slack webrtc2

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 with WebRTC and not support Firefox like this:
 

Maybe you had your reasons. As usual, I tried to get a dump from chrome://webrtc-internals to see what is going on. Thanks to Dag-Inge Aas for providing one. The most interesting bit is the call to setRemoteDescription:

I would like to note that you reply to Chrome’s offer of UDP/TLS/RTP/SAVPF with a profile of RTP/SAVPF. While that is still tolerated by browsers, it is improper.
Your a=msid-semantic line looks very interesting. “WMS janus”. Sounds familiar, this is meetecho’s janus gateway (see Lorenzo’s post on gateways here). Which by the way works fine with Firefox from what I hear. ...

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I Think I'm Being Watched

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.

Claims

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

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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 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. ...

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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 use the RTP protocol for transferring media. RTP uses a packet header which is not encrypted and contains a number of attributes such as the payload type (identifying the codec used), a synchronization source (which identifies the source of the stream), a sequence number and a timestamp. This allows routers to identify RTP packets and prioritize them. This also allows someone monitoring all network traffic (“Pervasive Monitoring“) to easily identify VoIP traffic. Or someone wiretapping your internet connection. ...

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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. What benefit does the open source code provide if you can’t figure out how to build from it?

As WebRTC matures into mobile, native desktop apps, and now into embedded devices as part of the Internet of Things, working with the lower-level source code is becoming increasingly common. ...

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