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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. Continue Reading

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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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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” SDP for the time being.

At a recent WebRTC Boston, Nick Gauthier of MeetSpace described how he used SDP modification and other techniques to jam up to 10 video callers into a single conference without a media server. Not everyone has a good reason to do this, but there are certainly plenty of applications where having more precise control of your bandwidth consumption would be useful. You can see his video here or check out his technique and thorough explanation on how to munge SDP to adjust individual bandwidth usage below.

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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 several community experts with deep expertise in this domain to help. One of those experts who has always been happy to share his learnings is past webrtcHacks guest author Luis López Fernández.

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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 own personal website is about to expire and I’m not too excited about paying $70/year to renew it. Fortunately, there is a new way to get certificates for free through Let’s Encrypt. Let’s Encrypt is a non-profit certificate authority that formed with the backing of many major industry players like Mozilla, Akamai, Cisco, and many others to simplify and automate the process of setting up encryption for your website. Oh, and its completely free.

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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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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:

  1. Full mesh – each client sends their audio to every other client; the individual streams are then combined locally before they come out of your speaker
  2. Mixed with a conferencing server acting as a Multipoint Control Unit (MCU) – the MCU combines each stream and sends a single set to each client
  3. Routed with a conferencing server in a Selective Forwarding Unit (SFU) mode – each client sends a single stream to the server where it is replicated and sent to the others

This architecture represents a fourth type: client-mixed type where one of the clients acts like the server. This provides the server-less benefits of mesh conferencing without the excessive bandwidth usage and stream management challenges.

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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 update.

We also had some rendering issues on the interactive guide. After failing to figure out how to fix it, I decided to completely rewrite it. It is still has some issues, so please make your pull requests to fix and update it on our github repo here.

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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 at the link above or read the follow-up post for details. Then last week Microsoft announced plug-in free Skype on the Edge browser.

I had some questions; Fippo had some questions; so we asked Bernard if he could publicly respond here. It turned out Bernard and his teammate on the Edge Browser team, Shijun Sun, were building a running list of questions they wanted to address too. Here it is.

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