Walkthrough

All posts tagged Walkthrough

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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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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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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No thoroughfare

“Only Secure Origins Are Allowed”

    – Chrome 47

Chrome 47 now forces secure origins (mostly) with HTTPS. This can be a pain to deal with, but Xander Dumaine is here to help with some guidance. Xander is a Senior Software Engineer who deals with the good and bad of WebRTC for Interactive Intelligence in Raleigh, NC. He is helping maintain simpleWebRTC and organises the Triangle WebRTC Meetup group in that area.

{“editor”: “chad hart“}

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Speak No Evil

A few days back my old friend Chris Koehncke, 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.

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ORTC support in Edge has been announced today. A while back, we saw this on twitter:

“This release [build 10525] lays the groundwork for ORTC” was quite an understatement. It was considered experimental and while the implementation still differs from the specification (which is still work in progress) slightly, it already worked and as a developer you can get familiar with how ORTC works and how it is different from the RTCPeerConnection API.
If you want to test this, please use builds newer than 10547. Join the Windows Insider Program to get them and make sure you’re on the fast ring.

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It turns out people like their smartphone apps, so that native mobile is pretty important. For WebRTC that usually leads to venturing outside of JavaScript into the world of C++/Swift for iOS and Java for Android. You can try hybrid applications (see our post on this), but many modern web apps applications often use JavaScript frameworks like AngularJS, Backbone.js, Ember.js, or others and those don’t always mesh well with these hybrid app environments.

Can you have it all? Facebook is trying with React which includes the ReactJS framework and  React Native for iOS and now Android too. There has been a lot of positive fanfare with this new framework, but will it help WebRTC developers? To find out I asked VoxImplant’s Alexey Aylarov to give us a walkthrough of using React Native for a native iOS app with WebRTC.

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The fact that you can use WebRTC to implement a secure, reliable, and standards based peer-to-peer network is a huge deal that is often overlooked.  We have been notably light on the DataChannel here at webrtcHacks, so I asked Arin Sime if would be interested in providing one of his great walkthrough’s on this topic.  He put together a very practical example of a multi-player game.  You make recognize Arin from RealTime Weekly or from his company Agility Feat or his new webRTC.ventures brand. Check out this excellent step-by-step guide below and start lightening the load on your servers and reducing message latency with the DataChannel.

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The “IP Address Leakage” topic has turned into a public relations issue for WebRTC. It is a fact that the WebRTC API’s can be used to share one’s private IP address(es) without any user consent today. Nefarious websites could potentially use this information to fingerprint individuals who do not want to be tracked. Why is this an issue? Can this be stopped? Can I tell when someone is trying to use WebRTC without my knowledge? We try to cover those questions below along with a walkthrough of a Chrome extension that you can install or modify for yourself that provides a notification if WebRTC is being used without your knowledge.

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