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webrtcH4cKS: ~ WebRTC Externals – the cross-browser WebRTC debug extension

Posted by Philipp Hancke on July 28, 2017
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I am a big fan of Chrome’s webrtc-internals tool. It is one of the most useful debugging tools for WebRTC and when it was added to Chrome back in 2012 it made my life a lot easier. I even wrote a lengthy series of blog post together with Tsahi Levent-Levi describing how to use it to debug issues recently.

Firefox has a similar about:webrtc page which shows the local and remote SDP for each page as well as a very useful grid of ICE candidates. But unlike Chrome it does not show the exact order of API calls or nice graphs obtained from the getStats API. I miss both features dearly. Edge and Safari don’t support similar debugging helpers currently either.

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webrtcH4cKS: ~ Reeling in Safari on WebRTC – A Closer Look at What’s Supported

Posted by Chad Hart on June 19, 2017
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Long have WebRTC developers waited for the day Apple would come around to WebRTC. It has not been simple for web developers and Apple due to their policy that requires web browsing functionality to use the WebKit engine along with Safari. This mean no WebRTC in Safari, no Firefox or Chrome WebRTC on iOS, no native WebView with WebRTC or iOS API’s (but plenty of 3rd party ones). Despite community efforts and active development inside the WebKit project, it was not entirely clear when there would be at launch. That changed earlier this month when Apple announced a WebRTC-enabled WebKit based on the Google-backed webrtc.org engine was coming to both High Sierra – the next version of OSX – and iOS 11. Even better, WebRTC is available today as part of the free Safari Technology Preview.

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webrtcH4cKS: ~ Am I behind a Symmetric NAT?

Posted by Philipp Hancke on May 15, 2017
Posted in: Guide.
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NATs can be a nuisance for VoIP, particularly Symmetric NATs . Fortunately WebRTC includes tools for dealing with them. Image source: http://pinktentacle.com/

WebRTC establishes peer-to-peer connections between web browsers. To do that, it uses a set of techniques known as Interactive Connectivity Establishment or ICE. ICE allows clients behind certain types of routers that perform etwork Address Translation, or NAT, to establish direct connections. (See the WebRTC glossary entry for a good introduction.) One of the first problems is for a client to find what its public IP address is. To do so, the client asks a STUN server for its IP address.

NATs are boxes (physical or virtual) that connect our local private networks to the public internet. They do so by translating the internal IP addresses we use to public ones. They work differently from one another, which ends up requiring WebRTC to rely on both STUN and TURN in order to connect calls. For background on these, check out some of our past posts on this topic like this one and this one.

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webrtcH4cKS: ~ Debugging VP8 is more fun than it used to be

Posted by Philipp Hancke on March 28, 2017
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insect

Editor Note: Fippo uses a lot of advanced WebRTC terms below – if you are a regular reader of this blog then don’t let that scare  you. Wireshark is a great tool for diagnosing media issues and inspecting signaling packets even if you’re not building a media server. {“editor”, “chad hart“}

Stuff breaks all the time and then you need to debug it. My favorite tool for this remains Wireshark as we have seen previously. Its fairly useful for debugging all the ICE and DTLS stuff but recently I’ve had to debug the media traffic itself.

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webrtcH4cKS: ~ New Windows into WebRTC with UWP: Q&A with Microsoft’s James Cadd

Posted by James Cadd on February 22, 2017
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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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webrtcH4cKS: ~ Chrome’s WebRTC VP9 SVC Layer Cake: Sergio Garcia Murillo & Gustavo Garcia

Posted by Sergio Garcia Murillo on February 14, 2017
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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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webrtcH4cKS: ~ Slack Does WebRTC Video – Here’s How (Gustavo Garcia)

Posted by Gustavo Garcia on December 15, 2016
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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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webrtcH4cKS: ~ How to limit WebRTC bandwidth by modifying the SDP

Posted by Nick Gauthier on October 26, 2016
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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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webrtcH4cKS: ~ WebRTC media servers in the Cloud: lessons learned (Luis López Fernández)

Posted by Chad Hart on September 9, 2016
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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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webrtcH4cKS: ~ Let’s Encrypt – how get to free SSL for WebRTC

Posted by Chad Hart on August 1, 2016
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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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