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webrtcH4cKS: ~ WebRTC media servers in the Cloud: lessons learned (Luis López Fernández)

Posted by Chad Hart on September 9, 2016
Posted in: Guide.
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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
Posted in: Guide.
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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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webrtcH4cKS: ~ Optimizing video quality using Simulcast (Oscar Divorra)

Posted by Chad Hart on June 16, 2016
Posted in: Technology.
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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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webrtcH4cKS: ~ Your Browser as a Audio Conference Server with WebRTC & Web Audio (Alexey Aylarov)

Posted by Chad Hart on May 20, 2016
Posted in: Guide.
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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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webrtcH4cKS: ~ Update: Anatomy of a WebRTC SDP (Antón Román)

Posted by Chad Hart on May 5, 2016
Posted in: Guide.
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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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webrtcH4cKS: ~ Sharpening the Edge – extended Q&A with Microsoft for RTC devs

Posted by Chad Hart on April 21, 2016
Posted in: Guide.
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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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webrtcH4cKS: ~ The Big Churn – learning from real usage stats (Lasse Lumiaho and Varun Singh)

Posted by Chad Hart on April 8, 2016
Posted in: Guide.
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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 your limited resources focusing on? No service can be perfect, so what is a reasonable error rate? These are all tough questions to answers without decent industry data.

Fortunately Lasse Lumiaho and Varun Singh from callstats.io have agreed to share some stats from their WebRTC monitoring service to help you answer these questions.  Their service does not monitor every WebRTC service,  but their 100+ customer base does provide a statistically meaningful sample to help give you some practical metrics for planning and comparison.

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webrtcH4cKS: ~ Is Slack’s WebRTC Really Slacking? (Yoshimasa Iwase)

Posted by Yoshimasa Iwase on March 24, 2016
Posted in: Technology.
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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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webrtcH4cKS: ~ Dear Slack: why is your WebRTC so weak?

Posted by Philipp Hancke on March 3, 2016
Posted in: Technology.
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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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webrtcH4cKS: ~ getUserMedia resolutions III – constraints unleashed

Posted by Chad Hart on February 8, 2016
Posted in: Guide.
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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 situation with Chrome was greatly improved, but Firefox at the time effectively only supported a single resolution so constraints were not much help. Specifically, I am interested in understanding what happens when you ask for a specific resolution. You might want to have a specific resolution returned by getUserMedia if you want to match the camera resolution to a specific video area to have a 1 to 1 pixel correlation, in a computer vision application where each pixel represents a distance, or if you are dealing with non-standard video devices.

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