This post was co-authored by Gustavo Garcia Bernardo, Philipp Hancke and Charley Robinson.
When WebRTC stuff is really broken, it gets fixed very quickly.
Early in December 2015, shortly after the release of Chrome 47 to the general public, we started to notice a subtle and strange behavior in the Audio/Video of streams during our many daily meetings using WebRTC: the video occasionally wouldn’t stay caught up with the corresponding audio. As with many bugs noticed internally by developers, it took a while for any of us to believe that what we were seeing was a real issue. We call this the inverse of productive dogfooding: rather than assume we are just like our users, we can just as easily decide we are nothing like them.
Have you ever had to support a WebRTC application and needed to get packet dumps from the user? Wireshark is a great tool for this, but asking a user to install it and make a dump rarely works. It’s just too complicated. So I was pretty excited when I read the Chrome 49 release notes which described (not in much detail) a new feature called the ‘RTC event log’. This is described as follows:
We now provide a new debug option in chrome://webrtc-internals for tracing internal details (e.g., BWE, jitter buffer state) for audio and video sessions. This option creates a log containing the timing and headers of packets as well as the timing of various internal events. We hope this will help resolve issues related to media transport and jitter buffers; attaching this log when reporting such issues will help us tremendously.
Early December saw the roll-out of Chrome 47. When doing anything with WebRTC, this is always an interesting time. A release brings new features or may break things, like removing the getUserMedia functionality for insecure origins.
Our metrics clearly track such roll outs as seen below:
At our September TechTok WebRTC consultant and analyst, Tsahi Levent-Levi, came along to discuss the power plays of the video coding industry.
Just when we thought we were done with the video codec wars in WebRTC – we found out we’re only just beginning. Tsahi was here to talk us through some important questions – how is this WebRTC codec war going to play out? Where do the alliances lie? And where are we headed?
Watch the full video here:
UPDATE 9/16/15: After we published this post Google announced that they are pushing back the release date of the HTTPS security change. They’re estimating that it will now be released to production in December 2015.
Google recently announced a security policy change that will impact future versions of the Chrome browser. Any website which has integrated geolocation technology, screen-sharing, WebRTC and more, will now be required to be served from a secure (HTTPS) site.
We at TokBox are supportive of this update as we believe operating your WebRTC-based applications through HTTPS is a best practice and offers enhanced security. In addition, it has the added bonus of improving your end-user experience by remembering their hardware settings, and even allowing screen-sharing (if you choose to implement it).
The Internet has fundamentally reshaped the world of work. Emails, instant messaging and video conferencing first transformed the way we communicate, and today as we increasingly move into the cloud, nearly every aspect of work takes place online in a globally connected 24/7 environment.
The physical office space as we knew it has become less relevant and an increasing number of employees are working from home or remote locations. Startups and even the most well established global firms, are realizing the benefits of flexible work and are embracing communication tools that enable teams to connect and collaborate effectively online (eg. Slack, DropBox).
Last week TokBox hosted the monthly SF WebRTC meet up at our offices in San Francisco.
It was a great evening, with a range of speakers and topics from the WebRTC world. This month we heard from:
- Ankur Oberoi from Tokbox
- Hadar Weiss from Peer5
- Feross Aboukhadijeh from WebTorrent, PeerCDN
- Dr Alex from Temasys
You can watch a full recording of the event below and if you are interested in hearing more about meet up events at TokBox, you can join our meet up group here.
WebRTC is maturing and we can see the needs in the market evolving along with this.
However, with the increased need for rich, digital experiences comes the challenge of building more advanced applications. We know that building real-time video communications can be challenging, especially when it involves more than two participants. To pull off a multi-party call using WebRTC off-the-shelf you’ll need a strong backend infrastructure and a deep understanding of media processing. That’s why we are looking forward to exploring this topic with WebRTC expert, Tsahi Levent-Levi, founder of bloggeek.me, in our upcoming webinar.
We’ve all seen the statistics and now know that mobile usage is at an all time high and is still on the rise. Increasing with this is mobile video consumption. As mobile data has become more affordable and reliable, and with more WiFi hotspots popping up, it is becoming increasingly easy to watch, share and communicate via video on mobile devices.
This is something that our partners at Simpleweb, a web and mobile development agency in the UK, have noticed too. They know that a new wave of apps is revolutionizing the way we view and broadcast video online, many of which are powered by WebRTC.
Mozilla has released further enhancements to Firefox Hello, powered by OpenTok, including the ability to screen share within a video chat. This new feature allows participants to open and share a browser tab or application window from within the chat, making browsing, shopping, drafting or any other activity more collaborative and engaging.