TokBox has always believed in the power of WebRTC to change the way that people communicate in the digital world. Not just in browsers, but also on phones and other connected devices as well as the amazing devices of the future that we know are coming.
WebRTC is transformative.
During the past two years, the TokBox team has been working hard to advance the WebRTC standard for the development community. We’ve made the standard more accessible and actionable through the OpenTok platform, giving developers all the features and functionality they need to deliver a robust, scalable WebRTC-powered application or website in the real world.
A major vulnerability was uncovered yesterday which affects a majority of web service providers. The exploit is related to OpenSSL’s heartbeat extension which could enable a malicious attacker to access private keys. The bug has been present in OpenSSL since December 2011, and was brought to light yesterday. You can find more information about the exploit termed “Heartbleed” (CVE-2014-0160) here.
Our operations team reacted immediately to this and has taken the necessary steps to secure our infrastructure, ensuring the appropriate secure versions of OpenSSL are in place.
Today we’re announcing new Intelligent Quality Controls in the OpenTok platform. To catch everyone up, Intelligent Quality Controls are the features and enhancements we’re developing to make sure that each participant in a video call has the best possible experience.
Update (Nov 25): Developers, check out our new blog post that provides details on using dynamic frame rate controls.
You may recall that over the summer we launched traffic shaping for the audio-only fallback feature. This feature drops video in low bandwidth situations to prevent a participant with poor QOS from dragging down the video quality for everyone else. Essentially, we built the automatic (video) mute button for “that guy on his cell phone in a convertible!”
The long-running video codec debate has, without a doubt, been the biggest open issue in the WebRTC standards effort.
In a surprise announcement last week, Cisco introduced a mechanism through which H.264 could be used in WebRTC browser implementations free from MPEG-LA’s licensing burden.
Cisco’s maneuver was a master stroke from the playbook of open standards strategy. The licensing deal they announced with MPEG-LA appears to cut the legs out from under the main pragmatic argument opposing H.264 (ie. the royalty problem). Mozilla’s support lent Cisco’s approach instant credibility from the ideological wing (ie. the open source camp). And by keeping this under wraps until a week before the upcoming IETF 88 meeting, at which the video codec debate is to be revisited, Cisco left no time for any coordinated response from the VP8 camp.
In the last year we’ve witnessed VP8 proponents and H.264 proponents debate which codec should become “official” for WebRTC. The main points of contention? Licensing fees associated with H.264 make it unaffordable for a non-profits like Mozilla to support. In addition, VP8 isn’t compatible with existing and legacy video conferencing platforms which are typically built to support H.264.
We saw Google draw a line in the sand early on by announcing the “perpetual, worldwide, non-exclusive, no-charge, royalty-free, irrevocable” licensing of VP8. In addition, they recently moved their flagship video conferencing product, Google Hangouts, on to VP8.
Yesterday, Cisco unexpectedly announced that they will release an open-source version of the H.264 codec. The open-source version will include a free downloadable binary module that can be integrated into any application. All without the cost of licensing the codec . This is a strategic precursor to the IETF #88 next week where a vote will take place about the MTI (mandatory to implement) video codec for WebRTC, with the dominant front-runners being VP8 and H264.