TokBox OpenSSL Heartbleed Disclosure

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.

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Introducing Dynamic Frame Rate Controls

artificial intelligence. Image shot 2008. Exact date unknown.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!”

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Is WebRTC Ready for H.264?

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.

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Cisco announces open-sourced version of H.264

Cisco_logoIn 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 [1]. 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.

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Proof: the rise and rise of WebRTC

webrtc iconWebRTC is clearly a hot topic. But in an effort to discover just how hot we conducted what we think is one of the largest global surveys of its kind. Today, we are pleased to share the results with all of you in the TokBox and greater WebRTC community.

The study, which analyzed responses from 1,161 people across 11 countries, found rapidly emerging interest amongst larger organisations (1,000+ employees), and also found rapid WebRTC adoption amongst smaller companies (fewer than 500 employees) where more than one in four (27.1%) developers say WebRTC is already critical to their work.

Some of the other key findings:

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Welcome Chrome for Android – Now part of our WebRTC family

chrome for android logoWe’re incredibly pleased to announce that OpenTok on WebRTC supports Google’s just-released Chrome 29 for Android. This brings Android support formally into the OpenTok on WebRTC family, and is a big step forward in increasing the number of WebRTC-ready endpoints in market.

We’ve been working with the Chrome for Android beta builds over the last few months, making sure that OpenTok on WebRTC works properly – and transparently – in that environment. In fact, attendees at WebRTC Expo in Atlanta saw us demonstrating OpenTok applications running in Chrome on Nexus tablets at the beginning of the summer.

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WebRTC: Your thoughts requested

Hello TokBox Community,

We have a small favor to ask of you. We’ve pulled together a brief survey about WebRTC that aims to measure the current level of awareness, interest, and activity around the standard and we need your input:

https://www.surveymonkey.com/s/TokBoxWebRTCSurvey

The results will be made public and will reveal:

  • The depth of WebRTC knowledge in the tech community
  • Which features/functionality are considered most important to you
  • How the tech community would like to see the standard develop over time

Added bonus? We’re raffling off five $100 Amazon gift cards to people that have completed the survey (you’re only eligible to win one). So take a few minutes, ponder what WebRTC means to you, and answer our survey. Thanks!

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Firefox support opens up new world of WebRTC possibilities

firefox_logo-only_RGB-300dpi

We’re incredibly pleased to see Mozilla launch Firefox with WebRTC enabled by default.  With Mozilla’s Firefox joining the WebRTC family, millions of people will have the opportunity to experience high-quality plugin-free face-to-face video within web applications.

TokBox’s OpenTok platform provides APIs and infrastructure that make it incredibly easy for web and mobile developers to build and deploy WebRTC-powered video applications.  WebRTC enables Firefox to give Javascript access to your webcam and microphone, and supports powerful P2P networking protocols.

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Mantis: Next-generation Cloud Technology for WebRTC

OpenTok_allplatforms (1)Today we’re proud to announce our latest WebRTC innovation: Mantis, a cloud-scaling infrastructure for our OpenTok on WebRTC platform.

This is another big step forward for the TokBox team as we continue to pursue our goal of providing application developers with simple yet powerful APIs. APIs that not only leverage the latest standards to deliver the best possible experience, but that are backed by a scalable, smart cloud which supports interoperability across a variety of end-points.

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New changes for WebRTC in Chrome 26

A new version of Chrome is out, and with it changes in the WebRTC stack. We dug through the commit logs for Chrome 26, and found the following list of WebRTC bug fixes, enhancements, and updates that we thought were relevant to the OpenTok community:

Highlights

  • A lot of audio bugs in WebRTC were fixed dealing with crashes and non-standard audio bitrates
  • Chrome on Android can now be WebRTC-enabled by enabling a flag
  • Improvements to the connectivity stack in WebRTC
  • Ability to set media constraints for audio

Full list

  • Avoids crash in WebRTC audio clients for unsupported capture sample rates.
  • Avoids crash in WebRTC audio clients for 96kHz render rate on Mac OSX.
  • Enable webrtc build on android.
  • Set WebMediaPlayerMS network state to loading instead of loaded
    • This indirectly fixes the problem where WebRTC audio is muted upon refresh. The HTMLMediaElement will try to cache fully Loaded videos when the element is destructed. This will signal to the HTMLMediaElement that the player was destroyed when loading, so it needs to recreate WebMediaPlayerMS upon destruction of the media tag.
  • Allowing multiple MediaPlayers to connect to WebRtcAudioDeviceImpl by sharing one WebRtcAudioRenderer.
    •  The audio is gone when new PeerConnection is connecting to a media stream. What is happening is that the stream will pause the existing MediaPlayer and create new MediaPlayers to associated to it. But since we only allow one WebRtcAudioRenderer to connect to WebRtcAudioDeviceImpl, the new MediaPlayers audio won’t be able to associate to stream.

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