Here at TokBox we are always trying to expand and improve our range of features, tailoring these features in line with real world developer needs. That’s why we are excited to announce some pricing and storage updates.
We’re still receiving a lot of feedback on our beta so we’d like to extend a big ‘thank you’ to our community for this.
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.
While WebRTC has been innovating at an impressively rapid rate, the users of the web and mobile apps have been delighted with lots of new experiences. We’ve started connecting to people across different timezones, countries, and even continents in real time, on just about every sort of device. But when we ask developers, the people who dream up the next wave of crazy ideas, what they need in order to keep delighting their users we hear a few things over and over.
One of the most requested features of the platform that developers are patiently waiting for is WebRTC broadcasts at scale. The technical challenge is about getting the right stream (with the right bitrate, and the right encoding) out to all the different types of people who are watching, with their vastly different networks and bandwidth capabilities.
AirPair, a startup that offers live online consultations with programming experts, today announced partnerships with TokBox and a handful of other API companies. That means AirPair users will have direct access to OpenTok platform exports when they need it. When developers run into a bug, have questions, or need help with implementation, an OpenTok expert can help resolve their problems quickly, in real-time.
Interested in giving it a shot? Check out the TokBox Experts page on AirPair!
In spite of limited specification of anything beyond one-to-one audio and video calls in WebRTC, one of the most popular usages of this technology today is multiparty video conference scenarios. Don’t think just about traditional meeting rooms. There are different use cases beyond meeting rooms, including e-learning, customer support, or real time broadcasting. In each case, the core capability is being able to distribute the media streams from multiple sources to multiple destinations. So… if you are a service provider how can you implement a multi-party topology with WebRTC endpoints?
Obama Called. And We Responded.
Yesterday President Obama kicked off the Hour of Code Campaign for Computer Science Education Week 2013 with a inspiring video calling for every American to learn code.
Here at TokBox we are excited to help! In this post we will help you jump the next hurdle.
After learning the basics of web and/or mobile programming, most people get bogged down by technical complexity and knowledge.
Say, after building your first app, you want to add a feature to let users video chat with each other. Learning about real time video streaming itself, let alone implementing it, can take months! This is why we highly recommend playing with platforms and APIs after learning the basics of web/mobile programming. You will be able to put together interactive apps that you never thought were possible. For example, with just basic web and/or mobile programming knowledge, you can add live video chat/streaming to your web or mobile app with the right library.
There are many platforms out there that let you build technically difficult apps with basic programming knowledge. Here is a short list of our favorites that we have worked with at various hackathons. Because of these platforms, developers were able to use them and build amazing applications within 24 hours.
- Archiving—You can record audio-video streams in a session and download the recording as an MP4 file (with H.264 video and AAC audio).
- Dynamic frame rate control—This feature lets you reduce bandwidth usage of a Subscriber’s video stream. This reduces CPU usage and the network bandwidth consumed, and it lets you subscribe to more streams simultaneously.
These are just a couple of the new features to be included in version 2.2.
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!”
Connectivity, we at TokBox believe, is one of the cornerstones of real-time communication applications. So we are happy to announce that we now support TURN over TCP.
There are several technologies which are used to help establish connectivity in WebRTC. The first mechanism is using a protocol called STUN. STUN uses a ping-pong mechanism to find the public IP of a client end-point so that a peer-to-peer session can be established and one can traverse a firewall. While this is useful in a number of scenarios, there are cases where one could be behind symmetric NATs, where STUN does not suffice. TURN helps in these cases. TURN is a mechanism by which real-time media can be relayed through a TURN server to punch through firewalls. OpenTok seamlessly supports STUN and TURN so a developer doesn’t have to worry about how to setup up these servers, scale them, establish connectivity etc.
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.