With our fourth WebRTC Conference & Expo now a week behind us, we wanted to take some time to reflect on the event. There was never a dull moment for Team TokBox in Atlanta. We gave a keynote address, participated in several panel discussions, gave a live demo of our video driver and manned our bustling booth complete with a Bridgestone Golf B-FIT kiosk.
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.
Hello! Ed from the BD team @ TokBox here. We’re always thinking of great ways to showcase cool partners, so we came up with an idea for a series called PartnerTok. This whole series will be done via our open source chat tool OpenTokRTC and recorded with our archiving stack! For our inaugural episode we are featuring our friends at Cambly. They’re a language marketplace for people who want to learn English or Spanish. We talked to them about where the idea came from, how it got started, their business model as well as their experience launching the app.
Cambly is also one of the partners testing out our new archiving stack – you can hear from them firsthand in the video about how easy it is to implement. In fact, we used the OpenTok API for WebRTC to power the live interview, and our new Archiving & Playback beta to record it.
- 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.
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.
In April we announced our new Mantis multi-party infrastructure for Web RTC was available for pre-production trials. Since that time, hundreds of customers have logged minutes against our new infrastructure, powering multi-party OpenTok calls around the world.
We’re seeing customers connect foreign language students from across the world, build classrooms of sizes well past what off the shelf WebRTC could support, and experience more stability and quality in their multi-party conversations.
You could consider Mantis to be a media router, but in reality it is so much more than that. As the OpenTok platform grows and evolves to better solve the use cases our customers are building, the Mantis infrastructure is going to allow us to deliver a level of quality (starting with our traffic shaping algorithms that we released in June), product enhancements (such as archiving), and other capabilities that will take WebRTC to a new level.
There’s a new digital version of an old analog joke that starts something like this: “Two WebRTC engineers walk into a bar to have a beer while they talk about signaling.” The problem is that it’s the hotel bar at the Hotel California, and the punchline is that the engineers never get to leave.
Hardly a day goes by without another blog post about signaling and WebRTC.
Some people think signaling should be standardized; others think we already have the answer in SIP or REST. Some think that the lack of a signaling specification (beyond the need to support SDP offer/answer) is a huge gap in the WebRTC standard.
We think that leaving signaling out was the smartest thing that the key drivers of the standard could have done, for three reasons:
Over the last two years, OpenTok has continued to break ground as a live video platform.
As we’ve watched use cases evolve from basic social chat all the way up to supporting complex customer support calls, we’ve also discovered that partners need more than just live video communications – they need a way to orchestrate and communicate between the application endpoints. So today, we are exposing our signaling layer to OpenTok 2.0 developers so that you can piggyback on the distributed, scaled infrastructure that’s been proven to work over the last two years.
On October 1st, 2013 we will be launching new pricing for the OpenTok platform.
We are updating our pricing to reflect the cost of operating the OpenTok Platform on WebRTC versus older versions of our platform.
Using WebRTC technology, live video streams are commonly delivered at several times the data rate of a Flash video chat stream. Simply put, WebRTC consumes a lot more bandwidth than Flash, which can affect our operating costs. While WebRTC is a free and open-sourced standard, it doesn’t include the back-end infrastructure required to operate a live video communications application in the real world.
That’s what we provide with OpenTok – a global platform that offers the advanced features, capabilities, and back-end infrastructure that make WebRTC viable for commercial applications. We believe that our new pricing structure fairly reflects our underlying delivery costs while delivering terrific value for our customers and partners.
WebRTC 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: