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?
Today we’re excited to announce the launch of the OpenTok Agency Program. We’re looking for world-class agencies that are WebRTC and OpenTok platform experts.
Demand for real-time communications applications is growing rapidly. As a platform provider, we focus on our core business: creating a scalable, easy-to-use and capability-rich WebRTC platform. Sometimes, though, customers approach us seeking development assistance as they integrate live video into their website or mobile application or build a new project altogether. This is where our agency partners come into play. We need a group of highly skilled and responsive agencies that can help turn our customers’ OpenTok concepts into realities.
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.
If you’d like to try the OpenTok platform simply sign up for an account! Want to try our new Archiving & Playback beta? You can request access to sign up for the program here.
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.
With last week’s WebRTC Conference and Expo in Santa Clara, California coming to a successful conclusion, the second big WebRTC event of the year is now behind us. Sure, there are other WebRTC-related conferences – the IIT RTC conference in Chicago, the WebRTC Summit at Cloud Expo, next month’s WebRTC 2013 conference in Paris – but with what looked like 700 people in attendance, the twice-annual WebRTC Conference and Expo is the big one.
- 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.
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.