The availability of off-the-shelf real-time video chat SDKs and cloud services makes developing communication apps and websites almost too easy; anyone who sits down and focuses on building an app or website can create a halfway decent finished product in a weekend or less.
Not all SDKs and APIs are created equal, however. Apps quickly put together with templates and free off-the-shelf code might look comparable with their more polished counterparts, but in subtle and meaningful ways they’re not. Key areas that separate business-ready apps from quickly-built creations are reliability and quality of experience. This is especially true when it comes to managing real-time video and audio.
While WebRTC by itself is good for small projects, however, it isn’t necessarily fully business-ready out of the box. It relies on the quality of peer-to-peer networks for transmission, and it suffers when there’s poor connectivity at the network’s edge. Try running WebRTC over a poor cellular connection and the results are uninspiring at best and unfeasible at worst.
Most of us accept poor audio and real-time video chat quality when we’re working from a remote location or our smartphone; anyone who uses Skype has been trained to accept latency, connection issues and picture degradation periodically.
Inconsistent connectivity is fine when we’re chatting with a friend in Cameroon or sharing the Hagia Sophia live with the family, as I did during a recent trip to Turkey. Unreliable connectivity is not acceptable in a business app that relies on audio or video, however. Think telemedicine, for instance, or e-classroom situations.
Some hospitals today now use video to include specialists from other parts of the country when they can’t afford to have these medical experts on staff. This is a brilliant use of videoconferencing, and a study by U.C. Davis last year showed that telemedicine could save rural emergency departments an average of $4,662 per use. Certainly there is no margin in an emergency room for inconsistent video connectivity, though.
Likewise, real-time communications can play an important role in education, both in the classroom and as the foundation for informal learning communities. HelloTalk, for instance, connects language learners from around the world and helps them brush up on their language of choice via one-on-one, video chat. An e-classroom where many people are meeting at the same time–or a service like HelloTalk that’s built around real-time video chat connectivity—won’t be effective if the video is not reliable when teachers and students need it.
That’s why business-ready apps stress stability and reliability. A research or personal project can get away with choppy video from time to time, but a telemedicine app cannot. For these business-ready applications, video chat quality-of-experience (QoE) matters. Areas such as video QoE are the little details with huge implications that separate the amateur from the professional.
Providing the same easy-to-use real-time communications functionality of WebRTC but with professional-grade video QoE is what Agora.io is all about.
Agora.io uses more than 70 globally-distributed data centers, smart routing and 24×7 network monitoring to ensure that the WebRTC connections that are run through the Agora global network hold up even in poor network connectivity environments. We both use our own patent-pending network transmission technology, as well as highly efficient real-time video chat and audio optimizations that we built from the ground up for reliable real-time communications.
We make it easy for developers at the same time, simplifying the use of WebRTC in apps and websites and offering a pay-as-you-go model so developers can add video QoE to their projects on an as-needed basis.
For small and less-critical projects, perhaps video QoE can be ignored. When real-time communications must perform, however, QoE becomes a big deal. At Agora.io, we help support those developers who need real-time audio and video that “just works”—reliably and with high quality.