VUC540 on Friday, May 8th, 2015 will feature Tim Panton, WebRTC guru, VUC regular and self-described protocol droid, detailing a recent pet project known as YoPet. YoPet is a WebRTC-based service built to allow pet owners a simple, secure way to video chat with their furry, scaly and/or feathered friends.
Yopet is comprised of a web service that connects the distant pet owner back to the home, where the pet has access to an Android device running the YoPet application. In the telling of this tale Tim will illuminate a variety of issues faced in developing the app & associated service.
As Randy is taking a vacation in Cuba, where internet access is extremely limited, I get to be pilot-in-charge for this little online misadventure. Now everyone please be seated, it could be a little bumpy on the climb to cruising altitude. Fear not, the unflappable Andy Smith of TruePhone will be in the co-pilot seat.
It’s been a year or more that tools like Google’s Hangouts have supported the ability to share a host computer screen with the viewing audience. This was rightfully heralded as “a very good thing indeed.” However, it’s current incarnation is considerably less than ideal and seems to be stalled. I’d like to lay out a challenge to see if anyone is interested into taking this to the next level, which is something that we’ve tried to do with a few VUC calls earlier this year.
Here’s the fundamental problem; people use screen sharing to give demos of software and share documents, which includes giving presentations a la PowerPoint, Keynote, etc. Currently, Hangouts, Jitsi Video Bridge and the like show either the screen share or the camera. In the case of slide presentations there can be very little activity in view as the presenter speaks to the points shown on the current slide. This creates less than compelling visuals.
Mr Prokop’s SIP Adventures blog has proven interesting, so I thought the podcast worth a listen. Sadly, while the host is presented full bandwidth, as might be expected from a local recording, the guest is presented in narrowband. Given that the subject matter is WebRTC I think that this is more than a little anachronistic. WebRTC-based services are in fact a very easy way to enjoy wideband audio for the purposes of producing a podcast.
Dragging the podcast in my trusty editor I find it to the a definitive example of full-band audio vs narrowband. The file is sampled at 44.1 kHz, so the top of the vertical axis is 22 kHz. The guests audio is a good quality PSTN call, but even that is quite a contrast from the host. This contrast is very jarring to the listener.
This morning’s inbox included an announcement from Voxbone about their own WebRTC service. Based in Belgium Voxbone are a provider of various services to telecom companies. They are most commonly known for providing DIDs in 50+ countries around the world. This allows companies that require multi-national connectivity to offer local-rate telephone access.
In this particular case Voxbone is leveraging their private IP network to pass WebRTC traffic without resorting to using the public internet. The benefits of a managed, private network include assured quality of service and security, as well as a single point of support and accountability.
Voxbone has a long history of integration using SIP. This new WebRTC service will allow it’s customers with existing SIP infrastructure to tie into the WebRTC realm. They report ten existing customers currently in private beta of WebRTC-based click-to-call for call centers, with broad roll-out projected for Q4/2014.
This week OnSIP, long known for its popular SMB hosted PBX service, launched a new initiative offering WebRTC– based platform-as-a-service. Their core business has been the hosted PBX service, which is built upon SIP standards. This new service targets web developers who want to easily incorporate WebRTC into their applications.
The companies web site has been extended to include an area described as OnSIP For Developers which details the service offering. The principle behind the service is to leverage their core SIP infrastructure to deliver the signaling solution that WebRTC alone does not provide. Thus a web developer can easily create a WebRTC based front-end that’s backed by the scalable, geographically distributed infrastructure of the OnSIP hosted PBX platform.
Last weeks commentary about how larger companies need to do better at rolling out WebRTC seems to have struck a chord in some circles. Apparently there are others who feel as I do that creating yet another free WebRTC-based video chat tool is just so much reinventing the wheel. It’s a pure marketing move aimed at establishing cool-by-association with something hip, shiny & new.
However, this critique we should reserve for developers who ought to know better. More specifically, those who are already involved in communications of some sort. Citrix’s GotoMeeting for example. If you have a track record of working with voice+video then WebRTC is novel, but it should not be entirely new.
In reality, the purpose of WebRTC is to enable an entirely new class of web developers with respect to online communication. That means word of WebRTC needs to spread. It needs to leave the confines of the crowd that knows that Opus is an audio codec, and find a home with the crowd that just needs to put some nice new feature into their web application. This is migration to face a much larger audience of developers.
For anyone who has been tracking WebRTC over time the interview is bit on the cursory side, but it does what is required. It gets people who are not already into communications excited about adding a new tool set to their web development arsenal.
This sort of training for developers that is going to be crucial to getting WebRTC used in the myriad possible applications that its creators might have imagined. It will enlarge the discussion around its use, bringing to bear the imagination of a massive new audience. This is where the new toolset that WebRTC presents will hopefully inspire innovation.