WebRTC is going to reach that milestone thanks in part to Firefox 22, which was just released this week. The new version of Firefox comes with WebRTC enabled in its beta version, which should add a large number of users to the addressable market for WebRTC developers.

Uberti also said that WebRTC is going to come to iOS devices soon: Apple hasn’t joined the efforts to implement and standardize WebRTC yet, but Google wants to nonetheless give developers a way to address users on iPads and iPhones through the release of a native toolkit.

Of course, there is another holdout: Microsoft has been pushing forward with its own version of WebRTC, which some have seen as an effort to torpedo the standard. Uberti had a much more optimistic take on the differences Friday, saying that Microsoft has been “a great participant in the standards.” He added: “I’m very optimistic that we are going to see a version of (Internet Explorer) that supports this technology in the not-too distant future.”

Image courtesy of Flickr user  Tsahi Levent-Levi.

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Skype is reportedly testing video messaging options for mobile users today. The service would let people leave a video-based message for their friends. I can see a whole new variant on the “wish-you-were-here” picture messages I send that could involve panoramic views or the local soundscape (good for concerts and birdcalls, bad for when I’m in New York City).

A few weeks ago, Twitter launched Vine, to let people record 6-second videos and post them easily from their mobile, and Facebook is testing a voice messaging application that will let you leave a voice-based message for friends from Facebook. While the Facebook example isn’t video-based it drives home the larger point: Our web interactions are pushing forward to mirror our real-world interactions as much as possible, which means that our bandwidth demands and our mobile devices need to keep up.

On the mobile device side, we’re doing fine. Processing, cameras and microphones on smartphones are enabling us to record quality videos, voice and images. In the case of images we even have enough processing power for some editing. But on the bandwidth side, it’s unclear if we’re going to have the capability to share our efforts. That’s why on the wireless and wireline side we need to keep adding capacity and lowering costs. Conducting a video call today sucks up a lot of bandwidth, but there are ways to reduce the impact on the network and drive down costs for consumers and the operators.

When I look at the increasingly visual nature of the web and the influx of video options for communication I realize that we can finally escape the limits that technology has imposed on how we communicate over long distances. Letter writing, postcards, voice calls and even static web pages are poor substitutes when you want to share an experience with someone, and they are substitutes that are driven by the limits of the technology at the time. Many of those limits are no longer there.

Adapting to this will require us to ditch centuries of habits and preferences, but it opens up much higher quality ways for people to communicate. We will still drag these other forms of communication into our video-based future but we’ll be able to choose when an email makes the most sense or when we’d rather stick with voice.

As I scroll down the pages of an online catalog, I am grateful that I have the bandwidth at home to load pictures quickly so I can see the details in the product. I can’t wait for the ability to see things in 3D — or even set up a quick video call with someone who is near the product for a closer look.

I assume my six-year-old daughter — who refuses to take phone calls from people she loves unless there’s a video component — will resort to voice only for strangers and business-related conversations. Getting to that point means more work needs to be done to seamlessly integrate the options available to people much like Apple has done with FaceTime on its platform, and then to spread that to all platforms.

Companies like Skype, BlueJeans Networks, Polycom, and countless others are all trying to make this real as are the people pushing for the WebRTC standards. Right now it’s a mish-mash of standards, platforms and options, but video will coalesce into something that as simple as picking up a phone or mailing a letter is today.

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Microsoft has been working for some time on web-based real-time communication, and could one day use this kind of technology to take Skype to the browser, as well as make it interoperable with other messenger platforms and applications. The company started to participate in efforts to standardize this kind of browser-based communication last summer, albeit with a somewhat different take than others.

Previous efforts around web-based, plugin-free voice and video chat were largely driven by Mozilla and Google, with the latter contributing a lot of its technology to an effort dubbed WebRTC, which is short for web-based real-time communications. Work on WebRTC had been progressing in 2012, and parts of the technology has already been implemented in Chrome and Opera.

However, Microsoft argued that some of the core assumptions of that approach were wrong. In particular, the company took issue with efforts to make Google’s VP8 video codec the default choice for WebRTC. Microsoft’s own proposal, dubbed CU-WebRTC, would instead leave it up to the developer of each app to settle on a codec as well as on other specifics, like the data formats used to communicate.

The company reiterated this position in a blog post Thursday, which was co-authored by Skype Senior Architect Martin Thomson, Lync Principal Architect Bernard Aboba and Microsoft Open Technologies Principal Program Manager Adalberto Foresti. In it, the trio writes:

“A successful standard cannot be tied to individual codecs, data formats or scenarios. They may soon be supplanted by newer versions that would make such a tightly coupled standard obsolete just as quickly. The right approach is instead to support multiple media formats and to bring the bulk of the logic to the application layer, enabling developers to innovate.”

The post also admits that Microsoft’s initial proposal wasn’t exactly embraced by everyone: “The proposal generated both positive interest and healthy skeptical concern from working group members,” it states. However, the trio believes that the industry is nonetheless making progress towards a common standard – it may just take a bit longer to get there.

Image courtesy of Flickr user  Tsahi Levent-Levi.

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“It’s the biggest milestone yet… web developers can now offer Chrome users the ability to have live, high quality audio and video communication as part of their web experience.”

Google has been a big champion of WebRTC, open sourcing key components of the technology and more recently adding it to the beta version of its Chrome web browser. By adding WebRTC to the stable version of Chrome, which will be downloaded by millions of consumers, Google signaled that the technology is getting ready for prime time.

WebRTC also got another boost this week when Mozilla announced that it started to include the framework in the nightly and Aurora (pre-beta) builds of its Firefox web browser. And the technology got some real-world validation when the freshly-acquired video chat platform provider Tokbox released OpenTok on WebRTC, enabling develpers to build WebRTC-based video chat applications that connect users on supported browsers with consumers using iOS devices. And web video chat provider Bistri added to the momentum by rolling out WebRTC-based video calling.

However, WebRTC still has some challenges to overcome before it becomes a universally adopted real-time standard. Among the challenges is the selection of codecs. Google and Firefox would love to see WebM become the default codec of browser-based video communication, but Microsoft is favoring a different approach that would leave it up to the individual developer to choose a codec, and Apple has been completely absent from the table.

Image courtesy of Flickr user  Tsahi Levent-Levi.

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Opus has been developed with real-time communication in mind, and is likely going to play a major role in WebRTC, the emerging standard for plugin-free voice and video communication within the browser. However, it turns out that Opus actually also performs really well for music, with Xiph.org’s Monty Montgommery telling me recently: “Opus has lower overhead, no big headers, scales better… it’s just a better codec.”

That’s why it could make sense for Spotify and others to use Opus for their service. Spotify currently uses the free audio codec Ogg Vorbis for most of its streaming. Eliot Van Buskirk of Evolver.fm reported yesterday that the company’s CEO Daniel Ek had told him that it would be relatively easy for Spotify to switch codecs, leading him to speculate that “Spotify could improve” with a switch to Opus.

VLC support should help Opus to get better known among early adopters. Users willing to try the codec can find some examples, including two audio streams that can be played back with the new VLC version, on the Opus website.

Image courtesy of Flickr user ghewgill.

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Google software engineer Justin Uberti shared the news on the company’s Chromium blog:

“Chrome now includes the PeerConnection API, which allows developers to create web apps with real-time audio and video calling without the need for a plug-in. Together, PeerConnection and thegetUserMedia API represent the next steps in WebRTC, a new standard which aims to allow high quality video, audio, and data communications on the web.”

Uberti illustrated the capabilities of the new API with a demo video chat application, which can be accessed after installing the latest Chrome beta release.

WebRTC is a big deal for Google: The company aims to replace the proprietary plugin it has been using to facilitate Google+ Hangout video chats with a standards-based solution that would work in a variety of browsers without the download of any additional plugins. WebRTC could also eventually lead to more interoperability across a number of video chat and messaging services, and it could be an even bigger disruption to mobile video communication.

However, it is likely going to take some time before WebRTC becomes a widely adopted standard. Microsoft publicly joined the WebRTC camp in August – but also slowed things down a bit by proposing a different approach than then one now implemented by Google.

Image courtesy of Flickr user  Tsahi Levent-Levi.

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There’s likely no one who could explain this better than Cisco’s Cullen Jennings, who co-chairs the Internet Engineering Task Force’s WebRTC working group. Jennings recently posted an introductory presentation about WebRTC on Vimeo, and you should watch it if you want to know what people are talking about when they’re saying WebRTC:

RTCWeb Explained from Cullen Jennings on Vimeo.

Two things are worth noting: First of all, Jennings actually does a really good job at breaking down the complex technology behind these efforts. And secondly, he’s trying really hard to steer clear of some of the politics in this space, most notably around codecs.

Google in particular is putting its weight behind VP8, the video codec it open sourced in 2010. Microsoft and others have raised doubts on whether VP8 is really royalty-free, or whether third parties may have patent claims on the codec. That may explain why Jennings seems to be a little unsure on how to talk about VP8, but it could also help to provide some common understanding of the technology without poking into some of its open wounds.

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And yet, we are glad to use Verizon’s tagline, “Can you hear me now?” all the time. In fact, we think of Verizon as the paragon of phone call quality in this country. And the reason we do that is because we have very low or little expectations from the voice experience. It is virtually impossible do have a decent conversation these days on any device. Calling mom via AT&T long-distance? The good news is that I don’t have to hear what she has to say about my marital status.

Calling my sister on the phone in Midwest on what is a Verizon-to-Sprint call is like playing a guessing game. Dropped calls on an AT&T network are routine. The blame goes squarely on carriers not investing enough in their voice infrastructure; instead, they invested all their dollars on data, which is essentially a big giant ATM machine for the mobile operators. The IP-ization of voice only added to the declining voice quality.

Thanks to constant complaints from consumers, availability of more bandwidth and, of course, new technologies, we can hope that those days bad calling experience are behind us. We have the emergence of what we’ll call HD, or hi-def voice, to thank.

What’s the reason for my optimism? Some recent developments:

• Skype recently introduced Opus, a new codec that it will use as part of its service and is also making it available for others to adopt. It is now an IETF standard. Skype argues that Opus is better quality than other existing codecs. They offer test results, but frankly like anything, we better believe our own ears than take word of a company that considers clutter to be a great user interface.
• Google is pushing its own hi-def codecs as part of Google WebRTC. Could it eventually find its way into Android? Maybe.
• But the biggest development came this week when Apple announced what it calls “wideband audio,” which is its attempt to create a better call experience. Apple announced that it was initially working with 20 carriers across the globe and I am guessing more and more will join in the future.

 In 2009, columnist Dan Berninger wrote:

iPhone and the Apple App Store may have set a new standard for design and availability of mobile apps, but the iPhone operates within the same decades-old voice quality constraints as other handsets. Indeed, the rapid pace of handset innovation does not change the fact that AT&T, BT, Telefonica et al cannot improve voice quality.

Apple has been good at pushing new technologies at scale. And thanks to the fact that iPhone is still a must sell-device for the carriers, we can expect wider adoption of wideband audio. We have been crying for HD voice for a long time. As Dan further observed, “people will embrace the HDVN as they experience the benefits of improved voice quality in contexts that involve emotional content.”

We agree. Earlier this year, my colleague Kevin Tofel played around with Bria for Android and was amazed by the wideband audio calls he was able to make. He shared his impressions with readers and like him, I cannot wait for improved quality of voice. Perhaps, maybe once again, voice will sound like voice and be as reliable as that BMW.

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Opus was originally developed with narrowband, real time applications in mind, including audio and video chat. It works with bitrates as little as 6 kbps, which makes it a great choice for VoIP and video chat applications on mobile data networks. However, the codec also supports bitrates up to 510 kbps in full 48 kHz stereo. “It is The One Codec,” Xiph.org founder Monty Montgomery told me via IM Tuesday morning.

Montgomery’s Xiph Foundation previously spearheaded the development of Ogg Vorbis, an open and royalty-free audio codec that was at one point supposed to replace the popular MP3 format. MP3 was developed by the German Fraunhofer Institute, which has been licensing the codec for commercial use ever since. Anyone using MP3 for a commercial music service has to fork over 2 percent of revenues generated with the format, and the use of the codec in applications such as iTunes or even video games costs money as well.

Ogg Vorbis never really caught on with end users, but it is powering some big applications under the hood. Spotify, for example, uses the MP3 competitor to stream music to its desktop application, and the codec is also used by popular video games like World of Warcraft. Some of these companies might be inclined to switch to Opus, Montgomery said. “Opus has lower overhead, no big headers, scales better… it’s just a better codec,” he told me.

Still, it’s likely that one the major use cases for Opus will be real-time communication. The codec has low latency, and can adapt to changing bandwidth conditions without any need for rebuffering. Opus is going to play a key part in WebRTC, the new real-time messaging framework backed by Google and others that is meant to enable voice and video chat in the browser without the need for any additional plugins. Firefox is already supporting native playback of Opus, and Google has expressed interest in shipping it as part of Chrome as well.

Image courtesy of Flickr user ghewgill.

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Tokbox has been offering developers a way to integrate video chat into their own services with a few lines of code through its Opentok platform, which has been used by more than 50,000 developers since its launch some 18 months ago. Small told me Monday during a phone call that Opentok’s ideal users are e-commerce sites and similar services that need to make it as easy as possible for users to use video chat. “If you want to do a Skype call, people will download Skype,” he said. But no one would download an app or a plug-in simply to talk to a customer service representative.

Tokbox believes that WebRTC won’t just be easier for end users who won’t have to deal with any plug-ins, but that it will also offer faster connections an better video quality.

In that context, WebRTC could be of great help. “WebRTC has a lot of promise and we are very excited about it,” Small said. Up until now, Tokbox has mostly been using Flash on the desktop, and the company also just rolled out a new SDK for iOS  devices.

With its new WebRTC release, the platform will automatically detect which devices and browser platforms users are running — and if possible — facilitate calls via WebRTC. The whole process will be invisible to end users, Small told me, and won’t require any big changes from developers. “The main thing we are focusing on is removing complexity from the process,” he said.

Of course, that’s easier said than done for a technology that has yet to become a standard, with different companies having different views on how things should be done. One of the key differences between Microsoft and Google is the video codec being used for WebRTC. Google wants to default to its own VP8 codec, which the company open sourced in 2010. Microsoft has been critical of VP8, and Matthew Kaufman, principal architect for Microsoft-Skype on WebRTC, told me that it would prefer to give developers choice on which codec to use for their WebRTC apps.

Small said that he hopes that this roadblock will be sorted out soon, adding: “In the meantime, it’s just going to increase fragmentation.” Still, Tokbox felt it was the right time to give developers access to WebRTC for their video apps – even if it might take some time until everyone can agree on a common standard. “There is a lot of promise, but there is a rocky road to get to that promise,” he said.

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