WebRTC Rocks for Music

Live DJ broadcasts powered by WebRTC

Please…not another video conferencing enterprise/consumer/whatever service of 2,4,6…people talking and chatting. Well, this one sure isn’t!

The exciting thing about WebRTC is that you can never predict what applications it will serve. Although many players are investing in these classic video conferencing services, others are using it for completely different applications. Such is the case with a French startup founded in 2013 by Yannick Gouez, Eymeric Pierre-Louis and Romain Agostini.

Introducing unltd.fm

unltd.fm is at the crossroads of Music Streaming Services and Web Radios: the startup offers an interface that allows DJs and Radio Hosts to produce live and interactive music broadcasts. With this platform, DJs and Radio personalities are able to create their own broadcasts directly from their web browser: Log-in, plug-in your equipment or use the in-browser mixing console and you’re ready to go.

Music Fans can discover music through live broadcast indexed by their musical content (Artist played, Music Style, Tempo, Popularity, etc.). Viewers can also interact with the broadcast using Social Networks like Twitter and Instagram and can also Comment, “Like” and Share the tracks played. 
 Untld.fm claims to offer viewers the experience of what “Smart Radios” should really be.



This web broadcasting service uses WebRTC as its core technology. “In the beginning, there were Shoutcast servers, created in 1999, and Icecast (open source). These servers had become pretty much the standards but remained complicated tools to use because you needed to know how to configure them. The arrival of HTML5 APIs have greatly simplified things,” explains Yannick Gouez. “Our service is live: the concept is that of a radio. It works in one to many broadcasting and can be scaled to support a high amount of simultaneous listeners”.

The core technology relies on a WebRTC MCU that allows to establish One-To-Many broadcasting connection from the Broadcaster to the listeners.

A Broadcasting gateway is also being implemented in order to allow ‘standard’ and plugin-less broadcasting to non-WebRTC devices (iOS devices, smartTVs, etc.).

Automatic ‘over-the-line’ track detection is provided by an Audio Fingerprinting engine integrated into the MCU.

“We have optimized our WebRTC platform to allow OPUS Stereo 48Khz broadcasting which provides a better quality than MP3”.


While Audio and Video streams are broadcasted using one-to-many connections over webRTC, the interactive features rely on the Data Channel.

Among Technical challenge faced are:

  • Achieving high quality audio broadcasting over the open internet using WebRTC
  • Feeding WebRTC with WebAudio
  • Broadcasting to non-WebRTC browser & devices (currently in development)


Untld.fm is going after the semi-professional DJ market, a rapidly growing population that needs to federate its fans. “Typically, a DJ who broadcasts one show per week and who is going to reach 100 to 150 listeners. There are many analogies with the principal of the webinar,” points out Yannick Gouez.


Untld.fm is currently in Alpha version with Beta version planned for Q4 2014.

Applications for the Beta program are open on untld.fm

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WebRTC Conference 2014

New Sponsors Are Jumping On-Board For The WebRTC 2014 Conference

Third edition of the WebRTC Conference & Expo Paris : Four new sponsors

Genband, Metaswitch, NG Media and Apidaze announce their participation to the WebRTC Conference Expo, which will take place in Paris from 16th to 18th of December 2014.

Other major WebRTC players will also soon confirm their participation. Within just two years, this conference has become the most significant one in Europe. The 2014 edition will gather together more then 40

exhibitors. Interest in WebRTC has created a rich and vibrant ecosystem of vendors. Traditional equipment vendors have also invested in this segment while the operators are forced to offer their own solutions.

WebRTC Conference 2014

The 2014 Agenda: new usages, customer profiles

The conference programme will highlight new usages of WebRTC: data channel video streaming, WebRTC & TV services, M2M applications.

Other important sessions will cover first customers profile, service providers strategies and technological issues with ORTC and WebRTC 2.0.

For more information visit our Website or Contact Us.

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Joachim Stegman Deutsche Telekom

Deutsche Telekom Taking a Stab at WebRTC: An Interview with Joachim Stegmann

Joachim Stegman Deutsche TelekomI had the pleasure of speak with Joachim Stegmann from Deutsche Telekom about his current work and view on WebRTC.

Joachim works at T-Labs, DTAG’s research & innovation lab, leading the Future Communication team. He has been working on WebRTC related topics for more than two years now.

Let’s hear what he has to say about Deutsche Telekom WebRTC activities and plans.


What are the opportunities of WebRTC for an operator?

Many operators see WebRTC as a threat because the world of telecommunications is now open to the web. In fact, it is now easier for web companies to integrate real-time voice, video, and data communication as part of their web applications. It is expected that this will increase the decline in operators’ traditional voice and messaging revenues. However, these new OTT applications create new communication islands because signaling and interoperability are not within the scope of WebRTC. On the other hand, WebRTC has the potential to create new integrated service offerings.

In principle, the opportunities for an operator can be classified into two groups:

–        Enable interoperability between communication islands: combine different WebRTC ‘bubbles’ and connect them with the Telco networks. Let the user do all his communication from a single (web-based) application.

–        New business development especially in the B2B and B2B2C segments:  Enable integration of real-time communication into business customers’ web applications. Examples are innovative customer service solutions, unified communication and collaboration services, and web-based mobile VoIP applications.


What are your current projects and with which partners do you work?

Some projects in Deutsche Telekom are performed within the technology departments. The main objective of these projects is to test the integration of WebRTC gateways into the IMS networks. Some prototype solutions have already been implemented in different countries. Within T-Labs we focus on new innovation based on WebRTC. Together with other Deutsche Telekom business units we define disruptive business opportunities. Additionally, we are currently working on a generic technology framework for web based communication. Partners are suppliers of core technology as well as new startups in this field.


Do you target mobile services?

The number of mobile devices with WebRTC enabled browsers is increasing very fast. Although WebRTC technology is not optimized for mobile communication yet, we believe that many problems can be solved in the near future. As we get better 4G coverage in the next years, some of the quality issues will most likely disappear. The advantage is that we can create a real cloud-based mobile service that integrates communication with other mobile web services and can be accessed from any device.


What is the business case for you? Do you plan to open your future platforms to other customers?

The business potential is different for the evaluated use cases. E.g. in the B2C area we expect additional revenues from higher data consumption due to an increase in video communication while cost reductions can be calculated when introducing WebRTC in customer service. In the B2B area we can offer more attractive products that integrate real-time communication in business applications. Even in the area of content delivery networks or M2M solutions, WebRTC could play an important role. In the future, we will open our platforms for selected partners with attractive product and service offerings.


What are the current main drawbacks and brakes of WebRTC?

One well-known drawback is that WebRTC is not supported by some major browser vendors. Additionally there still is a dispute about the video codec (VP8 vs. H.264) with related patent issues. However, our feeling is that these challenges can be solved in the near future. In the meantime it is necessary to create workarounds for the desired applications.

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WebRTC Voice Applications

WebRTC is commonly used for voice applications but video takes the highlights

Relatively speaking, WebRTC video communications is massively overstated in importance. That may sound like heresy, coming from a WebRTC analyst, but it’s also true. Now that doesn’t mean that video is unimportant, nor that it isn’t going to grow hugely in scope, but it’s certainly not the only game in town. And it highlights the surprising lack of voice-only use-cases for WebRTC so far.


This reflects a common fallacy in the telecoms industry that so-called “richer” or “multimedia” forms of communication are inherently better, when in fact, they’re just better-suited to certain use-cases or contexts.

Indeed, one only has to look at the huge proliferation of messaging-type applications in recent years, from SMS to web chat to Twitter to email to the various mobile IM models, to realise that often “less is more” in communications. (The obvious counterpoint is RCS/joyn, which amply illustrates that being “rich” doesn’t make you popular).

Given a broad choice of options, consumers tend to pick whatever seems to be the “right tool for the job”. Even when offered a “multimedia chainsaw”, there are still plenty of occasions when a good old-fashioned textual screwdriver or audio spanner is more appropriate. Globally, around 4-5 billion people use voice and text communications regularly. For video, it’s probably more like 100-200m – and for multi-party video, only a small fraction of that.

Too many commentators lazily refer to WebRTC as “Skype in the browser”, invoking an image of video chat or conferencing as the default mode. Few people use terms like “VoIP in the browser” or “Viber in the browser”. Yet ironically, it’s the audio codecs which are agreed, while video is still subject to debate.

Now to be fair, there are various WebRTC audio conferencing products out, while Vonage launched one of the very first mobile WebRTC apps last year. A number of internal contact-centre solutions use a browser dashboard instead of a traditional telephony platform. Twilio, Plivo and Tropo have voice-centric cloud platforms, while a couple of Telco-OTT propositions evolve the normal telephony model to WebRTC. There’s even one or two music-jamming applications around.

But these are exceptions. Most prototypes, demos and commercial WebRTC platforms are video-centric. There are dozens of lookalike video chat services, or video contact-centre concepts. There are innumerable presentations and white papers extolling a new age of video interviews, video telemedicine, video dating and connected video-capable “things”.

Yet almost no thought, design or marketing goes into new ways to extend human speech – or other forms of audio – view WebRTC. It all eyes, but no ears. It’s as if 120 years of “phone calls” has blinded (deafened?) us to the viability of other formats for voice.

Now, it could just be that video is just “shiny” and demo-friendly in a way that audio generally isn’t. It also attracts vendors selling bigger and costlier network boxes too, as mixing and transcoding aren’t as commoditised or easily-addressed by open-source. It could also be down to psychological or design-related reasons – talking to a browser seems a bit weird for some reason, compared to talking to a standalone app.

But the fact is that the bulk of today’s realtime communications is voice-centric, often for good practical reasons. A lot of people cannot or will not use video for many cases – it may be dangerous (eg while driving/walking), distracting, invasive or uncomfortable. In a multi-tasking world, looking at a camera often involves too much cognitive load (especially as you watch your own image), and may inhibit concurrent tasks such as note-taking, or reading presentation slides.

WebRTC-powered video will absolutely have many uses cases, but it equally can never be ubiquitous or the default mode for all instances of communications.

So it seems strange that so few WebRTC applications and services have been targeted at audio-only, or even audio-primary usage. There seems to be a significant gap for companies (or open-source) solutions to enable more pure-audio WebRTC than is currently seen. In particular, the assumption that anything based around speech is necessarily a “call” and could/should be interoperable with the phone system is wrong.

Yet even within the traditional telecoms industry, we’ve long had other formats for voice communications – walkie-talkies, private radio, push-to-talk, voice messaging, hoot-n-holler and so forth. Add in cloud capabilities like speech recognition, storage, translation, audio-processing of various types and we should have a wide range of WebRTC possibilities. Where’s the “Voice Instagram” that allows people to converse in Glaswegian accents or Donald Duck squawks? Where are the realtime profanity bleep-outs, or inline stress-analysis lie detectors?

And going beyond the actual transmission of spoken words, there’s another world of intent and purpose. Why exactly are people talking, and what are they actually hoping to achieve? How can the web – and the network – enhance that? The contextual capabilities of browsers and devices should be able to add to the experience of audio communications – recognising when to capture and emphasise the sounds of crashing waves on beach during a call. Or when to block out the sound of a crashing bore in the background at a party.

WebRTC video offers huge opportunities. But at the same time, we should remember that voice communications has delivered trillions of dollars in revenue in the past, and could continue to do so. Let’s ensure that the Future of Voice is as vibrantly-coloured as the Future of Video.

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