webrtcH4cKS: ~ Popular Posts

Ring! Sometimes you need an alert to get your attention. Traditional phone systems make this easy – if someone calls you your phone rings. The traditional telephony model assumes the called device is always on an available to ring and this is how it generally works across analog phones, mobiles, VoIP phones, and even desktop calling replacements like Skype. The challenge in the web model is that you can no longer assume the remote device is available to run your program’s ring command. Even if the called party has a browser open, it does not mean they have a tab running your app.  This means you need to find some other means of telling the called party to go to your URL. That can be limiting for a lot of apps. Fortunately there are solutions for this.

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The first post we published on webrtcHacks was ‘A Hitchhiker’s Guide to WebRTC standardization’ (July 2013) where we gave some initial insight on activities in the 3GPP around WebRTC and  IMS. Since then the situation has certainly evolved (well, probably not as fast as some would have expected). Since we regularly receive emails asking about the status/progress on WebRTC standardization within the 3GPP, we spent some time with our friend Antón Román, CTO at Quobis and author of the popular post ‘Anatomy of a WebRTC SDP’ to summarize the current status of the ‘WebRTC access to IMS’ effort.

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As I mentioned in my ‘WebRTC meets telecom’ article a couple of weeks ago, at Quobis we’re currently involved in 30+ WebRTC field trials/POCs which involve in one way or another a telco network. In most cases service providers are trying to provide WebRTC-based access to their existing/legacy infrastructure and services (fortunately, in some cases it’s not limited to do only that). To achieve all this, one of the pieces they need to deploy is a WebRTC Gateway. But, what is a WebRTC Gateway anyway? A year ago I had the chance to provide a first answer during the Kamailio World Conference 2013 (see my presentation WebRTC and VoIP: bridging the gap) but, since Lorenzo Miniero has recently released an open source, modular and general purpose WebRTC gateway called Janus, I thought it would be great to get him to share his experience here.

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Want to try out a newly released WebRTC feature or capability? Odds are Muaz Khan has already done it. I cannot think of any other individual who has contributed more open source WebRTC application experiments to the community than Muaz and his webrtc-experiment.com. His GitHub repository boasts 44 different projects.

He did all that in less than 2 years and he is just getting started. What is even more amazing is Muaz had done all this with very limited resources from a remote village. He doesn’t event have electricity for large portions of the day in some months!

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Many in the industry, including myself, reference Amazon’s Kindle Fire HDX Mayday button as using WebRTC or at least as something that is WebRTC like. The Kindle Fire HDX is not available everywhere, so if you have not seen this the Android Authority has a good video of this feature here.

First lets think about how we tell if an app is using WebRTC. If the app is a webpage it is fairly simple – just look for the use of the getUserMedia and CreatePeerConnection APIs in the site’s Javascript using your browser’s developer console. It is a little more complex if WebRTC is embedded inside a native application. We could start with a debate about “What makes an app a WebRTC app”? If it uses part of WebRTC source code and not the W3C API’s does it count? Since this blog is for developers, not philosophers, let’s start by figuring out what Mayday actually does by looking at a Wireshark trace.

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Next week Mobile World Congress (MWC) 2014 will take place in Barcelona, Spain. Since Barcelona is my hometown, it’s always a great opportunity to meet with industry friends and enjoy some local spots together.

Many webrtcHackers will be in Barcelona for the event, so we are organizing a meetup next Tuesday at 6PM CET. This event will be largely a social meet & greet but we will have some structured conversation to discuss the latest in WebRTC and developer needs. We will update with exact location details within the Fira Gran Via prior to the event but we will likely meet in one of the food court areas between center the exhibit halls.

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WebRTC and its peer-to-peer capabilities are great for one-to-one communications. However, when I discuss with customers use cases and services that go beyond one-to-one, namely one-to-many or many-to-many, the question arises: “OK, but what architecture shall I use for this?”. Some service providers want to reuse the multicast support they have in their networks (we are having fun doing some experiments with this), some are exploring simulcast-based solutions, others are considering centralised solutions like MCUs/mixers, and a bunch of them are simply willing to place the burden on the endpoint by using some variation of a mesh-based topology.   The folks at TokBox (a Telefónica Digital company) have great experience with multiparty conferencing solutions.  I thought it would be great to have my friend Gustavo Garcia Bernardo (Cloud Architect at TokBox) to share here his take on the topic.

At TokBox, Gustavo is responsible for architecture, design, and development of cloud components. This includes Mantis, the cloud-scaling infrastructure for the OpenTok, which uses the WebRTC platform. Before joining TokBox, Gustavo spent more than 10 years building VoIP products at Telefónica and driving early adoption of WebRTC in telco products. In fact, I’ve known Gustavo for 8 years now and the first time I met him it was preparing a proposal for a European Commission-funded research project on P2PSIP. Since then we’ve been collaborating in the IETF doing some work in the context of P2PSIP, ALTO and SIP related activities. A couple of years ago, while I was working with Acme Packet (now Oracle), we worked together designing and launching Telefonica’s Digital TuMe and TuGo.  Lately we have both shifted our focus towards WebRTC.

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As detailed in previous posts on webrtcHacks, the Internet Engineering Task Force (IETF) has worked for the past few years to standardize the “on-the-wire” protocols that make up the WebRTC engine. It is coming up on 3 months since IETF 88 in Vancouver, where the IETF was to have settled the matter of a mandatory-to-implement (MTI) video codec. The process resulted in no consensus, and the task of finalizing WebRTC 1.0 drags on with MTI video codec(s) in question.  A recent straw poll among IETF participants shows how divided the issue remains.

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As WebRTC implementations and field trials evolve, field experience is telling us there are still a number of open issues to make this technology deployable in the real world and the fact that we would probably do some things differently if we started all over again. As an example, see the recent W3C discussion What is missing for building (WebRTC) real services or Quobis‘ CTO post on WebRTC use of SDP.

Tim Panton, contextual communications consultant at Westhawk Ltd,  has gone through some of these issues. During the last couple of years we had the chance to run some workshops together and have some good discussions in the IETF and W3C context. Tim’s expertise is very valuable and I thought it would be a good idea to have him here to share some of his experiences with our readers. It ended up as a rant.

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As described in previous posts, WebRTC does not specify a particular signalling model other than the generic need to exchange Session Description Protocol (SDP) media descriptions in the offer/answer fashion.

During the last few months, my friend  Antón Román (CTO of Quobis) and I spent a lot of time with our team figuring out how to manipulate and adapt the SDP’s generated by web browsers to make them compatible with the different server/gateway technologies we’re working with.

As WebRTC makes use of new mechanisms but also existing ones that have seen few  deployment in real networks to date. SDP’s generated by Web Browsers are more complex and contain a number of new attributes that are unfamiliar in SIP or IMS networks. In the following post, Antón analyses the anatomy of a WebRTC SDP, giving a detailed description of what all those lines do.

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