Standards

If you are new to WebRTC then you have missed out on years of drama in the standards bodies over various issues like SDP and codecs. These standards dictate what vendors must implement so they ultimately dictate the industry roadmap.  To get a deep perspective and appreciation of the issues, we like to ask Dan Burnett, W3C editor to comment on where we are at with the standardization process. I caught up with Dan at this year’s IIT Real Time Communications Conference and had the more detailed Q&A with him shortly thereafter.

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This is the next decode and analysis in Philipp Hancke’s Blackbox Exploration series conducted by &yet in collaboration with Google. Please see our previous posts covering WhatsApp, Facebook Messenger and FaceTime for more details on these services and this series. {“editor”: “chad hart“}

Wire is an attempt to reimagine communications for the mobile age. It is a messaging app available for Android, iOS, Mac, and now web that supports audio calls, group messaging and picture sharing. One of it’s often quoted features is the elegant design. As usual, this report will focus on the low level VoIP aspects, and leave the design aspects up for the users to judge.

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Sorry. We really wanted to do a post-cap of the W3C WebRTC and IETF RTCweb meetings that took place at the end of October and November, but we did not get to it. Victor and Reid provided some commentary on the codec debate prior to the IETF discussion. The outcome of that discussion was widely publicized and we did not have a lot of value to add to this for the developer community.

Importantly, codecs were not the only thing discussed in this latest rounds of standards meetings. There were a couple items like the move to JavaScript promises, output device enumeration, and discussions of security implications that are very relevant to the average WebRTC developer that have gone under the general media radar. To get the whole on standards right from the horse’s mouth, I asked W3C WebRTC editor and founding author Dan Burnett for an update on the recent WebRTC standards meetings and for some details on some of the more significant issues like promises and screen sharing.

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MS Elephant

What about IE?

As WebRTC has matured to a state where it’s first implementations are ready for companies to launch real services around it, the readiness of various companies to adopt WebRTC has fanned out quite a bit. Some are already charging ahead as early adopters, while others are playing it conservative. Of those in the conservative camp, one of the common doubts that gives them pause is: “What about IE?”

When speaking to those interested in WebRTC, but concerned about Internet Explorer (IE), many times we’ve tried to assure them not to worry: our friends in Redmond won’t be too far behind. We often point to the undeniably significant contributions from Microsoft to WebRTC, especially considering that they bring to the table two titans of VoIP industry (Lync and Skype). We highlight some of their early IE WebRTC demos (using beta code) as signs of progress. We’ve rationalized the absence of  a Microsoft equivalent to what Chrome and Firefox are shipping, by noting the slower release cycle for IE. However, we’ve come to realize that to some, IE support is a really big deal.

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I’m at the IIT RTC Conference this week in Chicago which is an excellent, no-BS conference that featured many WebRTC luminaries and one of best events I have attended in a long time.

On Tuesday I moderated a panel with WebRTC contributors and ORTC promoters, Robin Raymond of Hookflash, Bernard Aboba of Microsoft, and Peter Thatcher of Google, asking many of the same questions I did on the ORTC Q&A several weeks ago.

Dan Burnett was in the room, asking a lot of questions. If you don’t know Dan, he is a long time W3C author and editor. He is also one of the Godfathers of WebRTC who was there right at the beginning. He also has a highly regarded book on WebRTC coauthored with Alan Johnston.

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Biggie vs. Tupac. Gates vs. Jobs. Apple vs. Samsung.  Nothing catches people’s attention for no legitimate reason like a feud. Unfortunately this isn’t just a celebrity phenomenon. Feuds have been endemic even to real communications as well. From the very beginning, Elisha Gray’s dispute with Alexander Graham Bell over the original telephone patent showed the industry has a propensity for squabbles. Unfortunately we have become so accustomed to feuds that we sometimes fabricate battles that do not really exist. I fear that this is often the case with one of the most important, but misunderstood efforts affecting WebRTC’s future – Object Real Time Communications (ORTC).

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One of the first posts we published on this blog a year ago was a ‘A Hitchhiker’s Guide to WebRTC standardization‘. Since then, the work has certainly progressed and we have been sharing here a number of updates on the topic. This week we’re having qn IETF meeting in Canada and when it comes to WebRTC some of the topics in the agenda include ALPN (Application Layer Protocol Negotiation)STUN Consent Freshness and audio (interop with legacy) and video (H.264 vs VP8 as mandatory codec is NOT discussed this week but you can see VP9 and H.265 are already mentioned in the slides) requirements. Several Working Groups other than RTCWeb will also discuss this week topics that are relevant to the WebRTC effort (e.g. MMUSIC WG). During the STRAW WG session, the working group I co-chair at the IETF, we’ll discuss some features of interest for those implementing WebRTC in servers like MCUs, Application Servers, WebRTC-SIP gateways or WebRTC-enabled SBCs: DTLS-SRTP, STUN and RTCP traversal/termination are examples.

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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 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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