SIP and the Browser: RTCWEB and HTML5

There’s a lot of discussion these days about an effort known as RTCWEB – Real-Time Communications in Web browsers.  It is part of the HTML5 effort to build base voice and video communication capabilities directly into web browsers.  What does this mean?  HTML allows a web site or developer to easily display an image or stream a video, simply by including a standard HTML tag in their web code.  The RTCWEB extensions will similarly allow Skype-like voice and video communication, simply by adding a few HTML5 tags and some Javascript or Java code.  There are websites offering this today, but you first have to download a browser plugin before you can use it.  The developer has to write plugins for each platform and browser they want to support.  As a result, few offer this today – GoogleTalk and Google+ Hangouts are an exception to this.  For this effort to be successful, there must be standards, and two Internet standards bodies are working together closely:  the IETF (Internet Engineering Task Force) and the W3C (World Wide Web Consortium).  I have been active in the IETF’s RTCWEB Working Group, and colleagues of mine have also been involved in the W3C WEBRTC Working Group.

So how does this fit with SIP, which I’ve spent much of the last 10+ years working on?  SIP or Session Initiation Protocol is the IETF protocol for establishing voice and video sessions over the Internet.  SIP is used all over the Internet today, and in private networks.  It is used by service providers for VoIP (Voice over IP) networks, and it is used by enterprises for their internal PBX (Private Branch Exchange) networks.  It is also in a number of applications and services including Skype In and Out and even Apple’s Facetime (kind of).

Does this mean SIP in the browser?  This is an open question today being debated.  Although I have written drafts on the topic, I am no longer so sure this the right approach.  The alternative approach, that says that we don’t need to standardize the protocol between the browser and the web server – just use some downloaded Javascript or Java.  But this doesn’t mean SIP will go away – rather, SIP will continue to be used to connect networks and elements, and this will include new RTCWEB websites that communicate with each other and service providers.

This topic will continue to be discussed in the standards bodies, and also in next month’s ITEXP Internet Telephony Expo.  I’m excited to be giving an all-day SIP Tutorial with Henry Sinnreich in which we will introduce and teach SIP and also the principles behind the RTCWEB effort and how SIP and RTCWEB relate.  You can find out more about the tutorial and register using this link.

One of the other hot topics of RTCWEB is security, and I have written and spoken out about the need for privacy – protection against eavesdropping on voice and video communication.  A media security protocol such as ZRTP would be an excellent choice, but there are other options.  Unfortunately, there is a contingent that wants to permit unencrypted voice and video media from the browser.  But that is a topic for another day…

Hope to see some of you in Austin, Texas at the SIP Tutorial on September 15. 2011!

  1. #1 by bla on August 28, 2011 - 8:44 pm

    Well it started well then but the article never got into any depth more a page of low level intro’s. Would have been nice to know where HTML5 is at since you’re suipposedly on a working group.

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