teleconference change to WebEx starting next week: 13 May

As noted on today's teleconference we plan to transition from using Zakim for voice to WebEx, starting with next week's teleconference (13 May 2015).

Please allow extra time to make the voice connection.

You can dial in directly: +1-617-324-0000, Access code:645 413 954

You can also use the web address to have a Skype like conference call, Beware: the system will try to install a program on your machine  and may have issue with certain browsers (e.g. Firefox)

We will continue to use zakim to manage the queue (q+, q-, q? etc) but it is not integrated with webex

Ivan put up a wiki page with more details, see

regards, Frederick

Frederick Hirsch
Co-Chair, W3C Web Annotation WG

> Begin forwarded message:
> From: Ralph Swick <>
> Subject: Upcoming change in teleconference services: WebEx now available to W3C groups, Zakim end of life
> Date: April 30, 2015 at 10:35:16 PM EDT
> To: Chairs <>
> Resent-From:
> Chairs and other interested parties,
> Over the next several weeks your Team Contacts will be helping you to
> transition teleconferences from W3C's Zakim teleconference bridge to
> MIT's badged Cisco WebEx service.
> Though many features we enjoy in the integration of the Zakim bridge and
> irc will end with this transition, two new capabilities will become
> available to you:
> 1. International dial-out to any phone number of your choosing
> 2. VoIP connectivity via Cisco's WebEx client(s)
> A wiki page [1] has been started with some recommended best practices
> for using WebEx in W3C meetings.  I know that some of you have
> experience with WebEx; I invite you to improve this best practices
> document.  The document may also serve as base guidance for the use of
> other teleconference systems in the W3C environment.
> [1]
> W3C's Zakim teleconference system has served us long and well.  Times
> move on, however, and the hardware comprising the Zakim system lacks the
> capability to provide connectivity that is of growing importance to some
> of our community.  Three years ago we began a search for a
> current-generation system that would provide both PSTN and direct SIP
> access from a vendor who would also commit to WebRTC support.  We are
> still looking for such a system that fits our limited budget.
> Meanwhile, the MIT campus telephone infrastructure has also undergone a
> major overhaul and the financial arrangement we have enjoyed with MIT
> that provided the physical telephone circuits to our Zakim bridge is
> coming to an end this June.  This is forcing us to make a change sooner
> than we otherwise would have.
> We will continue to monitor the progress of WebRTC deployment in the
> hope that we will have available a system that meets our needs in the
> not too distant future, including PSTN interconnection and APIs for
> integration into our W3C infrastructure.
> Regards,
> -Ralph

Received on Wednesday, 6 May 2015 20:57:10 UTC