From W3C Tools Support Wiki
Revision as of 13:00, 18 August 2011 by Lwatson (Talk | contribs)

Jump to: navigation, search

Documentation for W3C Zakim-SIP bridge system, based on Asterisk.

See also W3C Community contributed tips and tricks.

What is Zakim-SIP for?

Zakim-SIP completes the traditional way of joining Zakim , the W3C teleconference bridge, via landline or mobile phone.

With this system, you can call ZakimBridge via VoIP Phone using SIP protocol.

Among other things, this allows to join teleconferences from outside the US without having to pay for international calls anymore.

How to use

From your SIP phone you can call Zakim via and then enter your conference code followed by # sign by sending Dual Tone Multi Frequency (DTMF aka touch tone).

Echo Test

SIP service providers

You don't need a SIP account to call W3C's Zakim-SIP bridge, however some SIP software clients may require you have valid account information.

Also if you want to receive incoming calls you will want a SIP account. Here are some providers we confirm work with Zakim-SIP bridge. There are many free providers, feel free to recommend more.

SIP clients

The most adapted SIP client will probably depend on your SIP provider. The list below only documents clients that have been known to work with Zakim-SIP, but many more clients are likely to work as well.



  • Ekiga
  • Empathy (to use SIP calls you need to install Telepathy SIP plugin)
  • QuteCom
  • Linphone (reported to work with screenreaders, on Linux version only)


telephone requires SIP account to use it. Easy way to use it is getting a SIP account from Ekiga.



Starting with Android 2.3.4, Android has native support for SIP.

Otherwise, csipsimple is an open source application that provides SIP connectivity integrated in the main phone application.


  • I can't hear any sound.

Try to call (Echo Test). If it doesn't work, please confirm your SIP phone and account are operating properly. If they are then please contact

  • Why doesn't Zakim recognise me and announce my arrive in IRC as it does for PSTN phone lines?

We may add this at a later date. You can still mute/unmute, etc in IRC, see Zakim Tips for how to identify which line is yours.

  • Why don't you support @@@ VoIP protocol?

There are many popular VoIP protocols but limits to W3C Systems resources. Do consult the tips page to see if anyone has documented ways to call from your preferred platform.