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 complements 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 email@example.com and then enter your conference code followed by # sign by sending Dual Tone Multi Frequency (DTMF aka touch tone).
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.
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.
- FreeSwitch Client "accessible by design" and appreciate feedback
- Linphone command line UI works with screen readers
- Empathy (to use SIP calls you need to install Telepathy SIP plugin)
- Linphone (reported to work with screenreaders, on Linux version only). There is also a command line interface
- FreeSwitch Client "accessible by design" and appreciate feedback. There is also a command line interface
- Telephone requires a SIP account to use it. Easy way to use it is getting a SIP account from Ekiga.
- The following settings were used successfully with Telephone and an Ekiga account:
- Account Domain: ekiga.net
- Network: Local
- SIP Port: Blank
- STUN Server: stun.ekiga.net
- Use ICE: True
- If you can't connect, wait a moment and try again up to 5 times.
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.
Please confirm your SIP phone and account are operating properly and try to call firstname.lastname@example.org (Echo Test)
If it still doesn't work, it could be related to NAT. Check if you have an option "NAT/Firewall (use STUN to resolve)" with your SIP client. If so, enable it and use one of the STUN server in the list at https://gist.github.com/zziuni/3741933.
If the problem persists, then please contact email@example.com
- 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.