https://www.w3.org/2006/tools/wiki/api.php?action=feedcontributions&feedformat=atom&user=TimblW3C Tools Support Wiki - User contributions [en]2024-03-28T09:43:25ZUser contributionsMediaWiki 1.41.0https://www.w3.org/2006/tools/wiki/index.php?title=Zakim-SIP&diff=301Zakim-SIP2012-11-09T11:43:57Z<p>Timbl: /* What is Zakim-SIP for? */</p>
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<div>[[Category:VoIP]]<br />
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Documentation for W3C Zakim-SIP bridge system, based on [http://www.asterisk.org/ Asterisk].<br />
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See also W3C Community contributed [http://www.w3.org/2006/tools/wiki/Zakim-SIP-tips tips and tricks].<br />
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=What is Zakim-SIP for?=<br />
Zakim-SIP complements the traditional way of joining Zakim , the W3C teleconference bridge, via landline or mobile phone.<br />
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With this system, you can call ZakimBridge via VoIP Phone using [http://en.wikipedia.org/wiki/Session_Initiation_Protocol SIP protocol].<br />
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Among other things, this allows to join teleconferences from outside the US without having to pay for international calls anymore.<br />
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=How to use=<br />
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From your SIP phone you can call Zakim via [sip:zakim@voip.w3.org zakim@voip.w3.org] and then enter your conference code followed by # sign by sending Dual Tone Multi Frequency (DTMF aka touch tone).<br />
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=Echo Test=<br />
* [sip:123@voip.w3.org 123@voip.w3.org]<br />
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=SIP service providers=<br />
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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.<br />
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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.<br />
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* [http://www.ekiga.net/ Ekiga]<br />
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=SIP clients=<br />
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.<br />
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==Windows==<br />
* [http://www.counterpath.com/x-lite.html X-Lite]<br />
* [http://www.linphone.org/ Linphone] command line UI works with screen readers<br />
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== Linux ==<br />
* [http://www.ekiga.org/ Ekiga]<br />
* [http://live.gnome.org/Empathy Empathy] ''(to use SIP calls you need to install Telepathy SIP plugin)''<br />
* [http://www.qutecom.org/ QuteCom]<br />
* [http://www.linphone.org/ Linphone] (reported to work with screenreaders, on Linux version only)<br />
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==Mac==<br />
* [http://code.google.com/p/telephone/ telephone]<br />
telephone requires SIP account to use it. Easy way to use it is getting a SIP account from [http://www.ekiga.net/ Ekiga].<br />
* [http://www.counterpath.com/x-lite.html X-Lite]<br />
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==iPhone/iPad==<br />
* [http://itunes.apple.com/us/app/bria-iphone-edition-mobile/id373968636 Bria]<br />
* [http://www.domain17.net/weephone/ WeePhone]<br />
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==Android==<br />
Starting with Android 2.3.4, Android has native support for SIP.<br />
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Otherwise, [http://code.google.com/p/csipsimple/ csipsimple] is an open source application that provides SIP connectivity integrated in the main phone application.<br />
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=FAQ=<br />
* I can't hear any sound.<br />
Try to call [sip:123@voip.w3.org 123@voip.w3.org (Echo Test)]. If it doesn't work, please confirm your SIP phone and account are operating properly. If they are then please contact [mailto:sysreq@w3.org sysreq@w3.org]<br />
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* Why doesn't Zakim recognise me and announce my arrive in IRC as it does for PSTN phone lines? <br />
We may add this at a later date. You can still mute/unmute, etc in IRC, see [[Zakim_Tips|Zakim Tips]] for how to identify which line is yours.<br />
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* Why don't you support @@@ VoIP protocol?<br />
There are many popular VoIP protocols but limits to W3C Systems resources. Do [http://www.w3.org/2006/tools/wiki/Zakim-SIP-tips consult the tips page] to see if anyone has documented ways to call from your preferred platform.</div>Timbl