This page is for contributing suggestions and interest VoIP hacks for W3C's Zakim-SIP interface.
Feel free to add to this page.
Dealing with NAT (Network Address Translation)
NAT, common used in home and some office networks, can cause problems with SIP calls depending on the NAT implementation and SIP software trying to traverse it. Symptoms may include poor audio quality.
Entering your conference code
The default setting for sending touchtones (DTMF) in the Ekiga client was wrong, in my case: if you find your code is not getting through (you always get the "This passcode is not valid" reply), try changing it to RFC2833 at Edit/Preferences/SIP
A couple people have reported better luck entering DTMF tones by using the following settings:
- In Configuration/Preferences:
- Advanced pane: DTMF should have "Use in-band" selected
- Codecs pane: "G711 aLaw" and "G711 uLaw" should be selected
Not obvious to me initially, but to use X-Lite v4.6x (direct without a SIP provider):
- Softphone -> Account Settings -> Advanced (tab)
- Send outgoing requests directly to target (CHECKED)
- Softphone -> Preferences -> Calls
- Has the DTMF choices
- using a preconfigured SIP client might be using the tunnel of the SIP account which, at least in my case (SIPgate Germany) failed Paul Libbrecht 15:16, 17 February 2011 (UTC)
- I heard from someone that access using the application Telephone on MacOSX is working like a charm Paul Libbrecht 15:16, 17 February 2011 (UTC)
- I have been using X-Meeting happily but had issues with DTMF (it worked once over 10) Paul Libbrecht 15:16, 17 February 2011 (UTC)
- using SJphone, a venerable old SIP client for many platforms, the default shipped profile "PC-to-PC" can be used in open network conditions. Using the direct SIP numbers (e.g. sip:00NNNN@voip.w3.org) worked like a charm to avoid the DTMF issues. Paul Libbrecht 15:16, 17 February 2011 (UTC)
- Ekiga worked for me on Windows 7 yesterday, not today :-(
- Installed X-Lite, used it with my ekiga.net account no problem [HST 2011-02-18 170600Z]
- After some fiddling about, I successfully got a free SIP account with SIP2SIP. I am using both X-Lite 4 on my MBP as well as CSipSimple on my Android phone, both work well and allow me to make really good quality calls to Zakim from Ireland. [Joshue]
Accessible SIP Clients
On Linux <a href="http://www.twinklephone.com/">Twinkle's</a> command-line interface is accessible although its KDE interface currently is not. We understand the QT toolkit used by KDE is being wired into AT-SPI2/ATK which will likely make Twinkle's graphical interface accessible as well in the future.
GroundWire, available from Apple apps, is another accessible SIP client.
Linphone, originally a Linux client, is also available and accessible on Macintosh, IOS, and Android. Linphone is also available on Windows where its builtin command line client, linphonec, is accessible with screen readers (like JAWS and NVDA. more below), though its graphical client is not accessible.
Linphone on Windows (works with screen readers)
Linphone on Windows can be accessed through the command prompt. This has been tested on Windows 7 with Jaws 12 and NVDA 2011.2 and seems to work.
Quick user guide:
Install the Windows GUI client. The installlation seems accessible with Jaws/NVDA.
Open the Windows command prompt and move to the directory where Linphone is installed. For example:
cd c:\Program files\Linphone\bin
Run Linphone using the following command:
Ready. Warning: Video is disabled, use -v or -c or -d to enable.
Call Zakim-SIP using the following command:
Establishing call id to <email@example.com>, assigned ID 1
Call 1 to <sip:firstname.lastname@example.org> in progress
Call 1 to <sip:email@example.com> ringing
Call 1 with <sip:firstname.lastname@example.org> connected
Call answered by <sip:email@example.com>
Media streams established with <sip:firstname.lastname@example.org> for call 1
Enter the passcode for the telecon followed by #. For example:
When you finish the telecon, end the call using the following command:
Use Control c to close Linphone before typing exit to close the Windows command prompt.
In FreeSWITCH it is enough to register a new extension in the dialplan. Tested configuration in conf/dialplan/default:
<include> <extension name="voip.w3.org"> <condition field="destination_number" expression="^(w3c|932)$"> <action application="bridge" data="email@example.com"/> </condition> </extension> </include>
If you execute reloadxml in the FreeSWITCH console you can now call w3c or 932.
Give your Asterisk a Zakim Extension
If you run your own PBX server, e.g. Asterisk, you might want to add an extension (or two) to provide direct access to Zakim on all your registered clients (hardware and software). Here's how to do that in Asterisk:
Under the [default] context in your extensions.conf file, create an extension like one of the following:
exten => 2220,1,Dial(SIP/voip.w3.org/zakim) ;W3C Zakim exten => 2221,1,Dial(SIP/voip.w3.org/002119) ;W3C Zakim for HTML-A11Y Task Force (TF)
Reload your Asterisk configurations and any of your registered devices should now be able to dial 2220 to access Zakim, or 2221 to access the HTML-A11Y Task Force Zakim extension directly. You will, of course, want to choose extensions appropriate to your environment. These extensions are illustrative, though they are also in use on asterisk.rednote.net.