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<h4>Client Instructions</h4>
Below are instructions on configuring a SIP client to work with SDF's VOIP
service. Please see the <a href="?tutorials/sdf_voip">SDF VOIP tutorial</a>
for more information.
<h5>List of Clients</h5>
<li><a href="#linphone_for_ios">Linphone for iOS</a></li>
<li><a href="#ekiga_on_multi">Ekiga for Windows and linuxy things</a></li>
<li><a href="#grandstream">Grandstream GXP2000</a></li>
<li><a href="#handytone">Grandstream Handytone 286</a></li>
<li><a href="#android_native">Android native client</a></li>
<h4><a name="client_instructions">Client Instructions</a></h4>
<dt><h5><a name="linphone_for_ios">Linphone for iOS</a></h5></dt>
<dd>Linphone may be installed by searching for "Linphone" in the App
Store or by clicking <a
href=http://itunes.apple.com/us/app/linphone/id360065638?mt=8>here</a> to
open the App Store page on your iOS device.</dd>
<dd>Once you have installed the application, open the <b>general
settings</b> application on your iOS device. There are no settings within
the app itself. In the general settings application, scroll down until
you find an entry for Linphone. Tap it to open Linphone's settings</dd>
<dd>The following settings need to be filled:</dd>
<dd><b>User name</b> - Enter your SIP extension number</dd>
<dd><b>Password</b> - Enter your SIP password</dd>
<dd><b>Domain</b> - Enter <b>sip.sdf.org</b></dd>
<dd><b>Transport</b> - Be sure it is set to <b>UDP</b></dd>
<dd><b>Background Mode</b> - Turn this setting on if you wish to allow
Linphone to run and receive calls in the background</dd>
<dt><h5><a name="ekiga_on_multi">Ekiga for Windows and linuxy things</a></h5></dt>
Download and install <a href="http://ekiga.org/">ekiga</a>.
During the initial run, a wizard will appear. Cancel out of the
wizard and <b>manually add an account</b> with the steps
below. More information can also be found on ekiga's <a
<li><b>Cancel out of the wizard</b>, if it is still running.</li>
<li><b>Add an account</b> through "Edit&rarr;Accounts"</li>
<li>In the pop up, go to <b>"Accounts&rarr;Add a SIP account"</b> and
fill in the fields.</li>
<li>Give the account a name in the Name field.</li>
<li>For <b>Registrar</b>, use <b>sip.sdf.org</b>.</li>
<li>For <b>User</b>, use the <b>numeric extension ID</b> supplied in the
<li>For <b>Authentication User</b>, use the <b>numeric
extension ID</b> supplied in the email.</li>
<li>For Password, use the password supplied in the email.</li>
<li>For <b>Timeout</b>, make sure the value is large like
<li>Select the "Enable Account" box.</li>
<li>Select OK to complete this process.</li>
<li>Have fun.</li>
<dt><h5><a name="grandstream">Grandstream GXP2000</a></h5></dt>
<p>The Grandstream GXP2000 is an office SIP phone. It is fairly
straightforward to setup via the phone's web interface. Below is a
screenshot with highlighted options needed for it to register and work
properly with SDF's VOIP system. Here are some items to note:</p>
<li>Replace 1134 with your extension ('SIP User ID' and 'Authenticate
ID' options), and slugmax ('Name' option) with your own user ID (or
whatever you want - the 'Name' option gets displayed on the phone's
LCD display, but is not useful otherwise).</li>
<li>The 'Voice Mail UserID' is really the extension number for the
voicemail system. Currently, this is 1085. Once setup, hitting the
phone's 'msg' button will dial this extension.</li>
<li>I have NAT traversal disabled, as I have my home router configured to
forward UDP port 5060 to the phone's IP. You may need NAT traversal,
depending on your setup. <b>I've found it necessary with a standard UDP port
forwarding setup to select 'No, but send keep-alive' for this option,
without it longer duration calls were being dropped after about 10
<li>Make sure you choose 'via RTP' for the 'Send DTMF' configuration
option. Otherwise the voicemail system will not allow you to
<li>The 'Authenticate Password' option is the password given to you in
the VOIP signup email</li>
<p><img src="tutorials/images/grandstream_voip.png" /></p>
<dt><h5><a name="handytone">Grandstream Handytone 286</a></h5></dt>
<p>The Grandstream Handytone 286 is a simple analog telephone adapter.
It can allow you to use any analog phone with the SDF VOIP service.
It can be configured using the built-in web interface or through voice
prompts by dialing *** on an analog phone.</p>
<li>Add 'sip.sdf.org' to the SIP Server field</li>
<li>Add your extension to the SIP User ID and Authenticate ID fields</li>
<li>Add your VOIP password provided from 'maint' to the Authenticate
Password field</li>
<li>Add your name to the Name field, if you wish</li>
<li>Set Use DNS SRV to 'Yes'</li>
<li>Set NAT Traversal to 'No'</li>
<li>UN-check 'in-audio' and check 'via RTP (RFC2833)' for Send DTMF</li>
<p>You should also forward UDP port 5060 to the Handytone's IP address
through your router. It may be a good idea to set the Handytone to
a static IP address, which can be done on the Basic Settings tab.
Don't forget when doing this to add all the relevent fields, including
a proper DNS server, else the Handytone won't be able to resolve the
sip.sdf.org address.</p>
<dt><h5><a name="android_native">Android native client</a></h5></dt>
<p>Android 2.3 and up seems to have a built-in SIP client. The screenshots
below are from Android 4.3. This is tested with wifi data, and 3G/HSPA on
tmo. Some carriers or specific android versions may disable SIP calling
over cellular data.</p>
<li>From the Android home screen, tap the phone icon to go to the dialpad.
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/01-home.png"></p></li>
<li>Tap the settings icon on the lower left button.
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/02-dialpad.png"></p></li>
<li>Select the Settings option.
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/03-dialpad_settings.png"></p></li>
<li>Optionally, select the "Use Internet calling" option to select when
to use SIP calling, and when to use the regular calling function. In
this case, I slect "Only for Internet calls" only for SIP calls. (See
contacts discussion below.)
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/04-call_settings.png"></p>
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/05-internet_calling_setting.png"></p>
<li>Select Accounts to create the SIP account.
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/05.5-call_settings2.png"></p></li>
<li>Optionally, select "Receive incoming calls" if you want to receive
SIP calls on this phone.
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/06-add_account.png"></p></li>
<li>Select the "Add Account" option near the bottom.
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/06.5-add_account.png"></p></li>
<li>On the next screen, tap Username and enter only the extension
number. Tap password and enter your password. Tap Server and enter
sip.sdf.org. Optionally, select "Set as primary account." This option
does not seem to be necessary to make an outbound call. Maybe this is
only used when multiple SIP accounts are configured? Tap save to save the settings.
Now things should be ready for a test call.
<p><img src="/tutorials/images/sdf_voip_client_instructions/native_android/07-account_details.png"></p></li>
<p>When making a call from the dialpad, there does not seem to be a way to
enter the @ sign. If "Use Internet calling" is set to "For all calls", then
this is not an issue: just type the extension number and tap call..</p>
<p>Another way to make SIP calls is to add the SIP number
(nnnn@sip.sdf.org) into the contacts, and select the number from the
address book. Android seems to detect the @ sign and automatically switch
to internet calls regardless of what the "internet calling" setting and the
"Set as primary account" setting is set to.</p>
<p>Finally, Google Contacts also has the option of labeling a number as
"Internet call" which will trigger SIP calling as well.</p>
This tutorial is far from complete. Wanna make it better? Edit it!
<cite>$Id: sdf_voip_client.html,v 1.2 2013/09/07 18:41:38 wliao Exp $</cite>