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<h4>Setting up a SIP phone with SDF VoIP</h4>
<p>SDF Voice over Internet Protocol Telephony is a service
feature available to the <a href="">ARPA</a>,
<a href="">META</a> and Validated
<a href="">USERS</a> membership levels.
It allows an SDF user to make voice and video calls to
other SDF users and to linked Telephony services via a SIP client
on a 3G or WIFI cellphone, WIFI enabled mobile device, or an
Internetworked computer.</p>
<p>SDF VoIP set up and server configuration can be accessed via 'maint' at
the shell ('maint' -> v -> p).</p>
<p>Below is a list of known-to-work <a href="#sip_clients">sip
clients</a>, and <a href="#general_instructions">general
instructions</a> on configuring and setting up a SIP client.
There may also be <a href="/?tutorials/sdf_voip_client">client-specific
instructions</a> as well.</p>
<p>Wikipedia also has a list of notable <a
<h4>Tested SIP Hardware</h4>
<h5>Routers (with SIP capabilities)</h5>
<li><a href="">FritzBox 7340</a></li>
<li>Grandstream GXP2000</li>
<li>Grandstream DP750 with DP720 handset</li>
<li>snom 300</li>
Cisco SPA-525G2
<li><a href="">Cisco 7940G</a> (<a href="/?tutorials/sdf_voip_7940g">Client Instructions</a>)</li>
<h5>Analog Phone Adaptor</h5>
<li><a href="">Grandstream HandyTone 286</a></li>
<li>Grandstream HT801</li>
<h4><a name="sip_clients">Tested SIP clients</a></h4>
<li><a href="">Linphone</a> for iOS, Android, and Linux</li>
<li><a href="">Bria Solo</a> (formerly X-lite) Windows, Mac OS X - Desktop Client</li>
<li><a href="">Blink</a> Windows, Mac OS X, Linux - Desktop Client</li>
<li><a href="">Ekiga</a> Windows and Linux - Supports Video</li>
<li>Unlocked 2.3 (Gingerbread), 4.0 and up (including Android 10) handsets have native
SIP support</li>
<li><a href="">Fring</a> on the Apple iPhone, iPod, iPad</li>
<li><a href="">Acrobits Groundwire</a></li>
<li>Built-in client on Nokia phones and tablets (N810)</li>
<h5>Mac OS X</h5>
<li><a href=""
>Telephone</a>, available through the App Store.</li>
<li><a href="">MicroSIP</a> Open Source SIP client for Windows</li>
<h4><a name="general_instructions">General Instructions</a></h4>
All SIP clients must have at least 3 bits of information:
<dd>The "username" used for authentication. Currently, the
<b>extension is a series of numbers</b> such as "6000". A <a
href="">directory</a> of
users is also available. The assigned extension will be given in
an email. Do not use your SDF username.</dd>
<dd>This may also be known as the server to connect with. Use
<b></b> as the server; use the default port of
<b>5060/udp</b>. Please note: some clients infer the domain from
the extension. In this case, the extension will be in the form of
"extension@domain", or "sip:extension@domain"; eg,
"" would be a valid username that includes the
<dd>This is the password used to authenticate to the SIP server. This
will be provided in the email as well, but may change very soon
in the future. The password can be reset via the maint command.</dd>
<h4><a name="client_instructions">Client Instructions</a></h4>
Please see the <a href="/?tutorials/sdf_voip_client">client instructions</a>
page for specific client instructions.
<h4><a name="voicemail">Voicemail</a></h4>
Voicemail messages will be automatically mailed to your SDF email address in
WAV format.
<h4><a name="PSTN">PSTN</a></h4>
Want to call the public switched telephone network? Read the info <a
This tutorial is far from complete. Wanna make it better? Edit it!
<cite>$Id: sdf_voip.html,v 1.43 2019/12/13 03:39:56 jasmaz Exp $</cite>