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	<title>Voip Service Review &#187; linux softphone</title>
	<atom:link href="http://voiprelays.com/archives/tag/linux-softphone/feed" rel="self" type="application/rss+xml" />
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	<description>Voice Over Internet Protocol - Information and Guides</description>
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		<title>Ekiga &#8211; Open Source Softphone</title>
		<link>http://voiprelays.com/archives/50/ekiga-open-source-softphone</link>
		<comments>http://voiprelays.com/archives/50/ekiga-open-source-softphone#comments</comments>
		<pubDate>Fri, 22 Jan 2010 04:27:36 +0000</pubDate>
		<dc:creator>voip</dc:creator>
				<category><![CDATA[VOIP Reviews]]></category>
		<category><![CDATA[VOIP Software]]></category>
		<category><![CDATA[free softphone]]></category>
		<category><![CDATA[free voip softphone]]></category>
		<category><![CDATA[free voip software]]></category>
		<category><![CDATA[linux]]></category>
		<category><![CDATA[linux softphone]]></category>
		<category><![CDATA[open source]]></category>
		<category><![CDATA[softphone]]></category>
		<category><![CDATA[software]]></category>

		<guid isPermaLink="false">http://voiprelays.com/?p=50</guid>
		<description><![CDATA[
Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet.
It supports HD sound quality and video up to DVD size and quality.
It is interoperable with many other standard compliant softwares, hardwares and service providers as it uses both the major telephony standards (SIP and H.323).
Features

 Ease [...]]]></description>
			<content:encoded><![CDATA[<p><img class="alignnone size-full wp-image-51" title="ekiga" src="http://voiprelays.com/wp-content/uploads/2010/01/ekiga_s.png" alt="" width="274" height="311" /></p>
<p>Ekiga (formely known as GnomeMeeting) is an open source SoftPhone, Video Conferencing and Instant Messenger application over the Internet.</p>
<p>It supports HD sound quality and video up to DVD size and quality.</p>
<p>It is interoperable with many other standard compliant softwares, hardwares and service providers as it uses both the major telephony standards (SIP and H.323).</p>
<p><strong>Features</strong></p>
<ul>
<li> Ease of use with a modern Graphical User Interface.</li>
<li> Audio and Video free calls through the internet.</li>
<li> Free Instant Messaging through the internet with Presence support.</li>
<li> Audio (and video) calls to landlines and cell phones with support to the cheapest service providers.</li>
<li> High Definition Sound (wideband) and Video Quality up to DVD quality (high framerate, state of the art quality codec and frame size).</li>
<li> Free of choice of the service provider.</li>
<li> SMS to cell phones if the service provider supports it (like the default provider).</li>
<li> Standard Telephony features support like Call Hold, Call Transfer, Call Forwarding, DTMF.</li>
<li>Remote and Local Address Book support: Remote Address Book support with authentification using the standard LDAP technology, Local Address support in Gnome (Evolution).</li>
<li> Multi platform: Windows and GNU/Linux</li>
<li>Wide interoperability: Ekiga use the main deployed stantards for telephony protocols (SIP and H.323) and has been tested with a wide range of softphones, hardphones, PBX and service providers.</li>
</ul>
<p><strong>Call</strong></p>
<ul>
<li> Call Hold: This effectively pauses Video &amp; Audio transmission.</li>
<li> Call Transfer: You can transfer the remote user to another user.</li>
<li> Call Forwarding on No Answer, on Busy, on Always: This allows you to con?gure Ekiga to forward incoming calls to a speci?ed user.</li>
<li> Message Waiting Indications Support</li>
<li> Dialpad: This allow you to dial numbers using a graphical alphanumeric keypad.</li>
<li> DTMFs Support: This feature is necessary when using services asking you to dial numbers.</li>
<li> Calls History: This is a convenient way about all outgoing and incoming calls.</li>
<li> Call Monitoring: Statistics about the network traffic caused by Ekiga are displayed in the status bar.</li>
<li>Enum Support: Enum is a standard method to provide a unified numbering system between the public switched telephone network (PSTN) and various VoIP providers. It is officially deployed country wide.</li>
</ul>
<p><strong>Operating System</strong></p>
<p>Linux, Windows</p>
<p><strong>Download</strong></p>
<p><a href="http://ekiga.org/download-ekiga-binaries-or-source-code">Download Site</a></p>
<div id="crp_related"><br><br><br><br><h3>Related Posts:</h3><ul><li><a href="http://voiprelays.com/archives/34/eyebeam-softphone" rel="bookmark" class="crp_title">eyeBeam Softphone</a></li><li><a href="http://voiprelays.com/archives/38/bol-sipphone-v1-6-free-softphone" rel="bookmark" class="crp_title">BOL SIPPhone v1.6 - Free Softphone</a></li><li><a href="http://voiprelays.com/archives/47/twinkle-free-linux-softphone" rel="bookmark" class="crp_title">Twinkle - Free Linux Softphone</a></li><li><a href="http://voiprelays.com/archives/29/sjphone-free-softphone" rel="bookmark" class="crp_title">SJphone - Free Softphone</a></li><li><a href="http://voiprelays.com/archives/75/cisco-spa942-4-line-ip-phone-with-2-port-switch" rel="bookmark" class="crp_title">Cisco SPA942 4-line IP Phone with 2-port Switch</a></li></ul></div>]]></content:encoded>
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		</item>
		<item>
		<title>Twinkle &#8211; Free Linux Softphone</title>
		<link>http://voiprelays.com/archives/47/twinkle-free-linux-softphone</link>
		<comments>http://voiprelays.com/archives/47/twinkle-free-linux-softphone#comments</comments>
		<pubDate>Fri, 22 Jan 2010 04:12:43 +0000</pubDate>
		<dc:creator>voip</dc:creator>
				<category><![CDATA[VOIP Reviews]]></category>
		<category><![CDATA[VOIP Software]]></category>
		<category><![CDATA[free linux voip]]></category>
		<category><![CDATA[free linux voip software]]></category>
		<category><![CDATA[free softphone]]></category>
		<category><![CDATA[free software]]></category>
		<category><![CDATA[free voip softphone]]></category>
		<category><![CDATA[linux]]></category>
		<category><![CDATA[linux softphone]]></category>

		<guid isPermaLink="false">http://voiprelays.com/?p=47</guid>
		<description><![CDATA[
Twinkle is a softphone for your voice over IP and instant messaging communcations using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages.
Features
In addition to making basic voice calls Twinkle provides you the following features:
* [...]]]></description>
			<content:encoded><![CDATA[<p><img class="alignnone size-full wp-image-48" title="twinkle" src="http://voiprelays.com/wp-content/uploads/2010/01/twinkle.png" alt="" width="447" height="410" /></p>
<p>Twinkle is a softphone for your voice over IP and instant messaging communcations using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages.</p>
<p><strong>Features</strong></p>
<p>In addition to making basic voice calls Twinkle provides you the following features:</p>
<p>* 2 call appearances (lines)<br />
* Multiple active call identities<br />
* Custom ring tones<br />
* Call Waiting<br />
* Call Hold<br />
* 3-way conference calling<br />
* Mute<br />
* Call redirection on demand<br />
* Call redirection unconditional<br />
* Call redirection when busy<br />
* Call redirection no answer<br />
* Reject call redirection request<br />
* Blind call transfer<br />
* Call transfer with consultation (attended call transfer)<br />
* Reject call transfer request<br />
* Call reject<br />
* Repeat last call<br />
* Do not disturb<br />
* Auto answer<br />
* Message Waiting Inidication<br />
* Voice mail speed dial<br />
* User defineable scripts triggered on call events<br />
E.g. to implement selective call reject or distinctive ringing<br />
* RFC 2833 DTMF events<br />
* Inband DTMF<br />
* Out-of-band DTMF (SIP INFO)<br />
* STUN support for NAT traversal<br />
* Send NAT keep alive packets when using STUN<br />
* NAT traversal through static provisioning<br />
* Persistent TCP connections for NAT traversal<br />
* Missed call indication<br />
* History of call detail records for incoming, outgoing, successful and missed calls<br />
* DNS SRV support<br />
* Automatic failover to an alternate server if a server is unavailable<br />
* Other programs can originate a SIP call via Twinkle, e.g. call from address book<br />
* System tray icon<br />
* System tray menu to quickly originate and answer calls while Twinkle stays hidden<br />
* User defineable number conversion rules<br />
* Simple address book<br />
* Support for UDP and TCP as transport for SIP<br />
* Presence<br />
* Instant messaging<br />
* Simple file transfer with instant message<br />
* Instant message composition indication<br />
* Command line interface (CLI)</p>
<p><strong>VoIP security</strong></p>
<p>* Secure voice communication by ZRTP/SRTP<br />
* MD5 digest authentication support for all SIP requests<br />
* AKAv1-MD5 digest authentication support for all SIP requests<br />
* Identity hiding</p>
<p><strong>Download</strong></p>
<p><a href="http://www.xs4all.nl/~mfnboer/twinkle/download.html">Download Site</a></p>
<div id="crp_related"><br><br><br><br><h3>Related Posts:</h3><ul><li><a href="http://voiprelays.com/archives/50/ekiga-open-source-softphone" rel="bookmark" class="crp_title">Ekiga - Open Source Softphone</a></li><li><a href="http://voiprelays.com/archives/34/eyebeam-softphone" rel="bookmark" class="crp_title">eyeBeam Softphone</a></li><li><a href="http://voiprelays.com/archives/38/bol-sipphone-v1-6-free-softphone" rel="bookmark" class="crp_title">BOL SIPPhone v1.6 - Free Softphone</a></li><li><a href="http://voiprelays.com/archives/18/call-union-voip-telephony" rel="bookmark" class="crp_title">Call Union VOIP Telephony</a></li><li><a href="http://voiprelays.com/archives/42/zoiper-classic-free-softphone" rel="bookmark" class="crp_title">Zoiper Classic - Free Softphone</a></li></ul></div>]]></content:encoded>
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