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	<title>voip-sip.org - Internet VoIP calls, International SIP termination, VoIP hardware</title>
	<atom:link href="http://www.voip-sip.org/feed/" rel="self" type="application/rss+xml" />
	<link>http://www.voip-sip.org</link>
	<description>We share knowledge</description>
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		<title>Plain explain: IP-phone</title>
		<link>http://www.voip-sip.org/plain-explain-ip-phone/</link>
		<comments>http://www.voip-sip.org/plain-explain-ip-phone/#comments</comments>
		<pubDate>Wed, 27 Feb 2013 13:47:03 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Cisco]]></category>
		<category><![CDATA[SIP protocol]]></category>
		<category><![CDATA[SIP termination]]></category>
		<category><![CDATA[VoIP calls quality]]></category>
		<category><![CDATA[How it works]]></category>
		<category><![CDATA[ip-phone]]></category>
		<category><![CDATA[voip phone]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=820</guid>
		<description><![CDATA[<p></p> <p>We all strive to get a quality telephony, but when we do some analyze of the voice quality issues and most common reasons that may lead to the problems with the VoIP service quality we should take into account the VoIP equipment, including IP-phones.</p> Looking closer: what is an IP-phone <p>VoIP phone is a [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/plain-explain-ip-phone/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Plain explain: What is SIP termination and what is SIP origination?</title>
		<link>http://www.voip-sip.org/plain-explain-what-is-sip-termination-and-what-is-sip-origination/</link>
		<comments>http://www.voip-sip.org/plain-explain-what-is-sip-termination-and-what-is-sip-origination/#comments</comments>
		<pubDate>Tue, 26 Feb 2013 12:19:08 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[SIP termination]]></category>
		<category><![CDATA[VoIP service]]></category>
		<category><![CDATA[SIP origination]]></category>
		<category><![CDATA[SIP provider]]></category>
		<category><![CDATA[VoIP calls]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=800</guid>
		<description><![CDATA[<p>On the telecom market many (if not all) TELCO companies (including SIP providers of course) offer services that is based on  VoIP. Almost all of them may provide a SIP termination and SIP origination. In this short article I will try to define these terms (SIP termination and SIP origination) and briefly explain what is [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/plain-explain-what-is-sip-termination-and-what-is-sip-origination/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>US fixed VoIP market to see steady growth</title>
		<link>http://www.voip-sip.org/us-fixed-voip-market-to-see-steady-growth/</link>
		<comments>http://www.voip-sip.org/us-fixed-voip-market-to-see-steady-growth/#comments</comments>
		<pubDate>Fri, 11 Jan 2013 15:58:23 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[VoIP industry news]]></category>
		<category><![CDATA[US]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=756</guid>
		<description><![CDATA[<p>Those who have been paying attention to voice communications trends over the past couple of years should not be surprised by the numbers from the recent TechNavio report regarding fixed VoIP services in the United States. The company said that between 2012 and 2016, the market will grow at a rate of about 10.15 %per year.</p> <p>&#8220;One of [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/us-fixed-voip-market-to-see-steady-growth/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Save money with VoIP and unified communications</title>
		<link>http://www.voip-sip.org/save-money-with-voip-and-unified-communications/</link>
		<comments>http://www.voip-sip.org/save-money-with-voip-and-unified-communications/#comments</comments>
		<pubDate>Fri, 11 Jan 2013 15:53:29 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[VoIP industry news]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=754</guid>
		<description><![CDATA[<p>While travel and procurement costs have been rising for many aspects of business, such as travel and procurement of certain types of technology, Mini Swamy, a TMCnet contributor, wrote that VoIP and unified communications can help companies save some money. She looked at a report by Digitalolympus.com, which said the right implementation of these programs can help companies of [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/save-money-with-voip-and-unified-communications/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>IP telephony can help level the playing field for small businesses</title>
		<link>http://www.voip-sip.org/ip-telephony-can-help-level-the-playing-field-for-small-businesses/</link>
		<comments>http://www.voip-sip.org/ip-telephony-can-help-level-the-playing-field-for-small-businesses/#comments</comments>
		<pubDate>Fri, 11 Jan 2013 15:51:12 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[IT news]]></category>
		<category><![CDATA[VoIP industry news]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=752</guid>
		<description><![CDATA[<p>For years, small companies have been tied down by carrier-based phone systems that are expensive and cumbersome. The Guardian said there are ways to help small businesses compete with big businesses and one way to do that is by adopting an IP telephony system in lieu of phone lines.</p> <p>&#8220;While a business may be based in, say, [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/ip-telephony-can-help-level-the-playing-field-for-small-businesses/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>The loss of a landline means big change in communications</title>
		<link>http://www.voip-sip.org/the-loss-of-a-landline-voip/</link>
		<comments>http://www.voip-sip.org/the-loss-of-a-landline-voip/#comments</comments>
		<pubDate>Fri, 11 Jan 2013 15:49:15 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[VoIP industry news]]></category>
		<category><![CDATA[IP telephony]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=750</guid>
		<description><![CDATA[<p>The loss of a landline is something that every business will have to deal with over the next few years. A recent report by inetwork, a division of ATLANTIC-ACM, shows that 74 percent of those polled believe that the death of the office landline and deskphone will be one of the most significant forces in [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/the-loss-of-a-landline-voip/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Ubuntu (Linux) on your phone? Yes! And now officially!</title>
		<link>http://www.voip-sip.org/ubuntu-linux-on-your-phone-yes-and-now-officially/</link>
		<comments>http://www.voip-sip.org/ubuntu-linux-on-your-phone-yes-and-now-officially/#comments</comments>
		<pubDate>Fri, 04 Jan 2013 10:17:29 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Open source]]></category>
		<category><![CDATA[Ubuntu]]></category>
		<category><![CDATA[linux. phone. smartphone under linux]]></category>
		<category><![CDATA[ubuntu]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=745</guid>
		<description><![CDATA[News from Canonical: Ubuntu is coming to the phone <p>When we began developing Unity a few years ago, the aim was to create a single family of interfaces that work the same way on different devices. This means that unlike most of our rivals, we are able to use a single underlying OS across all [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/ubuntu-linux-on-your-phone-yes-and-now-officially/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>QoS For FaceTime, bandwidth requirements and Firewall config</title>
		<link>http://www.voip-sip.org/qos-for-facetime-bandwidth-requirements-and-firewall-config/</link>
		<comments>http://www.voip-sip.org/qos-for-facetime-bandwidth-requirements-and-firewall-config/#comments</comments>
		<pubDate>Thu, 03 Jan 2013 09:25:28 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Apple]]></category>
		<category><![CDATA[FaceTime]]></category>
		<category><![CDATA[Prioritization and traffic shaping]]></category>
		<category><![CDATA[QoS - Quality of Service]]></category>
		<category><![CDATA[VoIP calls quality]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=736</guid>
		<description><![CDATA[Facetime &#8211; Calls Quality, QoS and firewall ports in use <p>Video calling will become an increasingly widespread form of communication in coming years, moving from<br /> broad adoption in the consumer market to selective endorsement in the corporate world. The same executives<br /> who video-call their families when on the road will also use FaceTime [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/qos-for-facetime-bandwidth-requirements-and-firewall-config/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Merry Christmas and Happy New Year!</title>
		<link>http://www.voip-sip.org/merry-christmas-and-happy-new-year/</link>
		<comments>http://www.voip-sip.org/merry-christmas-and-happy-new-year/#comments</comments>
		<pubDate>Tue, 25 Dec 2012 00:01:56 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Non VoIP news]]></category>
		<category><![CDATA[Happy holodays]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=730</guid>
		<description><![CDATA[<p>Dear Friends, Readers, Writers and all colleagues of VoIP-SIP.ORG,</p> <p>We would like to thank all those with whom we have done some work on Open Source projects. A lot of stuff was done at this year and it was great to work together with you guys!</p> <p>Merry Christmas and Happy New Year! Stay crazy like [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/merry-christmas-and-happy-new-year/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>How to Configure Axvoice Equipment?</title>
		<link>http://www.voip-sip.org/how-to-configure-axvoice-equipment/</link>
		<comments>http://www.voip-sip.org/how-to-configure-axvoice-equipment/#comments</comments>
		<pubDate>Tue, 11 Dec 2012 12:55:36 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[VoIP service]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=718</guid>
		<description><![CDATA[<p>INTRODUCTION TO AXVOICE:</p> <p>Axvoice is a well known low cost VoIP phone service provider that provides its services in USA and Canada. Axvoice offers a huge variety of residential and business plans. Pay as you go is their cheapest plan ($4.99/month). This plan allows you to pay only when you need to call. $8.25/month is [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/how-to-configure-axvoice-equipment/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Cloud VoIP vs. on-premise VoIP: Choosing the right one for your business</title>
		<link>http://www.voip-sip.org/cloud-voip-vs-on-premise-voip-choosing-the-right-one-for-your-business/</link>
		<comments>http://www.voip-sip.org/cloud-voip-vs-on-premise-voip-choosing-the-right-one-for-your-business/#comments</comments>
		<pubDate>Thu, 06 Dec 2012 12:29:18 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Cloud VoIP]]></category>
		<category><![CDATA[Cloud]]></category>
		<category><![CDATA[voip]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=661</guid>
		<description><![CDATA[<p>As VoIP continues its growth in the business market, more and more companies are finding out that selecting the right type of phone solution sometimes can be a bit confusing.  This is because there are multiple options within the VoIP phone market.  One of the most common areas of misunderstanding is the distinction between on-premise [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/cloud-voip-vs-on-premise-voip-choosing-the-right-one-for-your-business/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>26 terabits per second data transmission achieved</title>
		<link>http://www.voip-sip.org/26-terabits-per-second-data-transmission-achieved/</link>
		<comments>http://www.voip-sip.org/26-terabits-per-second-data-transmission-achieved/#comments</comments>
		<pubDate>Tue, 04 Dec 2012 09:30:59 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[IT news]]></category>
		<category><![CDATA[bandwidth]]></category>
		<category><![CDATA[bandwidth speeds]]></category>
		<category><![CDATA[technology news]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=653</guid>
		<description><![CDATA[<p></p> <p>With video content consuming ever more bandwidth, the need for faster data transmission rates has never been greater. Now a team of scientists at Germany&#8217;s Karlsruhe Institute of Technology (KIT) are claiming a world record in data transmission with the successful encoding of data at a rate of 26 terabits per second on a [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/26-terabits-per-second-data-transmission-achieved/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>A fully functional VoIP Client (SIP) finally released (free download)</title>
		<link>http://www.voip-sip.org/a-fully-functional-voip-client-sip-finally-released-free-download/</link>
		<comments>http://www.voip-sip.org/a-fully-functional-voip-client-sip-finally-released-free-download/#comments</comments>
		<pubDate>Thu, 29 Nov 2012 10:28:55 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Open source]]></category>
		<category><![CDATA[Softphone VoIP]]></category>
		<category><![CDATA[Linux softphone]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=631</guid>
		<description><![CDATA[<p>Great news for today, guys! The legendary Linux softphone is back for more (available for Windows also)!</p> <p>Three years after the 3.2 release, Ekiga 4.0 aka &#8220;The Victory Release&#8221; is finally available.</p> <p>This is a major release with many major improvements.</p> <p></p> <p>Where you can download it:</p> <p>Ekiga sources and Windows binaries are available at:</p> <p><a [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/a-fully-functional-voip-client-sip-finally-released-free-download/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Fundamentals of SIP from Cisco :-)</title>
		<link>http://www.voip-sip.org/fundamentals-of-sip-from-cisco/</link>
		<comments>http://www.voip-sip.org/fundamentals-of-sip-from-cisco/#comments</comments>
		<pubDate>Thu, 23 Aug 2012 11:04:57 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Cisco]]></category>
		<category><![CDATA[SIP protocol]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=607</guid>
		<description><![CDATA[<p>Session Initiation Protocol, SIP, is poised to continue its reshaping of your collaboration and communication network. Hope you will enjoy watching this short, but very good video below!</p> <p></p> <p>This Fundamentals animation was produced as part of a TechWiseTV episode 66 &#8216;SIP, Session Management and Beyond. Just for the reference.</p> <p><br /> <br /> </p>]]></description>
		<wfw:commentRss>http://www.voip-sip.org/fundamentals-of-sip-from-cisco/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>WebRTC from Google: making real-time communication free to implement</title>
		<link>http://www.voip-sip.org/webrtc-from-google-making-real-time-communication-free-to-implement/</link>
		<comments>http://www.voip-sip.org/webrtc-from-google-making-real-time-communication-free-to-implement/#comments</comments>
		<pubDate>Thu, 12 Jul 2012 11:30:36 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Android]]></category>
		<category><![CDATA[Google Voice]]></category>
		<category><![CDATA[Open source]]></category>
		<category><![CDATA[browser]]></category>
		<category><![CDATA[calls]]></category>
		<category><![CDATA[google]]></category>
		<category><![CDATA[html5]]></category>
		<category><![CDATA[record]]></category>
		<category><![CDATA[web calls]]></category>
		<category><![CDATA[webrtc]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=568</guid>
		<description><![CDATA[<p>After Microsoft bought Skype for US$8.5 billion, Google has released a developer preview of WebRTC &#8211; an open framework enabling implementation of voice and video Real Time Communications in the browser with the use of HTML 5 and JavaScript APIs.</p> <p>It&#8217;s possible that the Skype acquisition could mean restricting the client from technologies and devices [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/webrtc-from-google-making-real-time-communication-free-to-implement/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>What is VoIP termination</title>
		<link>http://www.voip-sip.org/what-is-voip-termination/</link>
		<comments>http://www.voip-sip.org/what-is-voip-termination/#comments</comments>
		<pubDate>Wed, 11 Jul 2012 13:03:24 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[SIP protocol]]></category>
		<category><![CDATA[origination]]></category>
		<category><![CDATA[termination]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=536</guid>
		<description><![CDATA[Many people keep asking me &#8211; what is VoIP Termination? <p>There is an &#8220;official&#8221; answer: VoIP Termination is the process of routing a VoIP telephone call (or calls flow) from one telephone service provider to another. What is that meant? Ok, lets imagine you have some phone lines (IP-phones or VoIP gateways with regular phones [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/what-is-voip-termination/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Introduction to Voice over IP (VoIP)</title>
		<link>http://www.voip-sip.org/introduction-to-voice-over-ip/</link>
		<comments>http://www.voip-sip.org/introduction-to-voice-over-ip/#comments</comments>
		<pubDate>Wed, 11 Jul 2012 07:07:55 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Mobile VoIP]]></category>
		<category><![CDATA[Open source]]></category>
		<category><![CDATA[SIP protocol]]></category>
		<category><![CDATA[how to voip]]></category>
		<category><![CDATA[voip basics]]></category>
		<category><![CDATA[voip class]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=337</guid>
		<description><![CDATA[<p>One important step into adopting <a title="Internet calls" href="http://www.voip-sip.org/what-is-voip/">VoIP</a> is to choose a VoIP service, which will allow you to make and receive cheap or free local and international phone calls. It is important to choose the right type of VoIP service. Your needs and the way you will communicate should help you decide which type [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/introduction-to-voice-over-ip/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Linux: how to check OS version installed</title>
		<link>http://www.voip-sip.org/linux-how-to-check-os-version-installed/</link>
		<comments>http://www.voip-sip.org/linux-how-to-check-os-version-installed/#comments</comments>
		<pubDate>Mon, 18 Jun 2012 12:50:50 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[VoIP calls quality]]></category>
		<category><![CDATA[linux]]></category>
		<category><![CDATA[version]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=527</guid>
		<description><![CDATA[<p>Sometimes it’s a pain in the ass to install software on unix based systems without having prior knowledge to the OS/kernel versions. You’d do something on debian that doesn’t work on CentOS, Fedora has yum pre-installed where as RHEL4 comes with up2date but you have to have a key, then there’s always RPM’s but who [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/linux-how-to-check-os-version-installed/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Hey!</title>
		<link>http://www.voip-sip.org/hey/</link>
		<comments>http://www.voip-sip.org/hey/#comments</comments>
		<pubDate>Wed, 13 Jun 2012 13:01:02 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Asterisk]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=524</guid>
		<description><![CDATA[<p>To all people who has an experience with Asterisk and Linux &#8211; quote of the day: &#8220;I have not failed. I&#8217;ve just found 10,000 ways that won&#8217;t work!&#8221; by Thomas Edison. Keep trying guys and you&#8217;ll find how to do it right!</p>]]></description>
		<wfw:commentRss>http://www.voip-sip.org/hey/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>How to configure Internet tel. and SIP settings on Nokia phone (E52)</title>
		<link>http://www.voip-sip.org/how-to-configure-internet-tel-and-sip-settings-on-nokia-phone-e52/</link>
		<comments>http://www.voip-sip.org/how-to-configure-internet-tel-and-sip-settings-on-nokia-phone-e52/#comments</comments>
		<pubDate>Wed, 28 Mar 2012 10:33:46 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Google Voice]]></category>
		<category><![CDATA[Mobile VoIP]]></category>
		<category><![CDATA[Open source]]></category>
		<category><![CDATA[SIP protocol]]></category>
		<category><![CDATA[Symbian]]></category>
		<category><![CDATA[VoIP calls quality]]></category>
		<category><![CDATA[how to configure my Nokia]]></category>
		<category><![CDATA[Internet tel.]]></category>
		<category><![CDATA[nokia]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=475</guid>
		<description><![CDATA[<p>In this article we will talk about Nokia Internet telephony and SIP settings since many people asking me about that. I have check all configuration myself on my old Nokia E52 as well as E71 buddy and everything works fine. Therefore you can use the same setup procedure on any E-series with Symbian. Here we [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/how-to-configure-internet-tel-and-sip-settings-on-nokia-phone-e52/feed/</wfw:commentRss>
		<slash:comments>1</slash:comments>
		</item>
		<item>
		<title>TCP vs UDP &#8211; you must know this</title>
		<link>http://www.voip-sip.org/tcp-vs-udp-you-must-know-this/</link>
		<comments>http://www.voip-sip.org/tcp-vs-udp-you-must-know-this/#comments</comments>
		<pubDate>Mon, 12 Dec 2011 14:09:42 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Prioritization and traffic shaping]]></category>
		<category><![CDATA[SIP protocol]]></category>
		<category><![CDATA[VoIP calls quality]]></category>
		<category><![CDATA[How it works]]></category>
		<category><![CDATA[TCP]]></category>
		<category><![CDATA[UDP]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=463</guid>
		<description><![CDATA[I think TCP is an overused protocol and  I think that UDP is an underused protocol. <p>This is an argument I&#8217;ve been having quite a bit with people lately, so I&#8217;ve decided i&#8217;ll lay out my reasoning here so I don&#8217;t have to type or recite it at people over and over. Understanding how TCP [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/tcp-vs-udp-you-must-know-this/feed/</wfw:commentRss>
		<slash:comments>2</slash:comments>
		</item>
		<item>
		<title>Google voice / Google talk and Asterisk configuration</title>
		<link>http://www.voip-sip.org/google-voice-google-talk-and-asterisk-configuration/</link>
		<comments>http://www.voip-sip.org/google-voice-google-talk-and-asterisk-configuration/#comments</comments>
		<pubDate>Thu, 08 Dec 2011 09:30:52 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Google Voice]]></category>
		<category><![CDATA[Open source]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=390</guid>
		<description><![CDATA[<p>This article describes the configuration process of Asterisk with Google voice.</p> <p>Asterisk communicates with Google Voice and Google Talk using the chan_gtalk Channel Driver and the res_jabber Resource module. Before proceeding, please ensure that both are compiled and part of your installation. Compilation of res_jabber and chan_gtalk for use with Google Talk / Voice are dependant [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/google-voice-google-talk-and-asterisk-configuration/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>VoIP or IP Telephony?</title>
		<link>http://www.voip-sip.org/voip-or-ip-telephony/</link>
		<comments>http://www.voip-sip.org/voip-or-ip-telephony/#comments</comments>
		<pubDate>Tue, 15 Nov 2011 10:27:24 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[SIP protocol]]></category>
		<category><![CDATA[voip vs IP telephony]]></category>
		<category><![CDATA[whats the difference?]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=492</guid>
		<description><![CDATA[<p>What is The Difference Between VoIP and IP Telephony?</p> <p>Most people, including consumers, use the terms VoIP (Voice over IP) and IP Telephony interchangeably, equating one to the other. But what’s the difference between the two?</p> <p>Voice over IP (VoIP) is a subset of IP Telephony. IP Telephony, also commonly called Internet Telephony, is the [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/voip-or-ip-telephony/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>Asterisk 10.0.0-rc1 Now Available!</title>
		<link>http://www.voip-sip.org/asterisk-10-0-0-rc1-now-available/</link>
		<comments>http://www.voip-sip.org/asterisk-10-0-0-rc1-now-available/#comments</comments>
		<pubDate>Mon, 14 Nov 2011 11:48:13 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Open source]]></category>
		<category><![CDATA[SIP protocol]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=363</guid>
		<description><![CDATA[<p><a rel="nofollow" href="http://www.digium.com/">Digium</a> releases <a rel="nofollow" href="http://www.asterisk.org/">Asterisk 10</a>.</p> <p>As you may already know Asterisk is a communications platform that allows developers to create powerful business phone systems and unified communications solutions. Since its introduction 12 years ago, Asterisk has been used, free of charge, in nearly every country of the world to power telephone and other [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/asterisk-10-0-0-rc1-now-available/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>SIP based VoIP behind NAT</title>
		<link>http://www.voip-sip.org/sip-based-voip-behind-nat/</link>
		<comments>http://www.voip-sip.org/sip-based-voip-behind-nat/#comments</comments>
		<pubDate>Thu, 10 Nov 2011 10:39:31 +0000</pubDate>
		<dc:creator>Nik</dc:creator>
				<category><![CDATA[Asterisk]]></category>
		<category><![CDATA[Google Voice]]></category>
		<category><![CDATA[History]]></category>
		<category><![CDATA[Open source]]></category>
		<category><![CDATA[SIP protocol]]></category>
		<category><![CDATA[how to configure asterisk]]></category>
		<category><![CDATA[NAT]]></category>
		<category><![CDATA[rtp.conf]]></category>
		<category><![CDATA[SIP ports]]></category>
		<category><![CDATA[VoIP behind NAT]]></category>
		<category><![CDATA[VoIP via NAT]]></category>

		<guid isPermaLink="false">http://www.voip-sip.org/?p=343</guid>
		<description><![CDATA[<p style="text-align: center;" lang="en"></p> <p>For all the technology behind Voice over IP (VoIP), you&#8217;d expect that it would work on every network, but this unfortunately isn&#8217;t the case. Network Address Translation (NAT) is a common practice used in networks, and it doesn&#8217;t play well with VoIP. Solving this problem requires an understanding of NAT, VoIP [...]]]></description>
		<wfw:commentRss>http://www.voip-sip.org/sip-based-voip-behind-nat/feed/</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
	</channel>
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