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	<title>Comments on: Skype, The Bandwidth Hog</title>
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	<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/</link>
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		<title>By: StanTheMan</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113083</link>
		<dc:creator><![CDATA[StanTheMan]]></dc:creator>
		<pubDate>Sun, 16 Dec 2007 11:08:40 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113083</guid>
		<description><![CDATA[&lt;p&gt;...So, any news regarding this issue?
How can I prevent being misused as a host for Skype&#039;s data transfer? Is using a different programm the only solution? There must be a way to block the abuse -&gt; Kazaa also doesn&#039;t function on some networks, so there must be a way to block it.
I am sure there are better programms, but most contacts in my contact lists are hardly technical enough to use skype, let alone replace it by another programm.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>&#8230;So, any news regarding this issue?<br />
How can I prevent being misused as a host for Skype&#8217;s data transfer? Is using a different programm the only solution? There must be a way to block the abuse -&gt; Kazaa also doesn&#8217;t function on some networks, so there must be a way to block it.<br />
I am sure there are better programms, but most contacts in my contact lists are hardly technical enough to use skype, let alone replace it by another programm.</p>
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		<title>By: ohmer</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113082</link>
		<dc:creator><![CDATA[ohmer]]></dc:creator>
		<pubDate>Mon, 16 Apr 2007 06:00:05 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113082</guid>
		<description><![CDATA[&lt;p&gt;ive heard news that skype eat a lot of bandwidth though is there any concrete comparison between free voip p2p that can conclude or exaclty compare every product. so that user will now what is the realyy score for skype.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>ive heard news that skype eat a lot of bandwidth though is there any concrete comparison between free voip p2p that can conclude or exaclty compare every product. so that user will now what is the realyy score for skype.</p>
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		<title>By: Gert123456</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113081</link>
		<dc:creator><![CDATA[Gert123456]]></dc:creator>
		<pubDate>Wed, 17 Jan 2007 20:09:45 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113081</guid>
		<description><![CDATA[&lt;p&gt;To the guy who wrote the above post titled &quot;SOLUTION TO SKYPE THE BANDWIDTH HOG&quot;. What a pathetic attempt at advertising your product.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>To the guy who wrote the above post titled &#8220;SOLUTION TO SKYPE THE BANDWIDTH HOG&#8221;. What a pathetic attempt at advertising your product.</p>
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		<title>By: dennis</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113080</link>
		<dc:creator><![CDATA[dennis]]></dc:creator>
		<pubDate>Sat, 07 Oct 2006 00:11:32 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113080</guid>
		<description><![CDATA[&lt;p&gt;SOLUTION TO SKYPE THE BANDWIDTH HOG&lt;/p&gt;

&lt;p&gt;There is a much better solution alternative to Skype and its called Damaka (www.damaka.com).  They advertise that they can do all that Skype can do and even more like H.264 video and desktop sharing strictly peer to peer WITHOUT any servers in the middle and without any SUPERNODES in the middle.&lt;/p&gt;

&lt;p&gt;It seems like it can do IM, email, audio calls, audio conference, H.263 and H.264 video, video conference, multiple lines, music on hold, voice mail, file sharing and now DESKTOP SHARING all end to end encrypted using TLS and sRTP and strictly using peer to peer via SIP.&lt;/p&gt;

&lt;p&gt;Check it out: www.damaka.com&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>SOLUTION TO SKYPE THE BANDWIDTH HOG</p>
<p>There is a much better solution alternative to Skype and its called Damaka (www.damaka.com).  They advertise that they can do all that Skype can do and even more like H.264 video and desktop sharing strictly peer to peer WITHOUT any servers in the middle and without any SUPERNODES in the middle.</p>
<p>It seems like it can do IM, email, audio calls, audio conference, H.263 and H.264 video, video conference, multiple lines, music on hold, voice mail, file sharing and now DESKTOP SHARING all end to end encrypted using TLS and sRTP and strictly using peer to peer via SIP.</p>
<p>Check it out: <a href="http://www.damaka.com" rel="nofollow">http://www.damaka.com</a></p>
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		<title>By: Chris</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113079</link>
		<dc:creator><![CDATA[Chris]]></dc:creator>
		<pubDate>Fri, 01 Sep 2006 08:31:59 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113079</guid>
		<description><![CDATA[&lt;p&gt;All users in my office use skype and behind NAT, but is kind of annoying to see so many https connection to nodes &#8230; sometimes I just killed the connection to the nodes&#8230; I just wondering if the connections are consuming bandwith or not&#8230;&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>All users in my office use skype and behind NAT, but is kind of annoying to see so many https connection to nodes &#8230; sometimes I just killed the connection to the nodes&#8230; I just wondering if the connections are consuming bandwith or not&#8230;</p>
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		<title>By: ReliefSource &#187; Voice Over Something</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113078</link>
		<dc:creator><![CDATA[ReliefSource &#187; Voice Over Something]]></dc:creator>
		<pubDate>Tue, 21 Feb 2006 14:03:43 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113078</guid>
		<description><![CDATA[&lt;p&gt;[...] One of the interesting things about this assessment in Sudan is seeing how many people are using Skype - often against the wishes of agency IT departments, who don&#8217;t like Skype&#8217;s tendency to gobble bandwidth and breach firewalls. Keith Doyle of UNJLC commented to me yesterday that he was adding a new contact every day as more and more people were coming online. Now that more people in the field have access to satellite communications - whether their own or other agencies&#8217; - it makes low-cost communication with friends and colleagues a possibility. [...]&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>[...] One of the interesting things about this assessment in Sudan is seeing how many people are using Skype &#8211; often against the wishes of agency IT departments, who don&#8217;t like Skype&#8217;s tendency to gobble bandwidth and breach firewalls. Keith Doyle of UNJLC commented to me yesterday that he was adding a new contact every day as more and more people were coming online. Now that more people in the field have access to satellite communications &#8211; whether their own or other agencies&#8217; &#8211; it makes low-cost communication with friends and colleagues a possibility. [...]</p>
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		<title>By: Bits and Buzz &#187; P2P for Web 2.0: Brainstorming</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113077</link>
		<dc:creator><![CDATA[Bits and Buzz &#187; P2P for Web 2.0: Brainstorming]]></dc:creator>
		<pubDate>Wed, 18 Jan 2006 16:33:17 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113077</guid>
		<description><![CDATA[&lt;p&gt;[...] Some Skype P2P pitfalls on Om Malik&#8217;s blog. [...]&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>[...] Some Skype P2P pitfalls on Om Malik&#8217;s blog. [...]</p>
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		<title>By: The VoIP Weblog</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113076</link>
		<dc:creator><![CDATA[The VoIP Weblog]]></dc:creator>
		<pubDate>Thu, 12 Jan 2006 18:39:15 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113076</guid>
		<description><![CDATA[&lt;p&gt;&lt;strong&gt;Skype Gripes?&lt;/strong&gt;&lt;/p&gt;

&lt;p&gt;Due to numerous problems with the service and unfortunate marketing choices, Skype has an uncomfortable week in the spotlight. The Skype gripes began last week with Aswath&#039;s scathing rebuke of the new Netgear Skype phone: Netgear router can help...&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p><strong>Skype Gripes?</strong></p>
<p>Due to numerous problems with the service and unfortunate marketing choices, Skype has an uncomfortable week in the spotlight. The Skype gripes began last week with Aswath&#8217;s scathing rebuke of the new Netgear Skype phone: Netgear router can help&#8230;</p>
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		<title>By: chris holland</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113075</link>
		<dc:creator><![CDATA[chris holland]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 21:57:08 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113075</guid>
		<description><![CDATA[&lt;p&gt;moo: you don&#039;t have to have a &lt;em&gt;single&lt;/em&gt; host routing all the traffic. You can spread multiple hosts throughout the planet, and , as part of the SIP handshaking process, have calls routed through whichever host will result in less hops.&lt;/p&gt;

&lt;p&gt;Skype&#039;s approach to relaying traffic isn&#039;t P2P either. In fact, the very concept of &quot;relaying traffic&quot; is incompatible with the concept of Peer to Peer. Relaying traffic is just a reality we have to deal with for some rare really really messed-up networks and their users, where STUN has failed.&lt;/p&gt;

&lt;p&gt;Now, you can do that the sleazy way, and hijack someone&#039;s computer to relay the traffic. Or you can put your own servers in strategic locations throughout the planet to relay traffic for your users. Skype adopted the former approach. Gizmo adopted the latter approach.&lt;/p&gt;

&lt;p&gt;&quot;Supernodes&quot; are just &quot;servers&quot;, the same way certain SIP proxies may relay RTP traffic for you. It&#039;s the exact same thing. Except that Skype&#039;s approach hijacks some poor sap&#039;s computer to turn her/him into a &quot;Supernode&quot;.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>moo: you don&#8217;t have to have a <em>single</em> host routing all the traffic. You can spread multiple hosts throughout the planet, and , as part of the SIP handshaking process, have calls routed through whichever host will result in less hops.</p>
<p>Skype&#8217;s approach to relaying traffic isn&#8217;t P2P either. In fact, the very concept of &#8220;relaying traffic&#8221; is incompatible with the concept of Peer to Peer. Relaying traffic is just a reality we have to deal with for some rare really really messed-up networks and their users, where STUN has failed.</p>
<p>Now, you can do that the sleazy way, and hijack someone&#8217;s computer to relay the traffic. Or you can put your own servers in strategic locations throughout the planet to relay traffic for your users. Skype adopted the former approach. Gizmo adopted the latter approach.</p>
<p>&#8220;Supernodes&#8221; are just &#8220;servers&#8221;, the same way certain SIP proxies may relay RTP traffic for you. It&#8217;s the exact same thing. Except that Skype&#8217;s approach hijacks some poor sap&#8217;s computer to turn her/him into a &#8220;Supernode&#8221;.</p>
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		<title>By: Mull</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113074</link>
		<dc:creator><![CDATA[Mull]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 20:52:45 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113074</guid>
		<description><![CDATA[&lt;p&gt;Dave, trust me, the NSA doesn&#039;t need to host the supernodes, they do quite well getting into any conversation they like.  VoIP, PSTN, whatever.  They have done this for quite a while.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>Dave, trust me, the NSA doesn&#8217;t need to host the supernodes, they do quite well getting into any conversation they like.  VoIP, PSTN, whatever.  They have done this for quite a while.</p>
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		<title>By: chris holland</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113073</link>
		<dc:creator><![CDATA[chris holland]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 20:50:26 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113073</guid>
		<description><![CDATA[&lt;p&gt;Greg Parker: yeah you&#039;ll find that many providers that use SIP still strive to keep their members within their own ecosystems by giving them numbers, and then, selectively &quot;peer&quot; with other providers thru kludgy peering numbers.&lt;/p&gt;

&lt;p&gt;This is all silly. But in the end, practice shows that if you use SIP for signaling, your members can be reached through SIP addresses. It may be simpler or harder to find out what that sip address is, but in the end there&#039;s always a way.&lt;/p&gt;

&lt;p&gt;for example my earthlink SIP account can be reached by my email address. sip:hollandct@earthlink.net or mailto:hollandct@earthlink.net  . Same thing. Since earthlink does not yet have a Mac sip client, i&#039;ve configured SJPhone, and it works like a charm. The same way i can use any POP/SMTP client i want with my earthlink e-mail address.&lt;/p&gt;

&lt;p&gt;If i had my way, all phone numbers would be replaced by SIP addresses. Vendors would come-up with a consensus to extend the vCard specification to support SIP addresses, such as an &quot;X-SIP&quot; field. All handheld devices would be IP and SIP-ready, with a vCard-compatible importing facilities over IR, Bluetooth, or IP.&lt;/p&gt;

&lt;p&gt;In fact, in an ideal world, i&#039;d never exchange a &quot;phone number&quot; with someone. I&#039;d just &quot;beam&quot; them, e-mail them, a vCard, write an easy-to-remember sip-address on a piece of paper.&lt;/p&gt;

&lt;p&gt;in an ideal world, all Domain Name Registrars would offer free SIP forwarding, so it&#039;s super easy for someone to register their own domain name, and use it for a personalized SIP address. All calls coming to my personalized SIP address could be routed to whoever my SIP-provider-du-jour is, for example, if you SIP me at sip:chris@chriswuzhere.com  i would tell the domain registrar for chriswuzhere.com to route all SIP calls to sip:hollandct@earthlink.net  . &lt;/p&gt;

&lt;p&gt;There is no need for existing numbering scheme, or yet another numbering scheme, or another gold rush to some other centralized &quot;directory&quot;.&lt;/p&gt;

&lt;p&gt;DNS is all we need.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>Greg Parker: yeah you&#8217;ll find that many providers that use SIP still strive to keep their members within their own ecosystems by giving them numbers, and then, selectively &#8220;peer&#8221; with other providers thru kludgy peering numbers.</p>
<p>This is all silly. But in the end, practice shows that if you use SIP for signaling, your members can be reached through SIP addresses. It may be simpler or harder to find out what that sip address is, but in the end there&#8217;s always a way.</p>
<p>for example my earthlink SIP account can be reached by my email address. sip:hollandct@earthlink.net or mailto:hollandct@earthlink.net  . Same thing. Since earthlink does not yet have a Mac sip client, i&#8217;ve configured SJPhone, and it works like a charm. The same way i can use any POP/SMTP client i want with my earthlink e-mail address.</p>
<p>If i had my way, all phone numbers would be replaced by SIP addresses. Vendors would come-up with a consensus to extend the vCard specification to support SIP addresses, such as an &#8220;X-SIP&#8221; field. All handheld devices would be IP and SIP-ready, with a vCard-compatible importing facilities over IR, Bluetooth, or IP.</p>
<p>In fact, in an ideal world, i&#8217;d never exchange a &#8220;phone number&#8221; with someone. I&#8217;d just &#8220;beam&#8221; them, e-mail them, a vCard, write an easy-to-remember sip-address on a piece of paper.</p>
<p>in an ideal world, all Domain Name Registrars would offer free SIP forwarding, so it&#8217;s super easy for someone to register their own domain name, and use it for a personalized SIP address. All calls coming to my personalized SIP address could be routed to whoever my SIP-provider-du-jour is, for example, if you SIP me at sip:chris@chriswuzhere.com  i would tell the domain registrar for chriswuzhere.com to route all SIP calls to sip:hollandct@earthlink.net  . </p>
<p>There is no need for existing numbering scheme, or yet another numbering scheme, or another gold rush to some other centralized &#8220;directory&#8221;.</p>
<p>DNS is all we need.</p>
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		<title>By: Bohdanka Inkognito</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113072</link>
		<dc:creator><![CDATA[Bohdanka Inkognito]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 19:29:27 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113072</guid>
		<description><![CDATA[&lt;p&gt;I just have found out, that Skype blocks the port it uses for file transfer, simply no other program can use it...what a PIG! NO more skype for me. I have been using this program called Wavago and it seems to be pretty clean player.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>I just have found out, that Skype blocks the port it uses for file transfer, simply no other program can use it&#8230;what a PIG! NO more skype for me. I have been using this program called Wavago and it seems to be pretty clean player.</p>
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		<title>By: Greg Parker</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113071</link>
		<dc:creator><![CDATA[Greg Parker]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 19:26:46 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113071</guid>
		<description><![CDATA[&lt;p&gt;I have been following Skype for well over 2 years - since it&#039;s very early days. Mainly due to a personal interest in seeing the price of communications go to zero or close enough. I am the inventor of Wavago. Wavago was launched in March of 2005. Wavago was designed from the ground up to be a single tool for communications, information and entertainment services. &lt;/p&gt;

&lt;p&gt;On to the point, SuperNodes and proprietary protocols:
SuperNodes have long been known to be a part of Skype. After all, Skype&#039;s heritage is Kazaa to Joltid to Skype, which all believe that P2P means using intermediaries for the delivery/relay of data/information and the use of those intermediaries at their will. Skype&#039;s P2P idea/methodology may have come from the days of SETI - where any user anywhere in the world could install a screensaver that when engaged allowed SETI to use that computers resources for complex calculations for the benefit of advancing the human race. However, as we all know, this is not true Peer-to-Peer communications, and it appears that Skype/eBay are mainly interested in advancing themselves. Several players in the market do offer direct peering alternatives, including Wavago. But SuperNodes are consuming more than just bandwidth – they also consume computer resources (CPU and Memory) for other users’ communications. Each on there own is significant, and I have talked to many users that cannot understand why their computer is running so slowly with Skype running, and fine when it is not running – answer is a combination of the shear size of Skype (recently referred to as bloat-ware) running somewhere in the 25MB + memory range, it’s use of CPU, and the use of SuperNodes.
Proprietary Protocols: in my many years of experience, the question of using standards or not has presented itself many many times. I am a standards advocate, but just because a particular system conforms to a standard does not mean that they are interoperable for example. If we look at SIP, while the published protocol is a ‘standard’ the implementations may differ, and more to the point the ability for one SIP system to ‘converse’ with another is not guaranteed in any way. In fact, the real issue with communications across systems (providers) lies not only in the protocol but probably more importantly in the directory location services – where are you in this system in relation to who you wish to contact in another system, and how do the disparate systems relay that information? Now SIP has gone a long way in this direction, but in the end it relies on the ability and willingness of the individual system providers to conform and publish this information.&lt;/p&gt;

&lt;p&gt;Wavago’s approach was quite simple – support the existing major protocols transparently (Yahoo, AIM, ICQ, MSN, Google and Skype), support SIP and only if and where required utilize a proprietary protocol. No use of SuperNodes, use direct peering technologies. Include communications, information and entertainment services in one application. Minimize footprint for computer and network resources. Make the contact the center, remove the complexity of multi-protocol communications, and make it easy to use. &lt;/p&gt;

&lt;p&gt;www.Wavago.com&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>I have been following Skype for well over 2 years &#8211; since it&#8217;s very early days. Mainly due to a personal interest in seeing the price of communications go to zero or close enough. I am the inventor of Wavago. Wavago was launched in March of 2005. Wavago was designed from the ground up to be a single tool for communications, information and entertainment services. </p>
<p>On to the point, SuperNodes and proprietary protocols:<br />
SuperNodes have long been known to be a part of Skype. After all, Skype&#8217;s heritage is Kazaa to Joltid to Skype, which all believe that P2P means using intermediaries for the delivery/relay of data/information and the use of those intermediaries at their will. Skype&#8217;s P2P idea/methodology may have come from the days of SETI &#8211; where any user anywhere in the world could install a screensaver that when engaged allowed SETI to use that computers resources for complex calculations for the benefit of advancing the human race. However, as we all know, this is not true Peer-to-Peer communications, and it appears that Skype/eBay are mainly interested in advancing themselves. Several players in the market do offer direct peering alternatives, including Wavago. But SuperNodes are consuming more than just bandwidth – they also consume computer resources (CPU and Memory) for other users’ communications. Each on there own is significant, and I have talked to many users that cannot understand why their computer is running so slowly with Skype running, and fine when it is not running – answer is a combination of the shear size of Skype (recently referred to as bloat-ware) running somewhere in the 25MB + memory range, it’s use of CPU, and the use of SuperNodes.<br />
Proprietary Protocols: in my many years of experience, the question of using standards or not has presented itself many many times. I am a standards advocate, but just because a particular system conforms to a standard does not mean that they are interoperable for example. If we look at SIP, while the published protocol is a ‘standard’ the implementations may differ, and more to the point the ability for one SIP system to ‘converse’ with another is not guaranteed in any way. In fact, the real issue with communications across systems (providers) lies not only in the protocol but probably more importantly in the directory location services – where are you in this system in relation to who you wish to contact in another system, and how do the disparate systems relay that information? Now SIP has gone a long way in this direction, but in the end it relies on the ability and willingness of the individual system providers to conform and publish this information.</p>
<p>Wavago’s approach was quite simple – support the existing major protocols transparently (Yahoo, AIM, ICQ, MSN, Google and Skype), support SIP and only if and where required utilize a proprietary protocol. No use of SuperNodes, use direct peering technologies. Include communications, information and entertainment services in one application. Minimize footprint for computer and network resources. Make the contact the center, remove the complexity of multi-protocol communications, and make it easy to use. </p>
<p><a href="http://www.Wavago.com" rel="nofollow">http://www.Wavago.com</a></p>
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		<title>By: chris holland</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113070</link>
		<dc:creator><![CDATA[chris holland]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 15:32:23 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113070</guid>
		<description><![CDATA[&lt;p&gt;what baffles me most is why people even bother with and use Skype?&lt;/p&gt;

&lt;p&gt;It&#039;s a closed protocol. It&#039;s AOL Mail circa 1989.&lt;/p&gt;

&lt;p&gt;I&#039;ll second Rick&#039;s remark, Gizmo Project ( http://www.gizmoproject.com/ ) or any other SIP-based alternative is a far better way.&lt;/p&gt;

&lt;p&gt;One of the reasons SIP works very well 99.99% of the time for everyone, is because all SIP clients make good use of the STUN protocol which works very well around most NAT issues. I&#039;ve done SIP calls behind two layers to NAT without a glitch, a 192.168. network plugged to a 10.0.0 network plugged to my DSL pipe.&lt;/p&gt;

&lt;p&gt;And in the very rare cases where it&#039;s not possible for two SIP peers to connect, the handshaking process in the SIP protocol provides a framework for the data (RTP) traffic to get relayed by a host between the two peers.&lt;/p&gt;

&lt;p&gt;I believe Gizmo (sipphone.com) has set-up a bunch of dedicated hosts throughout the planet to handle traffic relay between SIP peers who can&#039;t connect to each other.&lt;/p&gt;

&lt;p&gt;None of that silly monkey supernode bullcrap.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>what baffles me most is why people even bother with and use Skype?</p>
<p>It&#8217;s a closed protocol. It&#8217;s AOL Mail circa 1989.</p>
<p>I&#8217;ll second Rick&#8217;s remark, Gizmo Project ( <a href="http://www.gizmoproject.com/" rel="nofollow">http://www.gizmoproject.com/</a> ) or any other SIP-based alternative is a far better way.</p>
<p>One of the reasons SIP works very well 99.99% of the time for everyone, is because all SIP clients make good use of the STUN protocol which works very well around most NAT issues. I&#8217;ve done SIP calls behind two layers to NAT without a glitch, a 192.168. network plugged to a 10.0.0 network plugged to my DSL pipe.</p>
<p>And in the very rare cases where it&#8217;s not possible for two SIP peers to connect, the handshaking process in the SIP protocol provides a framework for the data (RTP) traffic to get relayed by a host between the two peers.</p>
<p>I believe Gizmo (sipphone.com) has set-up a bunch of dedicated hosts throughout the planet to handle traffic relay between SIP peers who can&#8217;t connect to each other.</p>
<p>None of that silly monkey supernode bullcrap.</p>
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	<item>
		<title>By: rick</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113069</link>
		<dc:creator><![CDATA[rick]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 15:13:05 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113069</guid>
		<description><![CDATA[&lt;p&gt;One other thing worth mention.  I&#039;ve had no problem with Gizmo and NAT/firewalls.  I believe this is a major reason Skype began sharing computers.  Now, it&#039;s not an issue, so Skype needs to reevaluate their entire network.&lt;/p&gt;

&lt;p&gt;DTMF works on Gizmo too.  Skype doesn&#039;t most of the time.  Gizmo is SIP too.&lt;/p&gt;

&lt;p&gt;Gizmo is just one example of a better service.  More to come.  Gizmo has PSTN calling, so it&#039;s my favorite.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>One other thing worth mention.  I&#8217;ve had no problem with Gizmo and NAT/firewalls.  I believe this is a major reason Skype began sharing computers.  Now, it&#8217;s not an issue, so Skype needs to reevaluate their entire network.</p>
<p>DTMF works on Gizmo too.  Skype doesn&#8217;t most of the time.  Gizmo is SIP too.</p>
<p>Gizmo is just one example of a better service.  More to come.  Gizmo has PSTN calling, so it&#8217;s my favorite.</p>
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	<item>
		<title>By: rick</title>
		<link>http://gigaom.com/2006/01/10/skype-the-bandwidth-hog/#comment-113068</link>
		<dc:creator><![CDATA[rick]]></dc:creator>
		<pubDate>Wed, 11 Jan 2006 15:08:56 +0000</pubDate>
		<guid isPermaLink="false">http://gigaom.wordpress.com/2006/01/10/skype-the-bandwidth-hog/#comment-113068</guid>
		<description><![CDATA[&lt;p&gt;moo, P2P means many different things.  You&#039;re probably correct in it&#039;s purest form, but it&#039;s not necessary.  An example of a service using their own servers is Gizmo.  Their call quality is excellent.  Calls get connected much better than Skype.  And they don&#039;t abuse people&#039;s computers.&lt;/p&gt;]]></description>
		<content:encoded><![CDATA[<p>moo, P2P means many different things.  You&#8217;re probably correct in it&#8217;s purest form, but it&#8217;s not necessary.  An example of a service using their own servers is Gizmo.  Their call quality is excellent.  Calls get connected much better than Skype.  And they don&#8217;t abuse people&#8217;s computers.</p>
]]></content:encoded>
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