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	<title>Comments on: SIP, Interrupted</title>
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	<link>http://mrblog.org/2008/06/05/sip-interrupted/</link>
	<description>Mr Blog.  Very technical, or silly, sometimes absurd.</description>
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		<title>By: johnson789</title>
		<link>http://mrblog.org/2008/06/05/sip-interrupted/comment-page-1/#comment-2219</link>
		<dc:creator>johnson789</dc:creator>
		<pubDate>Sat, 12 Jul 2008 03:49:30 +0000</pubDate>
		<guid isPermaLink="false">http://mrblog.org/?p=428#comment-2219</guid>
		<description>I like Email as an analogy over Web for SIP is that Web is strictly a client/server solitary affair, whereas email crosses domains and connects end-points in a more client-server-client fashion, which is much closer to how VoIP with SIP works.
=====================================
johnson789
The VoIP/TDM Routes Marketplace
&lt;a href=&quot;http://www.minutetraders.com&quot; rel=&quot;nofollow&quot;&gt;
The VoIP/TDM Routes Marketplace&lt;/a&gt;</description>
		<content:encoded><![CDATA[<p>I like Email as an analogy over Web for SIP is that Web is strictly a client/server solitary affair, whereas email crosses domains and connects end-points in a more client-server-client fashion, which is much closer to how VoIP with SIP works.<br />
=====================================<br />
johnson789<br />
The VoIP/TDM Routes Marketplace<br />
<a href="http://www.minutetraders.com" rel="nofollow"><br />
The VoIP/TDM Routes Marketplace</a></p>
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		<title>By: SkykingOH</title>
		<link>http://mrblog.org/2008/06/05/sip-interrupted/comment-page-1/#comment-1836</link>
		<dc:creator>SkykingOH</dc:creator>
		<pubDate>Wed, 18 Jun 2008 03:17:43 +0000</pubDate>
		<guid isPermaLink="false">http://mrblog.org/?p=428#comment-1836</guid>
		<description>There is a quiet revolution in Open Source telephony.  

Asterisk makes it very simple to publish your SIP URI.

ENUM is fully supported so it is very simple to translate your old e.164 address to your sip URI.  Several free ENUM services are emerging.

It&#039;s a drop in the bucket not only from a percentage a traffic basis and even when compared to proprietary Voip solutions, every day a bit more traction is obtained.  

Every Asterisk system I setup has a sip alias for each extension.    With two ENUM enabled SIP servers your dream of complete circuit switched bypass is realized.  From the users perspective they dialed a regular phone number.

All revolutions start with a whisper.</description>
		<content:encoded><![CDATA[<p>There is a quiet revolution in Open Source telephony.  </p>
<p>Asterisk makes it very simple to publish your SIP URI.</p>
<p>ENUM is fully supported so it is very simple to translate your old e.164 address to your sip URI.  Several free ENUM services are emerging.</p>
<p>It&#8217;s a drop in the bucket not only from a percentage a traffic basis and even when compared to proprietary Voip solutions, every day a bit more traction is obtained.  </p>
<p>Every Asterisk system I setup has a sip alias for each extension.    With two ENUM enabled SIP servers your dream of complete circuit switched bypass is realized.  From the users perspective they dialed a regular phone number.</p>
<p>All revolutions start with a whisper.</p>
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		<title>By: MrBlog</title>
		<link>http://mrblog.org/2008/06/05/sip-interrupted/comment-page-1/#comment-1650</link>
		<dc:creator>MrBlog</dc:creator>
		<pubDate>Sat, 07 Jun 2008 23:34:09 +0000</pubDate>
		<guid isPermaLink="false">http://mrblog.org/?p=428#comment-1650</guid>
		<description>Right, Brough. They are both analogies.  I think my point was that P2P is a bit of a red herring. The bigger problem is SIP was never exposed directly to the end users as POP3/SMTP were for email or HTTP was for web. And I&#039;d argue that this has nothing to do with the technology or excuses (or complaints) about the protocol limitations, NAT etc.

With VoIP like Vonage (and all Vonage clones), all that is exposed to the end-user is an analog POTS wire, just like the old phone company exposed.  with Skype, all that is exposed is a closed proprietary client (not a protocol).  That leads us nowhere.

Why I like Email as an analogy over Web for SIP is that Web is strictly a client/server solitary affair, whereas email crosses domains and connects end-points in a more client-server-client fashion, which is much closer to how VoIP with SIP works.

If we take email or the web as the analogy, probably more people have their email or web site hosted somewhere (by a service provider), but there&#039;s nothing stopping them from hosting it themselves as many also do.  That&#039;s because the protocols are known and exposed to the end-users and at the server provider borders. If the VoIP providers exposed the protocol across domains (at their borders), then the same thing could occur with VoIP - but really, whether someone runs their own server or not is orthogonal to the problem.</description>
		<content:encoded><![CDATA[<p>Right, Brough. They are both analogies.  I think my point was that P2P is a bit of a red herring. The bigger problem is SIP was never exposed directly to the end users as POP3/SMTP were for email or HTTP was for web. And I&#8217;d argue that this has nothing to do with the technology or excuses (or complaints) about the protocol limitations, NAT etc.</p>
<p>With VoIP like Vonage (and all Vonage clones), all that is exposed to the end-user is an analog POTS wire, just like the old phone company exposed.  with Skype, all that is exposed is a closed proprietary client (not a protocol).  That leads us nowhere.</p>
<p>Why I like Email as an analogy over Web for SIP is that Web is strictly a client/server solitary affair, whereas email crosses domains and connects end-points in a more client-server-client fashion, which is much closer to how VoIP with SIP works.</p>
<p>If we take email or the web as the analogy, probably more people have their email or web site hosted somewhere (by a service provider), but there&#8217;s nothing stopping them from hosting it themselves as many also do.  That&#8217;s because the protocols are known and exposed to the end-users and at the server provider borders. If the VoIP providers exposed the protocol across domains (at their borders), then the same thing could occur with VoIP &#8211; but really, whether someone runs their own server or not is orthogonal to the problem.</p>
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		<title>By: Brough Turner</title>
		<link>http://mrblog.org/2008/06/05/sip-interrupted/comment-page-1/#comment-1647</link>
		<dc:creator>Brough Turner</dc:creator>
		<pubDate>Sat, 07 Jun 2008 19:43:01 +0000</pubDate>
		<guid isPermaLink="false">http://mrblog.org/?p=428#comment-1647</guid>
		<description>I think we&#039;re almost on the same page...  Your hope was that SIP would be as widely usable as email.  My hope was that SIP would be as widely usable as the World Wide Web.  With both email and the web, individuals can be their own service providers if they wish.  The are more web servers than email servers and it&#039;s a bit easier for individuals to run their own web server than to run their own mail server, but in either case there are zillions of choices versus perhaps a few hundred VoIP service providers.</description>
		<content:encoded><![CDATA[<p>I think we&#8217;re almost on the same page&#8230;  Your hope was that SIP would be as widely usable as email.  My hope was that SIP would be as widely usable as the World Wide Web.  With both email and the web, individuals can be their own service providers if they wish.  The are more web servers than email servers and it&#8217;s a bit easier for individuals to run their own web server than to run their own mail server, but in either case there are zillions of choices versus perhaps a few hundred VoIP service providers.</p>
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