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	<title>PhoneCard Focus &#187; VoIP phone cards</title>
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	<description>Spotlight is on YOUR phone cards!</description>
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		<title>THE OVERALL MODEL OF TRACING ANONYMOUS PEERTOPEER VOIP CALLS</title>
		<link>http://www.phonecardfocus.com/2009/08/model-tracing-anonymous-peertopeer-voip-calls.html</link>
		<comments>http://www.phonecardfocus.com/2009/08/model-tracing-anonymous-peertopeer-voip-calls.html#comments</comments>
		<pubDate>Mon, 31 Aug 2009 14:00:35 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[calling cards]]></category>
		<category><![CDATA[international calls]]></category>
		<category><![CDATA[internet telephony]]></category>
		<category><![CDATA[IP telephony]]></category>
		<category><![CDATA[phone cards]]></category>
		<category><![CDATA[VoIP calling cards]]></category>
		<category><![CDATA[VoIP phone cards]]></category>
		<category><![CDATA[VoIP telephony]]></category>

		<guid isPermaLink="false">http://www.phonecardfocus.com/?p=83</guid>
		<description><![CDATA[Given any two di®erent Skype peers A and B, we are interested in determining if A is talking (or has talked) to B via Skype peer-to-peer VoIP. As shown in Figure 1, both Skype peers A and B have outgoing and incoming VoIP °ows to and from the Internet cloud. The Skype peers could be [...]]]></description>
			<content:encoded><![CDATA[<p><strong>Given any two di®erent Skype peers A and B, we are interested in determining if A is talking (or has talked) to B via Skype peer-to-peer VoIP.</strong> As shown in Figure 1, both Skype peers A and B have outgoing and incoming VoIP °ows to and from the Internet cloud. The Skype peers could be behind ¯rewall and NAT, and peer A and/or B could be connected to some low latency anonymizing network. Here we view the Internet cloud and any low latency anonymizing network as a black box, and we are interested only in the Skypy °ows that enter or exit the black box.</p>
<p><strong>We assume that (1) we can monitor the Skype °ow from the black box to the Skype peer; (2) we can perturb the timing of the Skype °ow from the Skype peer to the black box.<span id="more-83"></span></strong></p>
<p>Here we do not intend to track all the peer-to-peer VoIP calls from anyone to anyone, nor do we assume the global monitoring and intercepting capability. <strong>Instead we focus on ¯nding out if some parties in which we are interested have communicated via peer-to-peer VoIP calls anonymously, and we only need the capability to monitor and intercept IP °ows to and from those interested parties. </strong>This model is consistent with our understanding of the common practice of lawful electronic surveillance by the law enforcement agencies. Because the Skype VoIP °ows are encrypted from end to end, no correlation could be found from the °ow content.</p>
<p><strong>Given that the Skype VoIP °ow could pass some intermediate Skype peers and some low latency anonymizing network, there is no correlation from the VoIP °ow headers.</strong> Among all the characteristics of the VoIP °ows, the inter-packet timing characteristics are likely to be preserved across intermediate Skype peers and low latency anonymizing network. This invariant property of VoIP °ows forms the very foundation for tracking anonymous, peer-to-peer VoIP calls on the Internet.</p>
<p><em><strong>Source: gmu.edu</strong></em></p>
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		<item>
		<title>Establishing VoIP Connections with SIP</title>
		<link>http://www.phonecardfocus.com/2009/08/establishing-voip-connections-sip.html</link>
		<comments>http://www.phonecardfocus.com/2009/08/establishing-voip-connections-sip.html#comments</comments>
		<pubDate>Tue, 11 Aug 2009 08:50:05 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[calling cards]]></category>
		<category><![CDATA[international calls]]></category>
		<category><![CDATA[internet telephony]]></category>
		<category><![CDATA[IP telephony]]></category>
		<category><![CDATA[phone cards]]></category>
		<category><![CDATA[VoIP calling cards]]></category>
		<category><![CDATA[VoIP phone cards]]></category>
		<category><![CDATA[VoIP telephony]]></category>

		<guid isPermaLink="false">http://www.phonecardfocus.com/?p=76</guid>
		<description><![CDATA[Many VoIP networks use the IETF’s signaling protocol, SIP, to handle the setup and tear down of multimedia sessions between endpoints. This lightweight, text-based signaling protocol is transported over either Transmission Control Protocol (TCP) or UDP. SIP uses invitations to create Session Description Protocol (SDP) messages to carry out capability exchange and to setup call [...]]]></description>
			<content:encoded><![CDATA[<p><strong>Many VoIP networks use the IETF’s signaling protocol, SIP, to handle the setup and tear down of multimedia sessions between endpoints</strong>. This lightweight, text-based signaling protocol is transported over either Transmission Control Protocol (TCP) or UDP. SIP uses invitations to create Session Description Protocol (SDP) messages to carry out capability exchange and to setup call control channel use. These invitations allow participants to agree on a set of compatible media types.</p>
<p><strong>The powerful SIP client-server application supports user mobility with two operating modes: proxy and redirect.</strong> In proxy mode (shown in Figure 3), SIP clients send requests to the proxy server. The proxy server either handles the requests or forwards them to other SIP servers. Proxy servers can insulate and hide SIP users by proxying the signaling messages. To the other users on the VoIP network, the signaling invitations look as if they are coming from the proxy SIP server.</p>
<p><em><strong>Source: Juniper Networks, Inc. White Paper</strong></em></p>
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		<title>VoIP telephony advantages</title>
		<link>http://www.phonecardfocus.com/2009/08/voip-telephony-advantages.html</link>
		<comments>http://www.phonecardfocus.com/2009/08/voip-telephony-advantages.html#comments</comments>
		<pubDate>Wed, 05 Aug 2009 16:15:43 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[calling cards]]></category>
		<category><![CDATA[international calls]]></category>
		<category><![CDATA[internet telephony]]></category>
		<category><![CDATA[IP telephony]]></category>
		<category><![CDATA[phone cards]]></category>
		<category><![CDATA[VoIP phone cards]]></category>

		<guid isPermaLink="false">http://www.phonecardfocus.com/?p=72</guid>
		<description><![CDATA[1. Cost
The feature of VOIP that has attracted the most attention is its cost-saving potential. By moving away from the public switched telephone networks, long distance phone calls become very inexpensive. Instead of being processed across conventional commercial telecommunications line configurations, voice traffic travels on the Internet or over private data network lines.
 VOIP is [...]]]></description>
			<content:encoded><![CDATA[<p><strong>1. Cost</strong></p>
<p>The feature of VOIP that has attracted the most attention is its cost-saving potential. By moving away from the public switched telephone networks, long distance phone calls become very inexpensive. <strong>Instead of being processed across conventional commercial telecommunications line configurations</strong>, voice traffic travels on the Internet or over private data network lines.</p>
<p><strong> VOIP is also cost effective because all of an organization’s electronic traffic (phone and data) is condensed onto one physical network, bypassing the need for separate PBX tie lines.</strong> Although there is a significant initial startup cost to such an enterprise, significant net savings can result from managing only one network and not needing to sustain a legacy telephony system in an increasingly digital/data centered world. Also, the network administrator’s burden may be lessened as they can now focus on a single network. <span id="more-72"></span>There is no longer a need for several teams to manage a data network and another to mange a voice network. The simplicity of VOIP systems is attractive, one organization / one network; but as we shall see, the integration of security measures into this architecture is very complex.</p>
<p><strong>2. Speed and Quality</strong></p>
<p>In theory, <strong>VOIP can provide reduced bandwidth use and quality superior to its predecessor, the conventional PSTN.</strong> That is, the use of high bandwidth media common to data communications, combined with the high quality of digitized voice, make VOIP a flexible alternative for speech transmission. In practice, however, the situation is more complicated.</p>
<p>Routing all of an organization’s traffic over a single network causes congestion and sending this traffic over the Internet can cause a significant delay in the delivery of speech. Also, bandwidth usage is related to digitization of voice by codecs, circuits or software processes that code and decode data for transmission. That is, producing greater bandwidth savings may slow down encoding and transmission processes.<strong> Speed and voice quality improvements are being made as VOIP networks and phones are deployed in greater numbers,</strong> and many organizations that have recently switched to a VOIP scheme have noticed no significant degradation in speed or quality.</p>
<p><em><strong>&lt;-to be continued-&gt;</strong></em></p>
<p><em>(Source: National Institute of Standards and Technology)</em></p>
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