<?xml version="1.0" encoding="UTF-8"?>
<rss version="2.0"
	xmlns:content="http://purl.org/rss/1.0/modules/content/"
	xmlns:wfw="http://wellformedweb.org/CommentAPI/"
	xmlns:dc="http://purl.org/dc/elements/1.1/"
	xmlns:atom="http://www.w3.org/2005/Atom"
	xmlns:sy="http://purl.org/rss/1.0/modules/syndication/"
	xmlns:slash="http://purl.org/rss/1.0/modules/slash/"
	>

<channel>
	<title>PhoneCard Focus &#187; IP telephony</title>
	<atom:link href="http://www.phonecardfocus.com/wiki/ip-telephony/feed" rel="self" type="application/rss+xml" />
	<link>http://www.phonecardfocus.com</link>
	<description>Spotlight is on YOUR phone cards!</description>
	<lastBuildDate>Tue, 31 Aug 2010 19:03:18 +0000</lastBuildDate>
	<generator>http://wordpress.org/?v=2.9.2</generator>
	<language>en</language>
	<sy:updatePeriod>hourly</sy:updatePeriod>
	<sy:updateFrequency>1</sy:updateFrequency>
			<item>
		<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>
]]></content:encoded>
			<wfw:commentRss>http://www.phonecardfocus.com/2009/08/model-tracing-anonymous-peertopeer-voip-calls.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<title>A General Overview of Voice over IP</title>
		<link>http://www.phonecardfocus.com/2009/08/general-overview-voice-ip.html</link>
		<comments>http://www.phonecardfocus.com/2009/08/general-overview-voice-ip.html#comments</comments>
		<pubDate>Tue, 25 Aug 2009 10:42:01 +0000</pubDate>
		<dc:creator>admin</dc:creator>
				<category><![CDATA[VoIP]]></category>
		<category><![CDATA[international calls]]></category>
		<category><![CDATA[IP telephony]]></category>
		<category><![CDATA[phone cards]]></category>
		<category><![CDATA[prepaid phone cards]]></category>

		<guid isPermaLink="false">http://www.phonecardfocus.com/?p=81</guid>
		<description><![CDATA[Integration of communication services into the IP network infrastructure, and the Internet especially, is natural course that was started long ago with e-mail, continued with instant messaging and now taken one-step further with integration of standard, classical services like telephony. Voice over IP – the transmission of voice over packet-switched IP networks – is one [...]]]></description>
			<content:encoded><![CDATA[<p><strong>Integration of communication services into the IP network infrastructure, and the Internet especially, is natural course that was started long ago with e-mail, </strong>continued with instant messaging and now taken one-step further with integration of standard, classical services like telephony. Voice over IP – the transmission of voice over packet-switched IP networks – is one of the most important emerging trends in telecommunications. Two factors have an important role in the growing popularity of Voice over IP networks: the cost savings factor, inherent to the migration from standard to Voice over IP networks, and the flexibility factor that allows new services and new applications to be added to standard telephony services (video transmission, conferences, etc).</p>
<p><strong>With 2005 declared as the Voice over IP (VoIP) year and with predictions of fairly large budgets attributed to VoIP projects in the near future,</strong> this technology seems set to replace classic, circuitbased telephony in the near future. Even if they serve the same purpose, VoIP has a very different architecture from classic telephony.<span id="more-81"></span></p>
<p><strong>In VoIP networks voice and signalling are multiplexed and travel as normal data inside LANs, WANs or the Internet </strong>whereas in classical telephony each conversation has a private, physical, circuit and a dedicated infrastructure that serves only for its transmission. VoIP sound is sampled, quantified, encoded with an appropriate codec and streamed over traditional network architectures. It is and it behaves as normal IP data but at the same time has to obey to the rules imposed by classical telephony in terms of quality of service and availability.<strong> Developing a robust architecture that respects all these constraints and is secure is not an easy task, and the fact that many companies have implemented </strong>and tried to impose proprietary architectures has added a factor of uncertainty to the strength of this new technology. In the last period however, major companies and institutions have joined in a common effort to create a basic robust standard for VoIP architectures, and security beneficed from a special emphasis with the creation of such projects as VoIPSA.</p>
<p><strong>As with many new technologies, VoIP introduces new security risks and new opportunities for attack.</strong> Inheriting from both networks and telephony, VoIP is subject to security issues coming from both areas. Classical telephony security issues involving signalling protocol manipulations, phreaking (see [4] for more details) as it was dubbed in the seventies, find their mirror in VoIP specific protocol manipulations.<strong> The main purpose remains the same: fraud. Network security issues on the counterpart are far more complex and offer larger perspectives of attack than traditional phreaking.</strong> From physical layer to faulty applications, all network security items are relevant to VoIP security. In terms of exposure, the transport of voice data over the Internet, a highly insecure and unreliable environment, multiplies the attack surface and will surely lead to more attacks on this technology. Furthermore, the synergies of these two aspects of VoIP emerge to add new security threats such as signalling protocols Denial of Service.</p>
<p><strong><em>Source: VoIP Security &#8211; A layered approach</em></strong></p>
]]></content:encoded>
			<wfw:commentRss>http://www.phonecardfocus.com/2009/08/general-overview-voice-ip.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<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>
]]></content:encoded>
			<wfw:commentRss>http://www.phonecardfocus.com/2009/08/establishing-voip-connections-sip.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
		<item>
		<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>
]]></content:encoded>
			<wfw:commentRss>http://www.phonecardfocus.com/2009/08/voip-telephony-advantages.html/feed</wfw:commentRss>
		<slash:comments>0</slash:comments>
		</item>
	</channel>
</rss>
