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	<title>VoIP-Forum</title>
	<link>http://www.voip-forum.com</link>
	<description>About Open Source and Open Standards in IP Communications</description>
	<lastBuildDate>Sun, 07 Mar 2010 11:26:33 +0000</lastBuildDate>
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		<title>SIPit 26 - Why SIP testing is important to Asterisk and to you</title>
		<description>SIPit is the main interoperability event for all things SIP. It's organized by the SIP Forum and creates good feedback to the IETF. Asterisk has been participating in SIPit during many years and in many variants  - videocaps, Marc Blanchet's IPv6 branch and the standard Digium releases. All these tests has lead ...</description>
		<link>http://www.voip-forum.com/asterisk/2010-03/sipit-26-sip-testing-important-asterisk/</link>
			</item>
	<item>
		<title>Robert Sparks on the importance of John Postel&#8217;s robustness principle</title>
		<description>Jon Postel captured an approach to make the Internet a more robust place for applications to live in - and added it to RFC793 (the definition of TCP at the time) :"TCP implementations will follow a general principle of robustness: be conservative in what you do, be liberal in what you accept ...</description>
		<link>http://www.voip-forum.com/opensource/2010-02/robert-sparks-importance-john-postels-robustness-principle-2/</link>
			</item>
	<item>
		<title>Realtime communication - the Open way</title>
		<description>I am working with two very different implementation projects this winter. One is about using Asterisk in large call center environments. Another has a goal of building a SIP network for 15.000 phones in a university.Asterisk for large call centers - full controlThe call center platform is focused on a ...</description>
		<link>http://www.voip-forum.com/opensource/2010-02/realtime-communication-open/</link>
			</item>
	<item>
		<title>Important security advisory for Asterisk :: Dialstring injections</title>
		<description>Hans Petter Selasky alerted the Asterisk developer community about a potential harmful pattern in Asterisk dialplans on February 9th.  His example is as follows:[from_sip]exten =&#62; _X.,1,Dial(SIP/${EXTEN}@testsip)He writes: "And if ${EXTEN} = "000@testsip&#38;SIP/333" what turns out to happen then is similar to SQL injection :-( "He is exactly right. Many VoIP protocols, including IAX2 and ...</description>
		<link>http://www.voip-forum.com/asterisk/2010-02/securityalert-asterisk-dialstring-injections/</link>
			</item>
	<item>
		<title>Asterisk 1.8 LTS wishlist #1: New media negiotiation framework</title>
		<description>On my wishlist for 1.8 long-term-support release the #1 item is a new media negotiation framework for Asterisk.The history - something to learn fromIf we had integrated John Martin's videocaps in time for release of 1.4 we now would have enjoyed four releases of an Asterisk with much better video ...</description>
		<link>http://www.voip-forum.com/asterisk/2010-02/asterisk-18-lts-wishlist-1-media-negiotiation-framework/</link>
			</item>
	<item>
		<title>Why should the VoIP users care about IPv6?</title>
		<description>Let's change everything - and cause no damage for the end users!The current version of the Internet is up for a big overhaul. We have to change the whole infrastructure it runs on, the famous IP protocol. A lot of work needs to be done and it affects everyone that ...</description>
		<link>http://www.voip-forum.com/asterisk/2010-01/voip-users-care-ipv6/</link>
			</item>
	<item>
		<title>Test my RTCP test branch based on Asterisk 1.4!</title>
		<description>I've created a test branch for patches hidden in several Asterisk development branches - all based on Asterisk 1.4  	RTCP improvements from pinefrog-1.4	"Sip show chanstats" cli command	The branch pinequality-* giving you the manager "sipchannel" event to check QoSThis branch is now open for testing and I need feedback. Among ...</description>
		<link>http://www.voip-forum.com/opensource/2010-01/test-rtcp-test-branch-based-asterisk-14/</link>
			</item>
	<item>
		<title>Kamailio 3.0 is released</title>
		<description>Kamailio - the successor to OpenSER - has been released in a new version, 3.0. This version is based on the merge between SER, SIP Express Router, and OpenSER code bases and developer teams. It's a SIP server that you can run as a SIP proxy, a session border controller, a ...</description>
		<link>http://www.voip-forum.com/opensource/2010-01/kamailio-30-released/</link>
			</item>
	<item>
		<title>Measuring voice quality in Asterisk</title>
		<description>During the last week, I've been diving down into the RTCP protocol and Asterisk's implementation of it. What is RTCP? In short, it's the way to understand what's going on with your SIP calls on the network.VoIP relies on the IP network for media transmissionVoice over IP protcols, like SIP, ...</description>
		<link>http://www.voip-forum.com/asterisk/2010-01/measuring-voice-quality-asterisk/</link>
			</item>
	<item>
		<title>Jabber&#8217;s 11 year birthday and todo-list for 2010</title>
		<description>Peter Saint-Andre blogs about Jabber's 11th anniversary and what the focus will be for the XMPP community during 2010. Here's his list:	End-to-end encryption	Finalizing Jingle-based file transfer	Multi-user Jingle for voice conferencing and the like	Distributing chat rooms across servers	Bridging between serverless mode and server mode (very useful in distressed networks)	Reputation systems for XMPP ...</description>
		<link>http://www.voip-forum.com/security/2010-01/jabbers-11-year-birthday-todolist-2010/</link>
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