Source: andyabramson.blogs.com

PhoneBoy and to some extent Dan York get the gold stars here.

Both are security experts and have a real expertise in Voice. Dan hosts a regular podcast and Dameon does a wide range of blogging. But at heart both are white hat hackers, and what Dameon did in finding the flaw in a released version of iSkoot so quickly was worth its weight in gold.

This is just one more example of how the Blogosphere helps the startup community, whether asked or not.

It reminds me of the time when an former eBay executive said to some bloggers who were only trying to help "I’m not sure if you’re reporters, developers or consultants so I don’t know how to deal with you" and then repeated the same comment to me in a slightly different way. Basically, shortly after the eBay purchase Skype had a great chance to embrace the blogosphere, but only in the last six months or so has that really happened well, and largely at the direction of their out of house PR agency in NYC and Chaim Hass, who leads it.

Compare that to iSkoot whose team since the moment the first post went up, were engaging with the bloggers and after a first, reply, after some behind the scenes work a few of us to people we both know at iSkoot (as Dameon points out) the real mystery became clear and iSkoot did the right thing.

This is how the blogosphere helps and will continue to do so. Smart companies embrace passion. It’s only insecure executives who fear their help.

Bravo Dameon, Dan and Jim. Bravo iSkoot.

Source: www.voip-news.com

VoIP seems to be everywhere these days. But VoIP dating? Learn about a new, free service at Click for Nick. (The maps thing really is creepy.)

Voice Over IP Weblog is dishing on the alternatives to the now-defunct PrivatePhone. Check it out here.

VoIP Information tells us about the negatives of VoIP service. Negatives? There are negatives? Seriously though, the author misses one big negative: it won’t work in a power outage. VoIP equipment needs electricity to run, unlike traditional phone lines . . . so if you are in an area of the country that loses power due to snow or tornados or other weather conditions, you could be in trouble. Unless you have a generator, that is.

Source: andyabramson.blogs.com

Pal Paul Amery, who was unceremoniously made redundant in the month of December by the grinches at Skype along with a few others in the developer program, has joined the blogosphere.

Paul will provide some very good insight into VoIP and VoIP software development as well as how to build an ecosystem with developers.

I got to know Paul at the Skype Developer Day in Mountain View last fall, and then met up with him and got to speak at Skype House a few weeks later.

Personally I’m touched by his kind words here on his thank you page. While many people in my profession are strictly mercenary, I’m the kind of person who regularly lends a helping hand to others, especially when times turn tough. It’s part of my early upbringing in business, about giving back to the community that gives unto you something that was made apparent to me by Philadelphia Flyers owner Ed Snider when I worked with the hockey team in the 70s and 80s.

In many ways the giving back is not monetary, it’s helping in ways that create far greater growth. Sometimes its just lending an ear, being another set of eyes or actually doing some work gratis to get things on the right direction. Other times its being supportive, even if you have no personal gain. By giving something good, one gets more in return. I’m happy to have helped Paul and look forward to continuing to.

Source: snapvoip.blogspot.com

VoIP IP Telephony @ http://snapvoip.blogspot.com

Atlanta, Georgia (USA) –– Feb 14, 2007. TransNexus, Inc. has made the OSP Toolkit and RAMS open source projects publicly available on SourceForge. The OSP Toolkit is a client side implementation of the OSP peering protocol. The OSP Toolkit, written in C, is a mature open source project begun in 1999 and has been integrated into numerous commercial and open source VoIP products. The RAMS OSP server is a java based OSP server developed for testing and as a reference implementation.

The Open Settlement Protocol (OSP) is an IP Operations and Billing Support Systems (OSS/BSS) protocol defined by the European Telecommunications Standards Institute (ETSI TISPAN), www.etsi.org. OSP is officially known as ETSI Technical Specification 101321 for inter-domain pricing, authorization and usage exchange. OSP is unique because the way it uses PKI services to enable secure peer to peer communication between VoIP networks. “The OSP protocol was developed to enable direct multi-lateral peering among VoIP networks. OSP provides secure inter-domain access control and eliminates costly network bottlenecks”, stated Richard Brennan, Chairman of ETSI TISPAN Next Generation Network Architecture working group.

RAMS is a java based server useful for managing inter-domain VoIP routing, called number translation and Call Detail Record collection. RAMS supports the European Telecommunications Standard Institute’s OSP Peering protocol (ETSI TS 101 321).

The OSP Toolkit and the RAMS OSP test server is available at links provided below. A free version of the TransNexus commercial OSP based peering server is available at www.transnexus.com.

Links;
The OSP Toolkit
RAMS server
www.transnexus.com

Source: snapvoip.blogspot.com

Anglero at Telecom’s Tsunami tells us about OSP from VOIPUSER.ORG, OSP is as old as VoIP but has been used very little. All the occasions that I know, the OSP was used for billing purposes. When I ran a H323 gatekeeper, for servicing some of the big time providers requested me to provide call information via OSP. I had to get a SUN box just to run the OSP code, because it did not run on anything else at that time.

OSP is an Operational Support System (OSS) protocol well suited for managing inter-domain routing, access control and accounting of SIP transactions. OSP uses the communications protocols below to convey messages. The content of an OSP transaction is an HTTP message formatted according to the standard for MIME. Individual components in the message are XML documents and the message may be signed with an S/MIME digital signature.

So VoIP peering among all the Providers and routing of SIP information among users will provide an stable standard among VoIP users. As Anglero says, OSP guarantees that when "Pamela.Anderson@voipuser.org" is calling you, it is actually Pamela Anderson.

Links;
Telecom’s Tsunami: Setting a new standard in VoIP Peering -VoIPUser.org
VoIPUSER.ORG