SIP

Pingtel

Doug Mohney — Thu, 13 Nov 2008 09:34:08 -0500

Founded: 2004, sort-of (see below)

Status: Suffering a death of a thousand cuts

What does it do: Scalable SIP-based open source IP PBX software

Once upon a time, Pingtel made Java-based desktop IP phones. Wisely, the company decided to get out of hardware and into software and services, moving its SIPxchange IP-PBX platform into open source. The business model was going to be just like how RedHat operated Linux subscriptions.

It was a very Fierce move. SIPxchange won wide acclaiming as being easier to work with in an all-SIP environment, and for being more scalable than Asterisk. Tthe company also picked up Amazon.com as its marquee customer.

However, the IP-PBX open source market didn't turn into the cash cow initially predicted. Sources indicated Pingtel began seeking a buyer in early 2007, but it didn't manage to find one until Bluesocket snatched them up in July 2007.

Bluesocket said the company would fit into its wireless mobility and security solutions, but it wasn't that great of a fit. Nortel came along in mid-August 2008 and paid some amount of cash to Bluesocket to pick up "key assets" and Pingtel employees.

Taken a look at Nortel's financials lately?

TruPhone, Top VoIP Company 2008: FierceVoIP, Fierce 15

Tue, 04 Nov 2008 14:49:08 -0500

Truphone

Where it's based: London, England
When it was founded: 1999
Website: www.truphone.com

Why it's Fierce: Truphone decided to power its way around the me-too mobile VoIP pack by buying roaming operator SIM4travel. While it is still in the me-too free client/lowest-rate voice game, it hasn't been afraid to fight established cellular operators in court to get the settlements it wants.

On the technology side, Truphone has developed clients for a variety of phones, including the BlackBerry and iPhone; Truphone had the first Internet telephony app for the iPhone on the iStore, er, Apple App Store. The company has a carrier-grade, global telephony operator infrastructure on the backend to handle up to 40 million calls worldwide that includes a global network of SIP gateways, a Nokia Siemens Networks mobile network Home Location Register (HLR) and a GSM identifier.

Sipera Systems, Top VoIP Company 2008: FierceVoIP, Fierce 15

Tue, 04 Nov 2008 14:44:55 -0500

Sipera Systems

Where it's based: Richardson, Texas
When it was founded: 2003
Website: www.sipera.com

Why it's Fierce: Proactive VoIP security testing and vulnerability assessment is something Fierce to our professionally paranoid hearts, so Sipera would win consideration simply for its in-house VIPER lab. However, Sipera has moved beyond simply flogging the latest VoIP threats to incorporate unified communications into its assessments and security appliances. Its latest product release offers advanced security for SIP trunking, better integration into SIP trunk provider services and adding additional SBC functionality. We expect to see more out of Sipera in the months to come.

SimpleSignal, Top VoIP Company 2008: FierceVoIP, Fierce 15

Tue, 04 Nov 2008 14:42:25 -0500

SimpleSignal

Where it's based: Dana Point, Calif.
When it was founded: 2004
Website: www.simplesignal.com

Why it's Fierce: SimpleSignal keeps rolling out the Voice 2.0 mash-up apps on top of its hosting business, leveraging its BroadSoft expertise and layering in more functionality. Gotta love being able to manage your calendar items in Microsoft Outlook through voice commands using their hosted Exchange solution. There's the (almost usual) fare of being able to listen to your email via the phone and having your voice mail transcribed to text for quick email reading on your smart phone. You can also get more SIP trunking bandwidth via SimpleSignal web page within 60 seconds.

Customers love them. We can't wait to see the next set of apps they roll out.

Pactolus, Top VoIP Company 2008: FierceVoIP, Fierce 15

Tue, 04 Nov 2008 14:39:36 -0500

Pactolus

Where it's based: Marlborough, Massachusetts
When it was founded: 1999
Website: www.pactolus.com

Why it's Fierce: Pactolus doesn't grab a lot of headlines, but they got on the open-source and SIP bandwagon way early, founding SIPdev.org in the process. Today, the company's SIP/IMS-based apps process 3 to 4 billion minutes of usage monthly in over 140 service provider networks worldwide.

If you want a service creation environment for SIP, there's RapidFLEX running on industry-standard Linux or Solaris hardware with a supporting XML-based language, XTML (eXtensible Telephony Markup Language). Don't want to build apps from scratch? There's a portfolio of pre-built SIPWare customizable turnkey services to tap into, including residential/business Class 5 VoIP, large operator-assisted event and reservation-less audio conferencing, prepaid/post paid services and voice messaging.

Fonolo, Top VoIP Company 2008: FierceVoIP, Fierce 15

Mon, 03 Nov 2008 16:58:07 -0500

Fonolo

Where it's based: Toronto, Canada
When it was founded: 2007
Website: www.fonolo.com

Why it's Fierce: We're going to stick our necks out on this one; Fonolo is currently in a closed beta, but it seems to have the right combination of founders, open technology, and "Why didn't I think of that?" features.

Founders: Shai Berger & Jason Bigue. Shai and Jason were previously at Keynote leading up the team for streaming performance measurement. The two built Streamcheck leading to an acquisition by Keynote. So they understand performance, complex code, and building a successful company.

Open technology: Fonolo supports SIP - great for making it a plug-and-play option for other SIP services to incoporoate - and has released an open API for its Deep Dialing service. And they announced the API at the ClueCon open source telephony developer conference.

"Why didn't I think of that?": Deep Dialing allows people to skip through IVR phone menus (also known as "Phone hell") of large companies and navigate to the information/place they require. Find the company on Fonolo's website, click on the appropriate link, and then get a call back on your phone.

There's also an intelligent call history service - just the thing when dealing with crappy customer service - to automatic track calls made to a given company complete with recordings and user notes, regardless of the phone used.

Is there enough here for a sustainable business model? Good question but regardless, Fonolo has broken out of the drab set of me-too VoIP apps over the past year.

SPOTLIGHT: Fring integrates mobile VoIP client with carrier

Doug Mohney — Wed, 22 Oct 2008 11:06:27 -0400

Some mobile VoIP players actually have a business plan beyond "If we build it, they will come." It appears fring is one of them, as the company has established a relationship to put its client on some of Telekom Austria mobilkom's phones.

In the arrangement, fring will create a custom version of its VoIP client to support Mobilkom's A1 service over IP. The first version will be on the Nokia S60, and the mobilkom client, like any standard fring client, will be able to use any Internet connection. Since mobilkom supports SIP natively, users will get both voice and IM in one package along with the social media aspects. The client is expected to be available in mid-November.

For more:
- Ars Technica blog on fring working with Austria's mobilkom.
- Telegeography post on the fring/mobilkom partnership

Sutus raises cash, debt, hires new CEO

Doug Mohney — Tue, 21 Oct 2008 16:44:10 -0400

Vancouver-based Sutus has secured $2 million in venture funding, plus another $2.5 million in debt financing. It has also added a new CEO.

The $2 million in cash came from BDC Venture Capital, and the $2.5 million in debt financing is being provided by MMV Financial. In addition Sutus is bringing telecommunications veteran Mike Donnell on as CEO; Donnell previously lead New Global Telecom (NGT) as its CEO and has also clocked time as CEO of Zi Corporation and Cellular One.

Founded in 2002, Sutus has over $15.5 million invested into its all-in-one Business Central box targeted at the small office market. The company launched product in late 2007.

Sutus' Business Central was specifically developed for businesses with up to 25 employees and includes a phone system that supports both PSTN and VoIP connections, a file server, email server, router, firewall, wireless access point, VPN remote access server, and automated backups; it can also serve as a gateway for hosted servers and includes extensive remote management and support features for the VAR/resellers that install it. This reporter doesn't know if a kitchen sink option is available.

For more:
- Sutus announces new CEO, funding on MarketWatch. Release.

SIP Forum: Championing Interoperability and moving beyond SIP Trunking

Tue, 21 Oct 2008 16:26:25 -0400

By Marc Robins
Managing Director, SIP Forum

One of the most common themes -- and challenges -- being discussed in today's IP communications marketplace is without doubt Interoperability. Compared to the old guard (the PSTN and legacy TDM technology), the IP communications industry has at times operated like the Wild West, with vendors creating "envelope-stretching" IP-based products that unfortunately do not play well with others.

At the Forum, we believe that there is a growing realization within the IP communications industry that achieving interoperability among disparate products and services is key to sustained and long-term growth.

Say what you will about legacy TDM technology, but one great thing about the ole' PSTN was the requirements by regulatory agencies and stringent controls placed by network operators on what went into their networks or attached to them, and the standards that various types of equipment were required to be strictly compliant with. As a result things just worked.

That monolithic oversight is of course a double-edged sword, as while it ensured interoperability it also limited innovation and provided for a limited set of features and capabilities due to the need to serve the lowest common denominator.

The SIP Forum was originally founded 8 years ago as a non-profit IP communications industry association to promote the adoption of SIP by the companies developing products and services. Fast forward to today and now virtually every product is SIP-based. Since the original "battle" has been won, the SIP Forum is now onto "Job #2" - solving interoperability problems to allow different SIP-based networks and various IP communications products to seamlessly work together. This is a critical need because SIP is a complex standard, and there are a number of different options available to developers of equipment and services, creating a hodge-podge of development choices.

In this regard, the SIP Forum is responsible for producing new industry technical recommendations that provide a specific set of rules, guidelines and best practices that determine which SIP RFC and deployment option to use when and where. One example is the SIPconnect Technical Recommendation, which provides rules and guidelines for accomplishing SIP trunking, or direct IP peering, between SIP-enabled IP-PBXs and SIP-enabled VoIP service providers. SIPconnect is a voluntary but peer-reviewed process

The SIPconnect Compliant Program is an associated certification program designed to certify equipment vendors and service providers as SIPconnect compliant and thus offer the highest degree of confidence that they will be able to accomplish trouble-free SIP trunking.

This is one area where the SIP Forum's current industry relevance rings loud and clear. With respect to SIPconnect and our associated compliance program, we have effectively directed vendors and service providers to make important changes to the way they implement SIP so that their offerings strictly adhere to the rules we have established. Without naming names, a number of leading companies have made such changes so that they can be deemed SIPconnect Compliant.

Other notable activities of the Forum include a variety of educational workshops and trade show-related activities that highlight interoperability issues and cover successful applications and deployments, the production of bi-annual SIPit test events, and providing a "meeting place" for developers of commercial SIP-based products and services.

Beyond SIP Trunking

Areas that the Forum is working on other than SIP trunking - areas that cry out for industry consensus and technical interoperability -- are Fax over IP (FoIP), user agents/end-point devices, security and unified communications (OC).

FoIP, for example, is ripe for an industry-approved technical recommendation that spells out the technical method to ensure that Fax works reliably over an IP network. Many pundits predicted that fax was dead several years ago, so it might surprise you to know that due to a host of federal and state regulations (in the U.S. at least) that require that documents be transmitted via fax, including HIPPA (Health-Care related privacy regs and Sarbanes Oxley (public company accounting and reporting regulations), fax is alive and well.

As service provider networks migrate en masse to an IP infrastructure, and enterprises follow suit, fax has suffered. With FoIP, there is an unacceptably high degree of transmission failure (around 2 out of every 10 pages fail to transmit.) What this means is that out of ten 1-page fax transmission, 2 attempts will fail. But it gets much worse when you are trying to send 10 ten-page faxes - the likelihood that each fax transmission will experience issues is extremely high. To help jump-start this work, the SIP Forum will be hosting a FoIP Interoperability workshop in San Francisco during the upcoming VoiceCon show in November.

In addition, the Forum has restarted our Phone Technology working group (now called the UA Configuration Task Group) to develop an industry-approved SIP end-pointdevice profile that will help ensure consistent, reliable interworking across various vendors systems.

Looking down the road

In the near-term, the Forum will continue to expand our efforts around resolving issues relating to SIP trunking. Longer term, there are obviously other areas where interoperability is a critical issue. The SIP Forum will continue to direct our resources in interoperability testing, profile definition, and work with the IETF and other standards bodies to ensure SIP is the multi-vendor, robust protocol of choice for all IP communications products and services.

Marc Robins currently serves as president of SIP Forum LLC (the operating U.S. subsidiary of the SIP Forum) and as the consulting managing director of the SIP Forum. He is also founder and Chief Technology Evangelism Officer of Robins Consulting Group (RCG), an IP Communications industry consultancy, and partners with leading industry analyst Jon Arnold in the industry information portal IP Communications Insights.

SPOTLIGHT: NGN joins IMS Forum

Doug Mohney — Tue, 14 Oct 2008 16:07:07 -0400

Merging isn't just for commercial ventures. The IMS Forum and the NGN forum have merged to form the IMS/NGN Forum.

By combining IMS, SIP, UC and service deliver platforms (SDB), the new organization is said to unite developers, service providers, integrators and solutions providers to further NGN and IMS deployment. The new entity will continue IMS interoperability and IMS Plugfest events with focus on IP billing, OSS, security, reliability, SI, Diameter, UC, SDP and IP video over all types of broadband networks.

More than 2000 executives and technical and marketing professionals will be members of the new entity, which will include equipment vendors, solution providers, integrators, service providers and government agencies. The new entity also announced the formation of a joint Venture Capital study group consisting of its members and more than 20 venture capitalists, private equity groups and financial institutions.

For more:
- IMS Forum release on the IMS/NGN Forum merger

Microsoft announces R2 of OCS 2007

Doug Mohney — Tue, 14 Oct 2008 14:52:01 -0400

Microsoft Office Communications Server 2007 Release 2 (R2) has been announced and is currently in "private beta" testing. Customers will be able to buy the new release starting in February 2009.

OCS 2007 R2 (or maybe just R2) adds dial-in audio conferencing support, desktop sharing with audio communications and collaboration through a web-based interface, and persistent group chat. The persistent group chat feature provides a list of all available chat rooms and topics along with periodic archiving of discussions in an XML format; tools are available to search the entire history of discussion on a given topic, and filters and alerts are able to notify someone of new posts or topics on a particular topic.

Other enhancements include an attendant console that allows for delegation of call management with configuration of workflows to route calls, SIP trunking, a response group application to manage incoming calls based upon user-configured rules, and extension of Microsoft Office Communication Mobile functionality to Nokia S40, Motorola RAZR, Blackberry and Windows Mobile platforms. Developers get more APIs and Visual Studio integration.

For more:
- Microsoft release on the coming of OCS 2007 R2

Gizmo5 adds free conference calls

Doug Mohney — Sun, 12 Oct 2008 20:59:18 -0400

SIP/VoIP provider Gizmo5, home of SIPphone, has added free conference calling for its customers.

The company sent an email announcement about the new service over the weekend, reports Om Malik. Om is a bit skeptical of the service, having run into voice quality problems with Gizmo5 in the past, but he says he was able to get a room number with ease and a conference call number "surprisingly smoothly." Each conference room can handle up to 10 people, and it's easy enough to "tape" conversations for archival purposes.

Observers of entrepreneur/mad genius Michael Robinson might interpret this as an opportunistic move to snap users who liked Skype's free, but now dropped, conferencing service.

For more:
- Om blogs about Gizmo5 new conference call service at Web Worker Daily

Raketu adds SIP service

Pete Wylie — Wed, 01 Oct 2008 15:07:40 -0400

Raketu will now offer a client called RakSIP that will enable SIP-based calls through the company's RakOut VoIP service.

Users need only connect their third-party SIP client (mobile or desktop), IP phone or SIP hardware to RakOut to make outbound calls at low prices. RakSIP is compatible with the built-in SIP capabilities in Nokia smartphones and Blackberry devices, and it works with most iPhone and WinMobile SIP clients, according to Raketu's release.

Current users of the RakIn service can begin using RakSIP immediately, and new users can sign up easily for only 99 cents per month. The company has a laundry list of services for use with TVs, IM, mobil devices, businesses, Google widgets, and web phone clients. The addition of a SIP mode to the company's peer-to-peer service solution gives them one up on Skype. Is a channel driver for Asterisk soon to follow?

For more:
- see the RakSIP press release here

Voixtas to offer snom SIP phones

Pete Wylie — Thu, 25 Sep 2008 10:21:04 -0400

Voxitas and snom announced Wednesday that they will form a partnership that will see snom's SIP-based phones integrated into Voxitas's business-class VoIP services. The deal comes after Voxitas extensively tested snom's 3xx series VoIP phones and m3 IP DECT mobile phones and found them to be interoperable with all Voxitas's IP PBX customers.

Voxitas said in a release that the partnership would enhance the value proposition it offers its small to medium business customers with its SIP True Voice service.

In addition to wide-ranging support for various security protocols, snom's phones support several audio devices simultaneously, giving customers flexibility and customability, as well as knowledge that their secure communications are protected.

snom's phones will be available through Voxitas's reseller network, as the German firm continues to add partners in the SMB VoIP space. Earlier this month, snom announced a partnership with Fonality.

For more:
- see the joint snom and Voxitas press release here

Related Articles
Pushing SIP trunking
Fonality and snom partner

Polycom announces new interoperability for SpectraLink phones

Pete Wylie — Mon, 22 Sep 2008 09:47:42 -0400

Polycom announced it has extended the interoperability capabilities of its SpectraLink 8000 series of phones to meet open VoIP standards in addition to the proprietary protocols it already supports. SpectraLink now has full interoperability with SIP-based systems, and Polycom chose a fitting event, Interop New York, to launch the product.

The SpectraLink WiFi handsets can now operate as extensions of the VoIP call server and have full business capabilities, such as caller ID, conferencing and messaging, among others.

Ben Guderian, vice president of product management for Polycom's SpectraLink, said the offering is perfect for enterprises with large campuses that benefit from cost-effective WiFi communications.

"There is a lot of effort going into mobile UC right now with emphasis on cellular functionality, but what about those enterprises whose employees are not at desks, but rather working on a warehouse floor or hospital ward?" Guderian said. "Our phones allow for decreased use of paging and more expensive cellular connections, plus they are much more durable than cellular devices." Guderian literally hammered that point home by repeatedly banging the handset on the table where we sat, with no damage to the phone.

Polycom representatives said they are marketing the new offering to hospitality, health care, manufacturing and retail stores primarily.

For more:
- see the Polycom press release announcing SpectraLink 8000

SPOTLIGHT: iPod Touch finally VoIP-able

Pete Wylie — Fri, 12 Sep 2008 13:04:45 -0400

Clever hardware buffs have found a pathway to enable VoIP calls on the newly released iPod Touch and Classic. Apple enabled the headphone jack to accept external microphones, with the intention of letting clients record voice memos on the devices. But with the right application, users could manipulate the external microphone capability to enable VoIP calls over WiFi through clients like Jajah and Skype.

Apple has yet to clarlify on its stance on VoIP calls over WiFi; the company expressly forbids VoIP calls using either the EDGE of 3G networks.

For more:
- see the blog post on macrumors.com

Pushing SIP trunking

Doug Mohney — Wed, 10 Sep 2008 16:15:12 -0400

A range of IP communications vendors and service providers are continuing an uphill battle to promote SIP trunking. The latest efforts include workshops and a forthcoming draft of an updated interoperability standard.

Ingate is leading the charge and has organized a free three-day workshop called "SIP Trunking: Everything You Need to Know" at the Internet Telephony Expo in Los Angeles next week. Partners in the effort include Avaya, BandTel, BroadSoft, Cbeyond, Digium and Voxitas. The three day program includes a track introducing the concept of SIP trunking, several case studies including one featuring support for "American Idol," and a day long "SIP Trunking Boot Camp" for interested parties.

"There's a tremendous amount of misunderstanding as to what [SIP trunking] exactly is," said Joel Maloff, senior vice president of marketing for BandTel. "SIP trunking as the name of a service isn't all that sexy. People are not familiar with it, they don't really know about the benefits... the major effort needed on our part is educational."

Later in September, the SIP Forum will sit down with Cable Labs and representatives of cable companies to hammer out a draft of SIPConnect 1.1 - the next standard for implementing trouble-free SIP trunking between IP communications software and hardware manufacturers and service providers.

SIPConnect 1.0, the existing standard, has helped a number of service providers identify interoperability problems, said SIP Forum managing director Marc Robins."There's a comprehensive survey that companies must fill out, they need to demonstrate that they're checking the right boxes. They have to stand up an actual demonstration, show that they're actually working with another company to do SIP trunking."

Robins expects a draft of SIPconnect 1.1 to be hashed out in the Cable Labs meeting and released in about "a month or so."

For more:
- Agenda for the three day SIP trunking seminar. Link.

Verizon Business adds new features to wholesale VoIP

Pete Wylie — Tue, 09 Sep 2008 14:16:03 -0400

Verizon Business has adding the SIP Gateway Outbound Service to its VoIP wholesale product list. The new addition uses virtual rather than physical trunking to achieve higher transfer and processing speeds, which in turn delivers more calling options, faster service initiation and faster long-distance call processing, says Verizon.

Gary Elliott, senior project manager for the SIP product, said it completed beta testing around a month ago, and customers have already started using the new system.

"SIP Gateway Outbound Service provides scalable and cost-effective solutions to customers moving to SIP or VoIP offerings from TDM services," Elliott said. "The flexibility for the customer in routing traffic with Gateway Outbound is a big positive that makes it competitive in the market."

SIP Gateway Outbound Service allows calls to originate from URI and URL Internet formats, as well as the traditional numbering format. It also offers compatibility with Remote Party ID signaling configuration, in addition to the P-asserted ID signaling configuration, which previous iterations of the SIP client have supported.

Elliott said other companies charge their customers extra to have both configurations accepted, or only support one.

"It's just an example of how we try to do everything to make it easier for customers to get the best possible service for the lowest price," Elliott said. "I think Verizon's offerings just have more bang for your buck than other competitors in the space, and the SIP Gateway Outbound Service is another step in making our product line even more robust."

For more:
- see the Verizon press release about its outbound SIP gateway

Related Articles
Verizon gets props for wholesale VoIP
Verizon Business VoIP, UC Expand Across Europe

Voxeo Reaches into Asia

Doug Mohney — Tue, 19 Aug 2008 15:40:15 -0400

Voxeo Corporation has acquired Beijing-based Micromethod Technologies. Did they also score some Olympic tickets in the deal?

Voxeo, a leading provider of IVR and VoIP platforms and services, says the acquisition of Micromethod bolsters its line of SIP-focused products and services and serves as the foundation of Voxeo's expansion into the Asia Pacific market. As a part of the deal, all of Micromethod's employees have joined Voxeo and will work for Voxeo's newly launched Asia Pacific subsidiary. Micromethod CEO Kevin Lu has been named GM of Voxeo Asia-Pacific, while Micromethod CTO Wei Chen has been named Chief Architect of Voxeo worldwide.

Financial terms of the deal were, alas, not disclosed.

Customers currently using Micromethod technology include Tellme and GMarket, as well as "leading" video and wireless communications vendors. Micromethod products include an end-to-end SIP development platform, the SIPoint solution for SIP user collaboration, connectivity and messaging, and a SIP firewall/NAT/VPN solution.

Voxeo will continue to sell and enhance Micromethod products and will also integrate them with the company's Prophecy premise software, Evolution hosting, and new products and service to be announced over the next six months.

For more:
- Voxeo acquires Micromethod release

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Digium CTO parses unblocked Caller ID hack

Doug Mohney — Tue, 22 Jul 2008 12:05:53 -0400

Normally, punching *67 should block Caller ID information being passed through to a receiving caller. But, as security consultant Kevin Mitnick has demonstrated and Digium CTO Mark Spencer explains, it's not 100 percent foolproof.

At The Last HOPE hacker conference over the weekend, Mitnick demonstrated how an appropriately configured Asterisk box and a suitable SIP trunking service can be used to deliver Caller ID information even on inbound calls that have a "Private" flag set.

"There are legitimate reasons why you need to set the Caller ID to normal [and carry that information forward,]" said Digium CTO Mark Spencer. "If, for example, I'm in an enterprise environment and I want to have calls forwarded [from my office number] to my cell phone, [the PBX] needs that information."

Mitnick used the "enterprise class" VoIP/SIP trunking provider FlowRoute to get a phone number (DID) and service that would deliver all of the call information to an Asterisk server. The Asterisk server is simply setup/scripted to pass along all Caller ID information for inbound calls regardless of the setting of the privacy flag on the call.

Spencer also noted that Caller ID information is also carried along and recorded for "private" calls to toll free numbers; the information is necessary for proper billing.

Mark is not happy with the use of Asterisk for questionable uses, but since it is open source, there is little he can do about it. "I hate to say it, but the same reasons why Asterisk is attractive to a lot of businesses, it's low cost, it can be easily tweaked, it's more flexible, make it easy for using it for an illegitimate purpose," said Spencer. "It's a very powerful platform. I'm not thrilled about it being used for fraud and I'm not thrilled with companies who build products on it in competition with Digium, but there's not a lot I can do about it."

For more:
- Engadget snags Mitnick demo video from The Last HOPE conference

Related articles:
Last Hope Launches Security Season
VoIP Security and the Circle of Trust