CounterPath has long been known for its SIP softphone clients, Xlite and Bria. Xlite is a free PC-based softphone client which provides basic voice calling and instant messaging features.

Bria is their full-feature business grade offering which:

• Provides access to an IP-based telephone system or service from a PC (Windows, Mac and Linux), iPhone or Android
• Can communicate with contacts from multiple sources, such as Microsoft Outlook, mobile phone address books and company directories
• Provides presence status (currently PC editions only)
• Supports a wide range of business standard call management features
• Incorporates support for chat sessions, including group chat (currently PC editions only)
• Has security features that allow firewall traversal

Over the past year CounterPath has released its Bria iPhone (and iPod Touch) edition with several updates (current version 1.2.1) as well as Bria Android edition. Both work over 3G or WiFi connections. A key feature is the ability to use the iPhone or Android as an extension to a PBX. From their product description:

Using the iPhone or iPod’s existing contact list, Bria iPhone Edition facilitates easy and effective communication management with an intuitive interface that accommodates multiple calls. Call functionality includes the ability to swap between two calls, merge and split calls and perform attended and unattended transfers.

Yesterday CounterPath released Bria iPad Edition which basically can turn an iPad into an IP phone for use in business telephone systems that support SIP. Effectively it can replace a desktop IP phone but also allows you to take that phone off the desk and accompany you on your travels. Your extension becomes physically available wherever you have your iPad as long as it has Internet access via 3G/LTE or WiFi. From their product description:

Bria iPad Edition facilitates easy communication with an intuitive interface that includes all of the standard phone features such as call display, call history, voicemail indicator and multiple call support. Multitasking functionality also allows Bria iPad Edition to run in the background while you use your iPad, allowing you to reference other applications while on a call.

And from the press release, which lists in detail several key features:

Unlike other VoIP voice applications marketed for the iPad that were introduced primarily for the iPhone, Bria iPad Edition is designed specifically for all Apple iPads. This iPad-centric foundation means that Bria iPad Edition leverages the device’s large, high-resolution display, providing a richer user experience that enhances the features found in the Bria iPhone Edition.

Bria iPad Edition’s separated navigation panes make it effortless to find contact information and initiate calls. A one-touch popover dialpad is elegantly minimized yet easily accessible at all times, while the iPad’s native keyboard is also available for entering alphanumeric SIP URL/addresses.

During the past year I have set up Bria on a MacBook and iPhone, using it as one of my “phones” that is hooked up to my FREETALK Connect telephone system.  Yesterday I added Bria iPad Edition to my iPad and have done some inbound and outbound test calls through the FREETALK Connect. Effectively my iPad has become one more “phone” when using the FREETALK Connect’s “Find Me” feature where calling my extension rings a Linksys IP phone, my Skype client, my BlackBerry (via an external call) and my iPad.

I made a call with Alec Saunders who also has the same configuration; both iPads were connected via WiFi to our local LAN, each of which included a FREETALK Connect. During the call I was able to record our conversation; the screen had options for transferring the call, adding a third party and either accepting or placing another call.

With all three smartphone/tablet editions, Bria interfaces to the device’s local address book and runs in background to answer calls while running other applications. During a call I also confirmed that you can also reference other applications while the call is handled in the background. As shown in the images from the call screen it also includes features such as multi-call support, call transfer and three-way calling.

My only concern is that it requires some technical knowledge to configure with the FREETALK Connect but that is a one-time exercise. While my experience involved a business telephony system, Bria iPad Editon comes preconfigured to work with several SIP service providers and it can be manually configured for any SIP-based service.

During a call with Todd Carothers, CounterPath’s Senior Vice-President Marketing and Products, , he pointed out that one of their learning experiences with these Fixed-Mobile convergence clients was how to deal with bouncing between WiFi and 3G connections in, say, a multi-level building. Basically they will incorporate into Bria a button that provides control over this situation by moving the call between 3G and mobile at the user’s discretion. Also Bria will soon be incorporating features to support SMS messaging and video calling.

Bottom line is that with Bria iPad Edition, my iPad has become a fully-featured IP phone that can be used through an business-grade telephony system for small business from any location worldwide provided there is a wireless Internet connection.

Bria iPad Edition is available at the iPad App Store at a price of $14.99.

One final comment: I have Truphone for iPad and, now, Bria for iPad. When will we see a unique Skype for iPad? If you use Google’s keyword tool, you’ll find that “Skype iPad” (and variations) is one of the more voluminous Google search requests associated with Skype. Delivering Skype for iPad will demonstrate that Skype is continuing along its path of Skype Everywhere, even after the announcement of the pending Microsoft acquisition. (It should be noted that outbound calls made to try  out Bria iPad Edition were made over a SkypeOut service via the FREETALK Connect.)

Full disclosure: The producer of the FREETALK Connect has previously been a client of the publisher’s business development services; at this point, there is no business relationship. The publisher is opening a business providing consulting to small businesses who want to implement Skype and/or IP-based communications solutions. More details to follow next week. CounterPath’s Bria is simply one of the softphone products experienced over the past year while determining configurations that work with FREETALK Connect.

Related articles

• Bria iPad Edition 1.0 a winner! (saunderslog.com)
• Apple eases rules on iPad, iPhone subscriptions (seattlepi.com)
• Bria extends VoIP phone client to iPad (tuaw.com)

Since the initial announcement of Skype for SIP a year ago today, with the goal of integrating some elements of Skype’s infrastructure into legacy PBX systems that support SIP, Skype has been expanding its partner base as well as incorporating its deployment into the Skype for Business program.

To repeat: the purpose and goals of Skype for SIP are:

• to allow small-to-medium businesses to reduce their international calling cost to over 180 countries through use of Skype Out business subscriptions (without the need to negotiate and manage country-specific contracts with the telcos in each country)
• to provide a free, optionally international, click-to-call inbound calling capability through Skype Online numbers
• to allow integration of these elements of Skype into legacy PBX systems that support SIP

Recalling that Skype for SIP is still at a public open beta stage, Skype participated in four announcements yesterday that expand this ecosystem and the open beta activity with a “third party” Skype for SIP certification partner and three new PBX Gateway partners:

• Skype Selects tekVizion Labs to Manage Skype for SIP Interoperability Certification Program

tekVizion Labs has been introduced to the Skype partner community as the interoperability experts to help expedite Skype for SIP interoperability testing and certification. In addition to Skype for SIP interoperability testing, tekVizion Labs can provide consulting, integration and support services to Skype’s customers and partners and the systems integrators and value added resellers they work with.

• Grandstream Achieves Skype for SIP Interoperability Certification

Interoperability with Skype for SIP gives SMBs the ability to receive inbound calls through their Grandstream GXE IP PBX or related gateways from more than 521 million registered Skype users around the world at no additional cost to the Skype user when calling through Skype. Businesses can leverage the investment in their corporate website even further by enabling Skype users to simply click on a global click-to-call button on the website to place immediate calls to sales or support via Skype. Skype calls received through the Grandstream IP PBX are handled by the PBX in the same way as other inbound calls. For outbound calling, Grandstream customers can realize significant cost savings to landlines and mobile phones around the world using Skype’s low-cost calling rates.

• Skype for SIP now Interoperable with AudioCodes Mediant 1000 Multi-Service Business Gateways

By certifying the AudioCodes Multi-Service Business Gateways with Skype for SIP, customers can benefit from interoperability with hundreds of TDM PBXs, Key Systems, and IP-PBXs that have not been able to connect to Skype so far. In addition, if a company buys and associates an online Skype number with their AudioCodes Mediant 1000 Multi-Service Business Gateway solution, it can then receive inbound calls via Skype from business contacts and customers calling from landline and mobile phones.

• VoSKY’s New PBX Gateway Is [Amongst] Industry’s First to Achieve Skype for SIP Certification

The VoSKY SSG is available in two versions, supporting either 4 or 8 channels. It offers seamless integration with Skype for SIP Open Beta with any analog PBX or key system, connecting via a conventional (FXS) analog port. It allows businesses to take advantage of Skype’s cost savings and enhanced business features without restructuring their existing communications network and infrastructure.

As a result the Skype for SIP ecosystem grows beyond the initial partnerships announced with Shoretel, SIP Foundry and Cisco. Skype for SIP Certification ensures that Skype’s partners meet established standards not only for the technology behind the implicit interoperability required but also the user experience. And these partnerships allow Skype to leverage each vendor’s existing customer base to build awareness and revenues while disintermediating and disrupting legacy long distance business models.

Bottom line: With all the various Skype for Business activities coming this year, including the launch later this spring of the FREETALK Connect PBX incorporating both Skype for SIP and Skype for Asterisk (which also supports free Skype-to-Skype calling), small-to-medium businesses will have several options to both lower their communications cost, have customer calls directed to appropriate employees and/or end points, and expand their markets through low cost communications to new international markets.

Link to a list of all Voice On The Web posts discussing Skype for SIP.

Full disclosure: In Store Solutions, producer of the FreeTalk Connect PBX, has become a client of Denali InterConneXions, publisher of Voice On The Web, building on the author’s previous business development experience with establishing partnerships that can assist with the promotion of a primary vendor’s offerings. A more complete statement will follow shortly.

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• Grandstream Achieves Skype for SIP Interoperability Certification (eon.businesswire.com)
• Skype for SIP Now Interoperable with AudioCodes Mediant 1000 Multi-Service Business Gateways (eon.businesswire.com)
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• FREETALK Connect, the worlds first Skype-enabled Unified Communications system. (saunderslog.com)
• Skype for SIP could save businesses big bucks (thenextweb.com)
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Pardon my exuberance but I am about to write about a product that I have been beta testing for the past few months. Seven months ago I knew very little about small business PBX’s other than to know they had big $$$ signs associated with their operation.

Back in 1992 my employer of the day paid $28,000 for a PBX that would service our local Canadian office of six to ten employees – and all the extensions had to be physically in the same office. Of course we would also be paying significant long distance charges, especially for out-of-country calls. The same employer was paying about $10,000 per month for a T1 line between California and their European office in Ireland. The economics of worldwide communications has changed somewhat since then; today’s announcement from In Store Solutions of the FREETALK Connect small business PBX takes small businesses further down that path.

About six weeks ago, while visiting In Store Solutions head office in Spain, using Skype for iPhone in a WiFi zone at my hotel room, I called a SkypeID that gave me the options to press 1 for Jim, 2 for another person and 3 for another person. I pressed 1 and a few seconds later my BlackBerry was ringing me back in Spain, But the PBX was at my home near Toronto, Canada as was an attached Linksys VoIP phone that the PBX also rang simultaneously with my BlackBerry in Spain. Had I answered my BlackBerry the only charge would have been my voice roaming package $1.25 per minute. Had somebody else originated that call in the way I did, I also would have had the option to answer the call, with no resulting charges for either party, on my PC’s Skype client which at the time was also in my hotel room in Spain¹.

Bottom line: any Skype user worldwide could have reached me via this PBX on a “local” or “remote” phone, a mobile smartphone or my Skype client with minimal, if any, costs for the call. And, in practice, geography as a factor to Find Me or Follow Me was not in the equation.

Today In Store Solutions announced FREETALK Connect, a small business PBX that changes the algorithm and business model for small business voice communications. With FREETALK Connect a business can benefit from:

• accepting inbound calls made via Skype, a Skype Online number, a SIP client, Asterisk (IAX2) or the PSTN, via either an analog line or a VoIP service provider.
  • incorporating Find Me, Follow Me features
• accept calls on local analog phone lines, local or remote VoIP phones, mobile phones anywhere or remote PSTN-accessed phones – all provisioned within the PBX.
• placing outbound calls over Skype, SkypeOut, a SIP-connected VoIP service or an analog PSTN line.
• managed voice mail including consolidation of voice mail and email into one mailbox

As I experienced the different communications paths this product opened up, I realized that this box can revolutionize how small business communicates with their suppliers, customers and business partners – worldwide.

The press release lists all the services that will be on board the FREETALK Connect when it becomes available for purchase by U.S. registered Skype users in March on the Skype Store:

Some of the key features from the Jazinga platform found in the FREETALK Connect include:

• Callback / Dial-around
• Access to Skype Buddy lists
• Auto Attendant / IVR
• Paging
• Call Parking
• Remote Extensions
• Music on Hold
• Conferencing

The FREETALK Connect also has an easily configured and updated:

• Managing routes to users, telephone services, and applications
• Providing SIP/Skype telephone service management
• Router management (networking, port forwarding, DNS, DHCP)

“Jazinga’s products consistently ensure call integrity by integrating quality of service and prioritizing voice traffic on the network into an affordable, simple product,” said In Store Solutions COO Craig Smith. “There was no question that FREETALK wanted to partner with Jazinga to develop the FREETALK Connect, because it continues our goal of working with the best providers to distribute outstanding products around the world.”

“FREETALK Connect is designed for small businesses with between 2 and 49 users, an undersold market that desperately needs UC functionality,” said Randy Busch, CEO of Jazinga Inc. “As a result of our partnership with In Store Solutions, the telecom technology playing field is much more level between larger enterprises and their smaller competitors.”

In terms of the technology within FREETALK Connect it incorporates both Skype for SIP and Skype for Asterisk to provide an optimum level of Skype-based services but at the same time providing inbound calling flexibility and optimizing outbound calling expenses, all combined with the call integrity features quoted above.

FREETALK Connect will requires users to have a Skype for Business account, available through the Skype Business Control Panel, associated with each unit. More on this aspect of FREETALK Connect is the subject of a future post.

Bottom line: Targeted at small businesses with 2 to 25 employees and/or business partners with its wide range of inbound and outbound calling options, an attractive price point, incorporation of Skype for Business infrastructure and the ability to self-configure via wizards, FREETALK Connect represents a game changer in the small business communications market. Lots more will be written about this product as the applications experiences evolve.

¹In this situation I could not have the Skype client answering option since I was using my Skype account to originate the call.

Full disclosure: In Store Solutions has become a client of Denali InterConneXions, publisher of Voice On The Web, building on the author’s previous business development experience with establishing partnerships that can assist with the promotion of a primary vendor’s offerings. A more complete statement will follow shortly.

Other posts:

Saunderslog: FREETALK Connect, the worlds first Skype-enabled Unified Communications system.

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Skype launches an Open Beta phase for its Skype for SIP service. Over 10,000 businesses worldwide have lined up to join the program.

Over the past eight months, Skype has been running a closed Skype for SIP beta program. A couple of PBX vendor agreements have been signed while Skype personnel have been having discussions with potential Skype for Business customers about their use of business communications processes.

Yetersday Skype has announced the launch of the Skype for SIP Open Beta program. According to Skype’s press release nearly ten thousand businesses around the world have applied to participate in this open beta phase. However, this phase is also open to any new applicants, whether PBX manufacturers, SIP gateway vendors or individual enterprises, who want to learn how they can take advantage of Skype for SIP in their business communications.

And why the interest? Skype for SIP:

• allows a business to provide inbound calling from a Skype client (whether on a PC or embedded in devices such as the RTX 3088 Dualphone), web-based click-to-call buttons or a Skype Online number through a SIP-enabled PBX or Unified Communications system.
• provisions outbound calling to landlines and mobile phones worldwide at SkypeOut rates
• has a monthly fee of $6.95/€4.95 per month per concurrent voice channel (which can handle either inbound or outbound calls)
• is managed from the Skype Business Control Panel:
  • to create and manage employees’ or outsourced resources’ Skype accounts
  • to subscribe to Skype Online numbers
  • to allocate SkypeOut credits to employees’ Skype accounts
• allows management of Skype calls via the existing PBX or UC features, such as call routing, conferencing, phone menus and voice mail
• can bring significant communication cost savings to enterprises, whether large, medium or small, especially if they are addressing continental or worldwide markets

From the Skype press release:

A company’s IT administrator just creates and registers a Skype Business Control Panel (BCP) for the organization and then follows the instructions to configure Skype for SIP to work with its existing SIP-enabled communications system. This Web-based tool allows a company to control its Skype usage from a single point and enables centralized billing, easy allocation of Skype Credit, subscriptions and online numbers to users, as well as reporting about both usage and spending. Furthermore, it gives companies the ability to manage internal employee accounts, including who uses Skype and who doesn’t and what features they can use.

As an example of the cost savings, Enterprise Business Development Manager Matthew Jordan described a company who would replace a T1 line with 23 lines at a cost of, say, $400 per month plus per minute toll charges, at up to,say, $0.03/minute, for both inbound and outbound calls. With Skype for SIP, the company would be looking at a cost of $161 per month for the 23 voice channels plus low SkypeOut rates for outbound calls to landlines and mobile phones anywhere worldwide and no charges for inbound calls.

From Skype’s perspective, a primary goal of this beta phase is to monitor usage to determine experience-based usage patterns. To facilitate this goal,  during this open beta Skype for SIP will have a 20% burstage allowances such that a company with 10 channels can actually use up to 12 channels as a means to optimize bursting levels required  based on real calling patterns. With over 10,000 Skype for SIP customers, they hope to gain statistically significant data as well as learn more about scalability and robustness issues.

The entire systems is designed to be global and scalable as numbers build. The required SIP gateways (at sip.skype.com) will be independent of SIP gateways used for legacy consumer SkypeOut calling.

Skype for SIP is a classic case of a transitional offering. Enterprises continue to use their legacy SIP-enabled PBX and then add Skype for SIP such that they can continue making calls but run them over Skype for SIP to reduce costs as they feel more and more comfortable with the advantages of the Skype for SIP exerience. It provides a low cost means of adding voice channels; it also allows a company to expand their calling into new countries at low SkypeOut rates with the only provisioning being the initial setup of Skype for SIP via the business control panel.

More information, including an introductory video, can be obtained and Business Control Panel accounts set up at the Skype for SIP beta website at sip.skype.com.

• Peter Parkes: Skype for Business Blog: Skype for SIP beta program opens up to all businesses.
• All Voice On The Web posts mentioning Skype for SIP.
• Andy Abramson talks about the “Skype Pipe”; he thinks “…. Skype just created a massive pipe into the corporate market.”

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• Skype Taps ShoreTel for Enterprise Play (gigaom.com)
• Skype For SIP Interoperability With Cisco Gear Confirmed (techcrunchit.com)
• Skype Selects Acme Packet’s Session Border Controllers to Ease Interoperability with IP PBXes for Skype for SIP Beta Offering (skype.com)
• How Skype Plans to Dominate Business Telephony (gigaom.com)
• Barriers to using Skype for business fall away (infoworld.com)

Over the past two weeks we have started to see signs of a transition period for Skype’s developer support activities. Yes, Skype Extras is dead; however the current Skype API’s remain supported. Jonathan Christensen gave Michael Arrington at TechCrunch some hints of forthcoming developer support activity.  Skype for Asterisk was launched three weeks ago. All this builds on the “Skype Everywhere” theme announced last January at CES 2009.

And whither the Skype for SIP activity whose beta program was launched six months ago? Over the past two weeks we have seen a couple of Skype for SIP launch partner interoperability announcements incorporating Skype interoperability into their Unified Communications offerings:

• the initial announcement covered incorporation of Skype interoperability as a feature for Shoretel’s 11,000 enterprise customers using Shoretel’s IP Phone Systems.
• this was followed by an announcement of incorporation into the SIPfoundry’s open source enterprise IP PBX unified communications solution for organizations running a sipXecs IP PBX.

But today’s announcement of Skype for SIP interoperability with Cisco’s Unified Communications 500 Series for Small Business opens up significant new markets for Skype through Cisco’s extensive worldwide reseller channels. From the press release:

“Cisco is committed to delivering easy-to-use technology to small businesses to help them run their business better and gain a competitive advantage,” said Mark Monday, vice president and general manager, Small Business Solutions business unit, Cisco. “We are excited about Skype for SIP interoperability since it adds a very cost-effective choice to the many other options within the Cisco Unified Communications 500 Series.”

Note that Cisco VARS will be required to register for a new Skype Service Partner certification program covering configuration of the Cisco solution to support Skype for SIP. And that the Skype for SIP interoperability with the Cisco 500 Series goes through third party certification at tekVizion Labs, an independent interoperability testing facility in Richardson, Texas.

In all three cases Skype for SIP increases the enterprise’s or organization’s worldwide communications choices while reducing costs through:

• taking advantage of Skype’s interconnection to the PSTN that provides low cost (SkyepOut) calling to most countries worldwide
• receiving inbound calls from Skype users via a global click-to-call button on the enterprise’s or organization’s website
• receiving inbound calls via Skype Online numbers in 25 countries (previously SkypeIn)
• interoperation of these features with all the other features of the PBX offerings of  Shoretel, sipFoundry and Cisco’s 500 Series, such as call transfer, call forwarding, voicemail, etc.

For organizations and enterprises, the availability of Skype for SIP, Skype for Asterisk and the Skype API’s used by Skype’s other business communications partners, such as OnState and Scendix (PamFax), brings about significant overhead benefits. This offering:

• ensures that, with one service, the enterprise can achieve low cost calling to any country worldwide without the need to mange and use country- or region-specific calling plans offered by the legacy PSTN telcos.
• eliminates a layer of communications cost overhead through the value-add of making all Skype’s global termination agreements available through one source via the various Skype for Business partners’ distribution and reseller channels.

Bottom Line: These announcements all build on Skype’s “Skype Everywhere” strategy. The key point here is that, going forward, Skype’s developer and partner activities will focus on transitioning business markets into Skype services by Skype-enabling legacy communications services and platforms, complementing its prior focus on applications that bring about communications enhanced business processes.

As predicted last spring these announcements execute on Business Long Distance: Death by 1,000,000 PBX’s.

Dan York says “Congrats to Skype and Cisco on this announcement”.

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As the supplier of a key element in delivering voice-enabled solutions, Global IP Solutions’ activities provide a window into the world of voice- and video-enabled applications and IP-based communications beyond the Skype ecosystem. The GIPS Voice Engine contains API’s that allow a service provider or application publisher to readily incorporate robust, reliable, high quality voice and video calling into their offerings. A recent announcement from Global IP Solutions provides a classic example where we catch a glimpse of the role of IP-based communications in delivering commercial and large enterprise communications solutions.

I first learned of CommuniGate Systems as a provider of Internet-based communications infrastructure software solutions when one of my business planning clients from ten years ago, a business services ISP, was installing the CommuniGate Pro Messaging Suite as the backbone of their email offerings. From that time until last June, when an acquiring ISP failed to provide an appropriate level of support, my own email had been run through this particular CommuniGate-enabled server.

But CommuniGate, building on their CommuniGate Pro communications architecture, has gone on to be a major supplier of unified communications infrastructure. From their website:

CommuniGate Systems’ goal is to consolidate all forms of Internet Communications into one address space, making the single address for email, IM, VoIP and video calling, more productive, portable, and independent of tariffs and tolls of closed network topologies. We see all Internet Communications becoming accessible to multiple media types through one account, providing true portability of an “address” no matter where you access the Internet.

Communigate’s 12,000 strong customer base services over 2 billion email accounts through broadband, wireline and wireless service providers as well as larger enterprises who require “private” communications services. Secure airline communications services, such as required by British Airways, is one significant vertical enterprise market for CommuniGate.

In the fall of 2007 CommuniGate announced the launch of a Flash-based unified communications client, Pronto!, that now serves as a “single dashboard for e-mail, collaboration, secure instant messaging, RSS feeds, and Rich Media”. Pronto!, with an Outlook-style interface and built on Adobe Air, supports contacts, calendar, email, Instant Messaging, voice calling, conference calling, voicemail, news, photos, audio and video player. Users can access their communications services via the Pronto! client on any PC and network connection.

Over time CommuniGate has been adding features to Pronto!; for instance, last summer they announced the addition of video mail, contact and calendaring features, RSS “news” feed support and audio media support. But they also had been evaluating solutions that would improve the voice calling experience with a more robust voice engine that incorporates the enhanced sound quality characteristic of wideband audio HD Voice. After an investigation of their options over the past few months, it was announced at the end of March that CommuniGate had signed an agreement with Global IP Solutions to incorporate the GIPS Voice Engine into Pronto.

“Global IP Solutions has been a pioneer in HD Voice for many years. They have over 800 million downloads and a growing list of brand name customers to prove it,” says Jon Doyle, VP Business Development CommuniGate Systems. “GIPS has a unique engineering team that is focused on addressing the processing issues related to IP communications on various network topologies. With GIPS superior engineering skills, we are able to deliver on the promise of a truly business grade UC experience.”

In a recent interview with CommuniGate’s John Doyle and Larry Golob, GIPS Senior Director of Business Development it was pointed out that:

• CommuniGate was seeking a solution that met HD Voice specifications supporting 8KHz wideband audio during voice calls; the GIPS wideband iSAC codec meets these specifications.
• However, they were also looking for a full voice engine package that could call quality issues such as jitter, packet loss, echo cancellation while adapting readily to network congestion conditions through bandwidth management features.
• In the future, CommuniGate may elect to investigate Skype’s superwideband SILK voice codec for incorporation into their GIPS Voice Engine; however, as its royalty-free licensing had only recently been announced, they would consider the SILK codec at some point in the future, especially if customer demand existed
• CommuniGate is also evaluating the expansion of the role of video in Pronto!; they will be evaluating GIPS video technology later this year.

While services using Pronto! are distributed to end users via either Internet Service Providers or large enterprise “closed network” operations, one can experience Pronto! by setting up an account at TalktoIP.com.

A final question: how will Pronto! users be able to have access via mobile smartphones? Is there an opportunity for GIPS’ mobile offerings?

Bottom Line: Meeting large enterprise and service provider requirements for low cost, secure IP-based communications creates a separate market space – of significant size – not addressed by Skype’s current business model and strategy. On the other hand, it will be interesting to see if CommuniGate incorporates connections with Skype users as another offering within their unified communications suite. Certainly Skype for SIP opens up an interesting opportunity for CommuniGate to consider as they move forward.

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Following a link from Dan York’s update to his post outlining technical details of Skype for SIP, I found a link to this post from IfByPhone’s Irv Shapiro: Why Skype for Asterisk is more important than Skype for SIP.

First, Irv has done perhaps the best articulation yet differentiating Skype for Asterisk from Skype for SIP:

Skype for Asterisk, which is still in closed beta, is a true Asterisk channel driver. This allows Asterisk based solutions to make, receive and transfer Skype calls. A significant capability of the SFA solution is its support for terminating a call to a Skype user name, for example a PC based user of the Skype client.

Skype for SIP is a very different animal. This service provides VOIP trunk support for existing SIP based PBX systems, which may include Asterisk. Unlike SFA where calls may be placed to any Skype user, SFS calls may only be terminated to PSTN end points.

Irv goes on to provide his view on how this plays in the Voice/Telco 2.0 marketplace. Then he provides an example of where he sees the benefit of SFA for hosted services:

From our perspective as a cloud telephony company, providing hosted telephone applications, SFA is much more interesting. While either service would allow us to provide IVR services to Skype users, the SFA Asterisk channel driver architecture allows us to terminate calls into call centers with no PSTN transport. Each call center Agent would just utilize a headset connected to a computer running the traditional Skype application. Customers calls would be able to originate from either a PSTN device or a Skype client, then route through the Ifbyphone IVR infrastructure and terminate to a call center via Skype’s computer to computer transport. This has the potential to change the cost structure associated with supporting call centers. A Skype based call center would not require a PBX or for even any centralized telephony components. The call center agents could be virtually located anywhere in the world on high quality Internet connections.

Keep in mind that calls terminated on Skype, as demonstrated on the 3 Skypehone service, have no associated termination charges. Long time Skype-based call center provider OnState has already demonstrated this business model; their hosted OnState Call Center service eliminates the need for a call center PBX. While a (prospective) customer can call in via several modes (Skype Online number, DID numbers, 800 numbers, Skype or click-to-call from a website), all the calls are terminated on Skype via their agent client.

Andy Abramson at VoIP Watch has, in a reverse manner, demonstrated this cost advantage also in his post, All Google Voice Needs Is a Little More SIP and Skype’s Game Changes, where he points out that, if Skype for SIP did terminate calls on a Skype client (i.e. – supported SIP—>Skype), then one could terminate Google Voice calls at a Skype client. But, as I have commented on that post and included in Skype for SIP: Sorting Through the Issues, Skype for SIP supports three types of connections and “SIP—>Skype” is not one of them.

Irv demonstrates that Skype for Asterisk has some interesting potential; it’s simply going to require more creative innovation that goes beyond simply making call connections through a SIP interface. It’s a matter of invoking Alec Saunders Voice 2.0 Manifesto which points out that the value lies in the applications.

Skype announces launch of a beta version of Skype for SIP, connecting IP-PBX’s to the Skype cloud.

Dan York, while a keen supporter of Skype, has been campaigning for a Skype to SIP interconnect on his Disruptive Telephony blog. In a post last fall, “Skype and SIP – the two sides of the issue raised by Michael Robertson”, he says:

We need to build the interconnect.

Yeah, there are a TON of issues out there that we still need to address to build that interconnect. There’s a whole host of security issues… there are billing issues… there are trust issues… there are network plumbing issues. Yes, there are all those issues. But if we are to succeed in ultimately bringing about the rich communication experience we want, we need to make this happen.

And for that, Skype’s walls need to come down…. at least a bit.

What we need is that Interconnect from Skype’s cloud out to the emerging IP infrastructure. Think about it… Skype right now has a two-way interconnect between Skype’s cloud and the cloud we know as the PSTN. It’s called “SkypeOut” and “SkypeIn” (or whatever marketing names they are being called now). If you dial my SkypeIn number, you can reach me on Skype wherever I am. From my Skype client, I can call anyone on the PSTN. The two-way interconnect is already there.

This morning Skype announced the launch of a beta version of Skype for SIP as a major initiative for further penetration into the business market. Basically Skype for SIP allows SIP PBX owners to access the Skype cloud as additional inbound lines , via Skype Online numbers, and outbound extensions to their existing PBX, via SkypeOut. From the press release:

Key Features

The beta version of Skype For SIP will enable business users to:

• Receive and manage inbound calls from Skype users worldwide on SIP-enabled PBX systems; connecting the company Web site to the PBX system via click-to-call
• Place calls with Skype to landlines and mobile phones worldwide from any connected SIP-enabled PBX; reducing costs with Skype’s low-cost global rates
• Purchase Skype’s online numbers, to receive calls to the corporate PBX from landlines or mobile phones
• Manage Skype calls using their existing hardware and system applications such as call routing, conferencing, phone menus and voicemail; no additional downloads or training are required.

In an interview with Skype Senior Product Manager Chris Moore, I learned:

• Skype’s’ goal will be to have a totally user friendly interface such that Skype for SIP is a seamless service
• Administration will be done through an extended Skype Business Control Panel interface which, amongst other activities will establish Skype for SIP account mapping, manage online numbers and perform appropriate configuration for the various PBX’s.
• Skype will be looking to work with the various SIP PBX and Unified Communications vendors, such as Cisco, Avaya, Nortel and Siemens. When I asked about Microsoft Response Point, which had been demonstrated to me the previous day, Chris confirmed, yes, they were another prospective vendor to be approached.
• during the beta, Skype will be used on a pay-as-you-go basis; pricing for the service once it is launched has yet to be determined
• Skype will be including security issues as one major consideration (I expect Dan York to have more to say on this issue)
• The service will be configured to take advantage of Skype’s royalty free SILK codec; if both endpoints of a call support SILK, then the SILK support will carry across the Skype for SIP interface.

This has to be a landmark announcement for the IP-based communications space. It effectively allows businesses to integrate the Skype cloud into existing infrastructure to handle all long distance and international calling. Some other initial observations:

• Does Skype for SIP impact the space currently served by Vosky’s PBX services?  It definitely leaves open for Vosky the analog PBX market; one question will be how does Vosky figure out a way to work Skype for SIP into their business activities and offerings during the six- to twelve-month beta?
• It’s currently a beta; applicants for beta participation can apply at www.skypeforsip.com. From the press release: “Applicants will need to be businesses, have an installed SIP based IP-PBX system, as well as a level of technical competency to configure their own SIP-enabled PBX. The initial beta is available to a limited number of participants.”
• Skype also expects to determine an appropriate distribution channel. While ease-of-use is a key goal, there will still be small business owners who will want third party expertise to manage their communications activities. So expect Skype to look to a VAR or Systems Integrator channel to support implementation.
• Skype for SIP broadens Skype’s penetration into the small-to-medium business market. Where Skype for Asterisk introduces a Skype feature-rich Asterisk channel to Asterisk PBX’s, Skype for SIP simply brings Skype cloud connectivity to a major subset of existing small business PBX installations.
• Along another feature axis we see hosted services vs on-premise implementations. Will widely geographically distributed small businesses favor a hosted service, such as OnState’s Virtual PBX, or one of these on-premise PBX offerings.

One sure call: the market for business long distance, especially international calling, is about to be radically disrupted as these services come into play. It brings about new business opportunities and, most significantly, makes any business a global business, if the will is there. With probably over one million IP PBX’s out there (Skype’s press release says over 438,000 were sold in 2008 along), Skype for SIP brings the slow but sure death of legacy international calling, one PBX at a time.

And, of course, Dan York brings his deep perspective, not only on how Skype for SIP works, but also on the security and codec issues surronding Skype for SIP: Skype tears down more walls with “Skype For SIP”.

With the launch of Skype for Windows 4.0, Skype delivered its first client incorporating Skype’s new superwideband SILK codec providing a new user experience in voice calling. For example, Alec Saunders in “Skype 4.0 audio: smooth as SILK”:

Perhaps the biggest improvement, though, is audio quality.  We all thought that Skype audio was great, right?  Skype’s internally developed SILK codec slipped into the final release, despite not having been in prior betas. A wideband codec that delivers the goods at half the bitrate of prior codecs, SILK makes talking on Skype a pure pleasure.

The SILK codec will be rolled out with the final release of Skype for Mac 2.8 (currently in beta but without the SILK codec) in April and in the next Linux release (no date given). The coding was designed to allow easy embedding into hardware devices, such as sound cards, headphones and Skype phones. But this still limits the SILK experience to Skype-to-Skype calls. So they looked at the question of how can Skype most readily advance the deployment of this high quality codec  and the resulting user experience at billions of additional voice conversation end points?

At eComm 2009 this afternoon Skype announced, during Jonathan Christensen’s keynote presentation, that the SILK codec will be licensable, royalty-free, to device manufacturers and others; it has been designed independent from Skype signaling protocols such that it can be used in voice streams incorporating the SIP protocol. Jonathan mentioned that there is no requirement to use the SILK codec in conjunction with a Skype implementation.

From an interview with Jonathan Christensen, Skype’s GM for Audio and Video, with Alec Saunders in Skype’s SILKen bear hug: “Further, SILK media streams pass unchanged across Skype’s own media gateways, meaning that SIP endpoints have suddenly become much more compelling in the Skype world”.  Alec goes on to discuss two reasons for Skype’s strategy: (i) to spread superwideband audio everywhere and (ii) to give existing carriers a bear hug. Read the post for details.

In his post on this announcement, Jonathan pointed out what the three year Skype SILK development process focused on:

• • improving audio bandwidth going from 8 kHz to 12 kHz, meaning that a SILK conversation sounds like you are in the same room as the person you are speaking with
  • providing real-time bandwidth scalability to deal with degraded network conditions
  • balancing codec optimization between voice, music and background noise, each of which can have an impact on the overall user experience
  • delivering a robust solution that delivers a more consistent audio experience, regardless of network conditions and an individual user’s voice signature

I have watched as a wireless carrier attempts at new video calling services are introduced and subsequently fail due to a lack of adoption – the primary reason being there are too few devices out there supporting video calling and thus, there’s nobody to whom one can make a wireless video call. Skype’s announcement today overcomes a major barrier to allowing a much broader user base to take advantage of superwideband audio. It’s a first, but key, step towards a broader goal than “Skype Everywhere”, namely, superwideband audio on all voice calling.

Finally, this afternoon I had a chance to meet many members of the SILK development team who attended Jonathan’s presentation. They deserve kudos and congratulations for all their ground breaking efforts and success in setting new standards in voice call quality.

Developers can find more information at Skype’s SILK website.

Phil Wolff, Skype Journal comments on the SILK performance slides shown at the presentation: SILK performs better

Update: The transcript of Jonathan Christensen’s Codec Evolution and Industry Proposal presentation.

With the evolution of smartphones incorporating both WiFi and 3G wireless support, we are also seeing the application of Voice over IP technology as providing at least one leg of a wireless call.

• iSkoot was a pioneering example, with both their support of a wide range of wireless phones, including Nokia, Android and BlackBerry (full disclosure – I use iSkoot on my BlackBerry Bold) and their Skypephone so successfully launched on 3 services in nine countries.
• Recently we have seen the evolution of Skype Lite capable of providing access to Skype on over 100 Java-enabled mobile phones (not all smart phones), including Android.
• And Truphone has been launching services on Nokia, BlackBerry and, most recently, iPhone.

The real challenge for Skype was to get carrier adoption. Certainly the 3 Skypephone experience represents a successful business model where, as disclosed at eComm 2008, royalties associated with 3’s revenues are paid to both Skype and iSkoot. Skype announced at CES that they are developing carrier relationships in ten countries, including United States, that involve deployment of the Skype Lite client . With its new management team, including a former Motorola executive, it was only a matter of time before we would start to see deeper relationships with mobile device vendors.

And, of course, this initial device vendor relationship is with a vendor who has not exactly been a significant smartphone player in North American markets, although their market penetration has been more successful in European and Asian markets. Today, Skype and Nokia have announced a partnership where Skype will become deeply embedded into Nokia’s S60 Symbian platform with the launch device being the forthcoming Nokia N97. From the press release:

The Skype experience will be part of the address book of the Nokia N97, enabling presence – seeing when Skype contacts are online – as well as instant messaging. Nokia N97 owners around the world will also be able to use 3G and WLAN to easily make and receive free Skype-to-Skype voice calls, in addition to low-cost Skype calls to landlines and mobile devices.

The pros:

• Its support of both 3G and WiFi; this has certainly played well for Truphone adoption
• It supports the device at a platform level as evidenced by the integration with the native address book and the Nokia SIP stack. Is this a full VoIP client integration?
• The initial launch device has a QWERTY keyboard, making chat sessions on the device more viable and readily adopted
• An established smartphone vendor market leader outside of North America
• Probably a software licensing revenue generator for Skype
• Opportunity to drive (and share) SkypeOut calling revenue
• It provides free, beyond a monthly flat rate for the basic carrier service, voice and chat conversations with over 40 million Skype users worldwide

The cons:

• Nokia needs to establish North American smartphone market presence in a market dominated by iPhone and BlackBerry and where carriers largely control which devices and applications are available for use over their network
• Nokia needs to demonstrate they can provide a more user friendly smartphone user experience (a la iPhone) as opposed to simply being a handheld multimedia computer.
• Nokia needs to partner with carriers who are willing to allow open access to applications in their newly announced Ovi storefront

The questions:

• Has Nokia already signed up carriers to take on the (somewhat delayed) N97 when it launches in June? Following along with a Skype strategy announced at the CES press conference, we can probably assume that the device will first be available in non-carrier dominated markets.
• Does this represent an evolution of the Skype Lite client where chat and call setup occur over the network while the more robust and readily scalable voice channel carries the voice portion of a call?
• Where are the revenue opportunities for Skype, Nokia and the carriers?
• What is the ongoing relationship between Skype and iSkoot who pioneered this carrier adoption model? This Nokia-SKype partnership is totally independent of any iSkoot participation.
• How will the Nokia-Skype partnership compete with UMA plays such as described on the Tweet in the right?

Bottom Line: Liquid communications means turning on a device and finding Skype is “just there”. A partnership with the world’s largest vendor of mobile phones with hundreds of existing carrier relationships is not to be taken lightly. Combined with the Sony Ericssson Skype panel announcement yesterday, what hand will Skype play in working with Apple (iPhone) and RIM (BlackBerry)?

Update: Andy Abramson, VoIP Watch, attended the press conference:

Basically the underlying message the Skype CEO shared was carriers who will work with Skype will pick up users who spend more money than the non-Skype using mobile phone customer, or at least they did with 3.

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• Skype Comes to Nokia Phones (mashable.com)
• In An Effort To Become Ubiquitous, Skype Partners with Nokia (skypejournal.com)

Yesterday I wrote a post for GigaOm, Skype: Coming to a Cell Phone Near You, discussing how the announcement of two new beta versions of Skype on mobile devices gave a hint of Skype’s future mobile strategy.

At the same time Truphone announced a new version of their iPhone application. Whereas the version released at the time of the Apple App Store launch back in July only supported outbound calling over WiFi access points, the new release not only supports inbound calling to your iPhone number but also makes outbound calls via the 3G carrier networks that offer the iPhone.

Innovation driven competition in delivering low cost international calling services appears to be heating up during these challenging economic times – at least for calls originating in your “home” calling country or area code(s). Here’s a brief summary of what is evolving:

We’ve seen the evolution of two architectures for making VoIP-enabled calling from mobile devices; it’s all a matter of where the calling party’s Skype (or VoIP client) session is opened up – directly on the device or on a dedicated hosted server. This leads to two other considerations:

• Carrying the voice portion of the call from the mobile device into the network cloud: either via the carrier’s robust and proven (GSM) voice channel or over via a WiFi access point
• The need to support Skype’s instant messaging (chat and presence); this always occurs as a data activity

VoIP Client on the Mobile Device; VoIP over WiFi

Skype for Windows Mobile places the VoIP client directly onto the device. As a result the device must handle the “VoIP processing” to generate the packets that are transmitted over the supporting data network (either a carrier’s 3G network or via a WiFi access point.) As mentioned previously, it places heavy demands on the device’s resources, especially the processor (running at much lower speeds than on a PC) and the battery.

Truphone’s original voice offering also runs on the device (usually a Nokia Smartphone). While both Skype for Windows Mobile and Truphone can run over either WiFi access points or a 3G network, it is strongly advised to use these only over WiFi access points to have a reliable, robust, high quality voice service. For instance, the Skype for Windows Mobile download page says:

• Log into Skype from any WiFi zone to make free calls and send instant messages to anyone else on Skype, anywhere in the world, any day of the week.
• WiFi connection or 3G/2G data connection (we cannot guarantee voice quality over 3G/2G. You may also be liable to additional data charges so please check with your operator before using)

Truphone’s original iPhone outbound calling offering was also only available using the iPhone’s WiFi capability; however, details of their architecture were never revealed.

Accessing VoIP via a Wireless Carrier

Over the past year we have seen the rise of several services that use the alternate architecture where a call is placed via a local access point to a hosted server that then opens up a Skype client. The server-based Skype client then completes the call as a Skype-to-Skype call.

While originally pioneered by iSkoot, a service using this architecture, such as Skype Lite beta, makes a call to a SIP Gateway server via a local point of presence while data about the call is concurrently sent via the underlying data network to a hosted Mobile Gateway. This dedicated gateway then sets up a Skype-to-Skype call between the SIP Gateway — now connected to your cell phone — and the destination Skype contact. Skype chat messages can also be exchanged concurrently over the data network. We are now seeing various offerings using this architecture:

• The highly successful Skypephone offered by 3 in nine countries.
• iSkoot providing service for a wide range of phones including BlackBerry, Nokia and T-Mobile’s G-1.
• Truphone Anywhere: when Truphone found they could not offer a highly reliable service over 3G networks (largely due to device resource considerations), they launched Truphone Anywhere that allows Truphone calls to be made over a 2G (GSM/EDGE) or 3G (UMTS/HSPA) voice/data network as well as over WiFi access points.
• Skype for Mobile beta – Skype’s first attempt to go beyond Skype for Windows Mobile onto other platforms such as Nokia N-Series and E-Series devices. This never got out of the beta phase; while you could use Skype chat anywhere, the voice service was only to be available in a limited number of  countries (that did not overlap with countries where Skypephone was available).
• Skype Lite beta: building on the Skype for Mobile beta experience to a service that supports not only smartphones but also over 90 cell phones that support a Java client and include basic web browsing and data capability. According to the Skype Lite page it appears that Skype is working with carriers in ten countries to support this service.
• Truphone for iPhone 1.12 release: makes Truphone calls either over WiFi or any cellular network using an iPhone, building on their Truphone Anywhere experience.

Key features of these server-hosted VoIP client services:

• They are most cost effective when calling from your home country or local calling area. You could incur long distance or, when outside your home country, roaming charges that would run up quite quickly.
• An unlimited or high cap data plan minimizes costs associated with using these services.
• Only Skype provides a full Instant Messaging capability covering both chat and presence. Some Truphone offerings have shown support for SMS messaging.
• Calls to Skype contacts are free.
• Calls to mobile numbers outside U.S. and Canada will still invoke the charges incurred in “caller pays” mobile services.

Why only the cost of a local call? Your cell phone makes a call to a local number which puts the call through to the service’s SIP Gateway. At this point you connect into a Skype-to-Skype call for which there are no termination charges involved as a result of Skype’s unique (and secure) peer-to-peer architecture.  The same applies to Truphone where Truphone-to-Truphone calls are free.

This Skype Lite beta announcement portends that we could be seeing Skype-to-Skype calling, along the lines of 3′s popular Skypephone service in nine countries, become available to mobile customers having a much broader range of cell phones and in up to ten additional countries.

One other service that can be accessed from any phone is Mobivox. However, there you have to build up and manage your address book online such that VoxGirl can help you make your calls; it does not access your mobile phone address book. It’s purely a voice service with no messaging component (other than using SMS to facilitate setting up calls under certain circumstances).

While we’re getting a first step in driving down mobile costs for international calling, the next step needs to be finding a user-friendly way to drive out roaming costs. MaxRoam and Truphone’s SIM4Travel are starting to offer some hope on this front; however, at the moment their costs for USA-Canada calls are much more than my Rogers roaming charge. The winners will feature not only lower costs but a very friendly user interface, interacting with the device address book, that also provides the most complete ranges of services in terms of coverage and complementary conversation modes, such as IM.

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Following up on Thursday’s announcement of Skype for Asterisk for which details were posted on Skype Journal late Thursday, Dan York has published a more technical post, “Clarifying how Asterisk could possibly be used as a Skype-to-SIP gateway“, discussing how the Asterisk PBX treats incoming and outgoing calls, effectively independently of how the Asterisk PBX is accessed either internally or externally or is directed to make outbound calls.

However, the point I was making in my post yesterday was this announcement has the potential to turn Asterisk into a two-way “Skype-to-SIP” gateway. Asterisk – with the “Skype For Asterisk” module installed – could be deployed into a network where it could provide interconnection between Skype users and SIP users.

Dan goes on in three sections (with diagrams):

• Asterisk Interconnection Explained
• Diving a Bit Deeper
• So How *Might* This Work with ‘Skype for Asterisk”

And, in a concluding section “So What About ‘Skype-to-SIP’ states:

The point of my post yesterday was now that two-way Skype connectivity becomes just another channel driver for Asterisk, you have all sorts of interconnection possibilities. As a standalone system, you could connect SIP phones on an Asterisk server out to the Skype cloud.

If you’re into learning more technical detail of how Asterisk handles and directs inbound and outbound calling, Dan’s post is an excellent primer.

Also check out Dan’s previous post “More on how ‘Skype for Asterisk” actually works…” where he quotes an update post from Tom Keating and concludes:

If I understand this correctly, this has the potential to be huge! As far as I know, all the existing “Skype-to-PBX” solutions use the rather kludgey solution of basically running multiple instances of the Skype client on the system. Each “Skype trunk” is essentially just a separate instance of the Skype client. As Stefan Öberg indicates, there are serious scaling issues with this approach.

However, this has been the only options developers have had! Skype has not – prior to this (if it works how it sounds like it works) – provided any “back-end API” that would let a system interact directly with the Skype P2P cloud. The only API developers have had is the client API that lets them interact with a local Skype client. So that’s how all the “Skype-to-X” products have been built.

Does this mean that Skype has exposed some additional API that is available through this Skype For Asterisk product? If so, this could be VERY interesting…

Interesting times ahead.

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Just in case you’ve been on vacation on a remote island not reading Skype Journal the past few days, Phil and Michael Robertson have been having this debate about Skype and interop with SIP-based services (more here and here). It even spread to last Friday’s SquawkBox call.

This morning Dan York, who sits on IETF forums on VoIP security, weighed in with his perspective in a post: Skype and SIP Interop – the two sides of the issue raised by Michael Robertson where he lists all the various posts on this subject and comments:

About Wireless Openness:

So with that view, you can expect I applaud Skype’s efforts to open up the wireless networks and allow consumers to have a choice of what apps they want to run. I want the *wireless* carriers to be big, fat, dump pipes… give me an IP address on the *mobile* Internet and let me do what I want with it. Sure, the carriers can offer their own services, and maybe if I like them I’ll pay for them…. but I want the option to use other products and services – without degradation or prioritization…

To put it another way, I pay the wireless telcos for *dialtone* now. Once connected, I can call anyone and use any *voice service* over the PSTN. I could use someone else’s voicemail if I want (like GrandCentral), although the carrier’s offering may be more convenient (and is usually free). But I can call anyone on the PSTN and use any voice service I want. The carriers just provide me dialtone.

I want “IP dialtone”. I want a Big, Fat, Dumb Pipe.

So… go, Skype, go

About Skype Openness

We need to build the interconnect.

Yeah, there are a TON of issues out there that we still need to address to build that interconnect. There’s a whole host of security issues… there are billing issues… there are trust issues… there are network plumbing issues. Yes, there are all those issues. But if we are to succeed in ultimately bringing about the rich communication experience we want, we need to make this happen.

And for that, Skype’s walls need to come down…. at least a bit.

What we need is that Interconnect from Skype’s cloud out to the emerging IP infrastructure. Think about it… Skype right now has a two-way interconnect between Skype’s cloud and the cloud we know as the PSTN. It’s called “SkypeOut” and “SkypeIn” (or whatever marketing names they are being called now). If you dial my SkypeIn number, you can reach me on Skype wherever I am. From my Skype client, I can call anyone on the PSTN. The two-way interconnect is already there.

So why not offer the same on the IP side?

My feeling is that we are at a stage in the evolution of IP-based communications where the interconnectivity agreements between service providers still need to be worked through. While technically one can make the connection, it seems that every SIP gateway also requires a business agreement and transaction – similar to the agreements amongst PSTN carriers that are so transparent to our ability to make calls anywhere on the PSTN. I’m not a technical expert; I just want to be able to call as many of my contacts as possible – at as low a cost as possible – but I also want the privacy, security and authentication of Skype when connecting to other services.

In a Skype Journal post earlier today, Skype President Josh Silverman is quoted as mentioning that Skype is looking for a GM for Platform whose initial responsibility will be to lay out Skype’s future platform architecture. But Josh also assured us that user perspectives will be considered while designing this architecture. It would seem that, once the appointment is announced and Skype has a forum for user input, the debate engendered by these posts would be a good starting point for consideration of one aspect of Skype’s platform ambitions.

Guest Post: Hudson Barton is a communicatons consultant whose Borderless Communicator blog not only talks about Skype and related IP communications activity but also attempts to track Skype’s “real usage”. According to his analysis, Skype has just cracked the 30 million real or” currenlty active” user number (based on tracking Users Online vs time-of-day). What follows is his post earlier this week summarizing some issues that were raised and discussed on the Skype 3.x discussion Public Chat forum following Dan York’s recent guest post on SIP/Skype interconnectivity.

Skype’s competitors and critics continually point out that Skype’s VOIP architecture is closed and that its API is not adequate for creating a direct connection between the Skype “cloud” and the SIP “cloud”. This of course is true, but there are good reasons for it.

1. The security and reliability of the Skype cloud would be seriously compromised if SIP hackers were given the tools to create direct VOIP connections between Skype and the outside world.
2. A SIP gateway to Skype might work if it were handled like SkypeIn/Out. However, I don’t think there’s a large enough population of SIP users out there to justify the cost of SIPIn/Out. Skype is growing at a rate of 500k-1000k “real users” per month, which is probably 10x faster than the rest of the VOIP world combined. A third party could build these gateways with the presently available API, but nobody is trying it to my knowledge… presumably because there is no demand for it. In any case, the developer (even if it were Skype) would have to justify the cost of such a gateway.
3. An IM (text-only) gateway is very possible and would not compromise Skype’s security or strategic position. Look for future interconnections with major players like AIM, gTalk, Yahoo, and MSN.
4. The Skype cloud is far more complex and has far more features than the clouds of any of its competitors. It is not rational to expect any of them to create a feature-for-feature mirror of Skype even if this were something that would be good for Skype (which it is NOT). A partial list of these features: video, SMS, encryption, and file transfer.
5. Relationships between Skype and social networks like MySpace are already possible if there is a partnership agreement. It does not require a change to the Skype API. Note that MySpace is a social network… not a VOIP carrier.
6. Skype may double its revenues this year and it’s already profitable. No other VOIP carrier is profitable (unless you want to count a few of the hosted VOIP services from the Telecoms and cable companies). A gateway to Skype will help Skype’s competitors far more than it will help Skype, so from a strategic perspective it makes no sense to help the competitors survive. Without Skype’s help, they (SunRocket and Vonage for example) are failing at a rapid rate. Meanwhile, “successful” competitors like Packet8 are monetizing themselves by selling off intellectual property. Obviously they “see the writing on the wall.”

Tags: Skype, SIP Gateway, Hudson Barton, Borderless Communicator, Packet8, Dan York

Ever since the Internet became commercial in the mid-90′s, the term “Unified Communications” has been tossed around as if it is some form of “holy grail” for a communications offering. PhoneBoy Dameon Welch-Abernathy finds the term somewhat overhyped in his recent post “Unified Communications Is A Pipe Dream“:

Anyone who understands the technology knows that unified communications is a pipe dream. Perhaps within a small subset of the possible communication methods, for example the corporate PBX and the corporate-issued mobile phone, it is possible. In the real world, where people actually communicate, it’s not. There are too many ways to communicate and too many parties unwilling to open their networks to allow some unaffiliated third party to create an environment to manage all their communication.

To date, I have not seen a single unified communications “solution” – a buzzword if I ever heard it – incorporate all of the following:

• SMS (not just sending, but receiving)
• Skype
• IM–not just corporate IM, but all public network IM
• SIP, both outgoing to random SIP URLs and incoming SIP calls
• Email from multiple locations
• Social Networking (e.g. Twitter, Facebook, Jaiku)

Until it includes SMS on my mobile phone, which none of the solutions I’ve seen even attempt to deal with, it’s not truly unified. Until it includes Skype – a tool I am using more frequently – it’s not unified. Unless it includes a SIP URL that anyone with an open SIP client can reach, it’s not truly unified. Until it handles all my IM stuff, it’s not unified. Until I can get a unified view of all my email and social networking traffic, it’s not unified. [Ed bold]

It would be interesting to see if Microsoft embeds Skype into their Live Communications Server offerings. And note PhoneBoy’s positioning of Skype as a peer equivalent of SIP. On the other hand, Skype makes no attempt to position itself as a Unified Communications platform.

Note: Dameon more recently has started posting additional insights on The VoIP Weblog, a publication of the Creative Weblogging network.

Tags: PhoneBoy, PhoneBoy, Dameon Welch-Abernathy, Skype, Unified Communications

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Just over a month ago a two-day Skype outage caused great consternation with predictions of gloom and doom for Skype. Early yesterday here in California (around 1600GMT) I noticed almost 9.7 million users online — back to about the same number as peak loads immediately prior to the outage. Somebody out there is continuing to use it; for someone Skype is offering value-add.

Meanwhile Andy Abramson over at VoIP Watch wants Yahoo executives to admit “Yahoo isn’t talking.” And whither AOL’s AIM Phone Line? Om reports on Vonage: How Low Can You Go? And Matt Asay, over at CNet, writes: Swapping Vonage for Skype: One man’s search for VoIP that actually works where he starts out with:

Yes, you read the headline right. I have long been a critic of Skype, suggesting that eBay was foolish to buy the VoIP toy and generally ridiculing it as a serious business tool.

Today I’m eating crow, and it tastes great. Why? Because Vonage has been complete rubbish for me, whereas Skype is increasingly approaching perfection. I dropped my traditional phone service for Vonage. I’m now about to drop my traditionally awful Vonage for Skype.

Read Matt’s full story about how deterioration of service levels is driving away Vonage customers. (Hat tip to Andy for pointing out this story.) This morning Alec Saunders writes Walks like a telco, talks like a telco…. must be a telco where he discusses why many VoIP companies are dying when they simple try to offer lower cost versions of traditional legacy telco services. And he concludes with (my bold emphasis):

To get to free phone calls requires a fundamental change in architecture which Vonage et al have not embraced. It requires pushing the core calling functionality to the edge of the network, which implies turning off the “minute meter”. Voice, in this scenario, is nothing more than an undifferentiated stream of bits, charged at the bandwidth rate of the network operator. The profits must be made from the services surrounding the call – before and after – not during.

The SIP standard anticipates this model by allowing for both peer-to-peer calling models, and calling models which pass through a centralized proxy. While no VoIP “operator” has ever embraced the peer SIP model, Skype has delivered peer calling on their proprietary protocol. Skype understands that the money isn’t in transporting the bits, but rather [is] in all of the ancillary pieces that can be offered around that bit transport — ring tones, voice mail, phone numbers, and protocol licensing to third parties who wish to attach equipment to the Skype peer network. Similarly, by embedding conference calling in Facebook, at iotum we’re trying to create a better experience before and after the call, rather than during simply focusing on the cost of the call (although free is pretty compelling, I would argue…)

That’s the fundamental difference between the success of Skype and the failure of Vonage and SunRocket. Skype doesn’t look like a telco. Vonage, however, walks like a telco and talks like a telco…. without a telco’s deep pockets.

Not only does Skype offer an unequaled range of real time conversations services — chat, presence, video, 10-party conference calls, voice mail, SMS messaging, call transfer and file transfer; but we are also now witnessing the emergence of Skype’s Developer Partner program fully demonstrate the value of having service-oriented, ancillary offerings. Offerings that embed Skype into (business) processes such as call centers, collaboration tools, audio hosting tools, faxing, conversation archiving and CRM tools. To recall my post Skype Partners Answer Jeff’s Call for Innovation in Voice Services:

In the Skype ecosystem we can see the recipe for a foundation for innovative IP-based services.

• Start with a full real time conversation platform that combines voice, presence and text messaging.
• Start with a real time conversation platform that is enhanced with video, call transfer, call forwarding, voice messaging and file transfer.
• Start with an IP-based ecosystem that has a set of API’s to facilitate application development and mashups
• Start with a platform for which hardware has been developed to take advantage of many features of the platform.
• Start with a platform that can be accessed via not only Windows, Mac and Linux PC’s but also USB phones, PC-free phone sets, mobile phones, Blackberries (here and here) and the Nokia N800 Internet tablets.

The post goes on to list some of these Partner-generated services. And as Skype’s Partner program holds its various Developer events this month, we are seeing the beginnings of the evolution of Alec’s Voice 2.0 Manifesto. Over the past two years as Skype’s Partner services have evolved we have witnessed a withering of voice services at Yahoo, AOL and others. We have not see others recruit over 200 million account registrations (or disclose how many users are online in real time). Yet the Skype ecosystem is emerging from its cocoon, ready to fly as a major contributor to low cost worldwide real time communications services.

And the telcos don’t have the strategic horsepower to compete with Skype; their most valid strategy for competing with Skype is to partner with Skype for the services and just ship the bits.

Now if we could just figure out what’s going to evolve for Skype from the Google-eBay agreement announced just over a year ago that included this statement: “The companies will also explore interoperability between Skype and Google Talk via open standards to enable text chat and online presence.”

Tags: Skype, Yahoo, AOL AIM Phone Line, Alec Saunders, GigaOm, CNET, Matt Asay, Andy Abramson, VoIP Watch, Voice 2.0 Manifesto, Google, eBay, Skype Developer Partners

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One of the challenges of entering the VoIP consumer space is to simply sign up users; so how do you match up the fact that Skype has over 100 million registered users (even if only max 7 million, and climbing, are on Skype at any given time)?

A lot of posts yesterday about Gizmo Project’s newly announced “All Calls Free” program whereby registered GizmoProject users can make free calls from GizmoProject to either VoIP or PSTN phones, provided both parties are registered GizmoProject users. At the right is the resulting right click menu for individual contacts.

The genius in this program is the attempt to drive market awareness virally by getting all your (PC- and headset-equipped) friends and family to sign up for GizmoProject and experiment with it. You then have the option of calling them at no charge; they can receive the call on either the GizmoProject softphone or their legacy PSTN phones. However as Mark Evans states in his post, “Telco Hell“:

“The idea of free calls will no doubt appeal to the bleeding and leading edge who have no concerns about using software and computers rather than traditional telephones. For the mainstream, the Gizmo Project is probably difficult to grasp.”

Alec Saunders, in his post, Gizmo “Friends and Family” concludes:

“If you primarily call North American numbers, for now Skype is a better offering. It’s really free. It may not always be, since free calling is supposed to be a promotion. If you make lots of calls overseas, depending on where you call, Gizmo may be a better offering (if you can get your friends and family to sign up for it).”

Om Malik, in “Voice Now Nearly Free“, speculates on the business model: ”

“This free voice movement had me thinking – what kind of a loss is acceptable to these companies? Though it is hard to get a straight answer, Jajah officials say they can make up all the losses in premium services such as scheduled conference calling, or other such services.”

The limitation for GizmoProject’s “All Calls Free”, as Mark implies, is that the party initiating the call needs to be using a PC with a headset (or equivalent USB audio device). This highlights the thinking behind Skype’s decision to launch a hardware certification program which has resulted in devices that let a much broader user base of non-geek users effectively “add” Skype to their current phone services. Devices such as VoIPvoice’s Cyberphone and UConnect or Multi-Link’s TeleVoIP Stick.that provide links between traditional telephone keypads and Skype, And, of course, the forthcoming Skype for WiFi phones eliminate the need for a PC.