ConnectMeAnywhere (CMA)

London based ConnectMeAnywhere (CMA) would like you to use your mobile phone for international calls, also saving you money by using VoIP network (SIP backbone), but they can do so without you having to download any software or install any new hardware. In fact users don’t even need to connect to a computer to make a call.


What makes us different is that we allow anyone who can register to call almost anyone else from a real phone, not a pc. This means that you can call from your mobile, your land line and/or your work number. We do almost the opposite that Skype are doing with the DDIs. We don’t route inbound calls to you, we route your outbound calls to whoever you choose. We are focussing heavily on the phones and not the web to make the calls which makes the quality better (no pc or desktop software).” said Rayan Gallagher from CMA

Hullo : Voip through your landline
By S.M. Schrama

Hullo is the latest addition to Voip competitors. Where the american voip market is dominated by Vonage, the european market knows a different nr 1 : Skype. Where Skype works through a native client installed on your computer or portable device, Hullo is more like Jajah. On the website, you key in your number and the number of the person you want to call. Your phone starts ringing and voila, there's your call.

So what's the big difference that will make Hullo a true competitor ?

You can make a call from any normal phone or directly from your computer with speakers/headset and a microphone. You can also do group calling by dragging contacts into the call or adding a new phone number. If you need to switch phones, you can drop off the call and add your other number.

Hullo is also useful as a call forwarding device. They issue every user an extension on a normal phone number. When that number is called, Hullo will call your saved phone numbers in the order you tell it to, until it tracks you down.

That's nice. But will it beat the competition ? I don't think so. Where Hullo will track you down, you can take Skype with you using your mobile device. Where Skype is available for most platforms, Hullo is only available for Windows. So I can't give them a big chance.

iSkoot Mobile Network Goes Global

iSkoot, the leading service provider for mobile Internet communications solutions, today announced that its iSkoot Mobile Network, a gateway connecting cellular and PC-to-PC networks, is now offering global coverage for over 30 countries worldwide. With the newly deployed iSkoot Mobile Network, people from around the world will be able to place and receive calls with online buddies from their mobile phones without any need for PCs, special hardware, custom phones or Wi-Fi hot spots.

To take advantage of the new global iSkoot Mobile Network, users need only to download a small piece of client software to their mobile phones. The iSkootMobile phone application for Skype Software is now available for free download to your phone at go.iskoot.com and www.iskoot.com.

“We are extremely excited about extending our iSkoot Mobile Network around the world, giving the vast majority of the Skype user base mobility. Our US launch was very successful and there has been strong demand from international users for worldwide support. The Skype phenomenon is global. Skype calls are especially beneficial in making international calls and lowering costs substantially. With this announcement we will be making iSkoot available in over 30 countries with support for many additional countries coming soon. Our goal is to rapidly bring true mobility to people everywhere,” said Roy Erez, Vice President of Business Development, iSkoot.

The iSkoot Mobile Network frees Skype users from their PCs. It provides an on-ramp to the PC-PC network for cell phone users. With iSkootMobile client software on their mobile phones, users can make Skype calls to their online buddies without ever having to use their PC or a broadband connection. Once cell phone users download the iSkoot software to their regular handsets, they will be able to do everything they normally do with Skype at their PCs, but now they can take all of Skype’s functionality with them on their mobile handsets anywhere they go. iSkoot delivers true mobility, even away from Wi-Fi hot spots.

How iSkoot Works

The iSkootMobile phone application for Skype Software is available for free at www.iskoot.com and go.iskoot.com. Users can simply download the small-footprint iSkootMobile phone application to their handsets, log onto their Skype accounts, and start calling and receiving calls from their online buddies on their mobile phones. When a user selects a contact to call, the iSkootMobile software on the handset communicates with iSkoot’s carrier-class server. The server then connects the call using Skype’s Voice over Internet service so that the user only pays for air time on the mobile phone. If the user makes a SkypeOut call, it works the same way, but the user also pays SkypeOut minutes, just as if the call were made from a PC. When a user receives Skype calls on his or her handset, SkypeOut charges also apply.

About iSkoot

Headquartered in Cambridge, Mass., iSkoot is dedicated to becoming a leader in enabling mobile Internet telephony. iSkoot extends the reach of Internet telephony by allowing users to make and receive calls over the Web using any mobile phone. With iSkoot, VoIP users are no longer bound to their PCs to make and receive calls. iSkoot enables users to take advantage of Internet phone services and buddy systems to make unlimited, superior quality voice calls via next-generation peer-to-peer software from their cell phone. For more information, visit www.iskoot.com.

Fring

The fring™ Mobile VoIP solution is designed to optimise the embedded internet capabilities within the subscriber handset. The fring™ solution establishes a Peer-To-Peer VoIP connection between calling parties, enabling true VoIP sessions between fring enabled handsets and also between handsets and PCs.

fring™ is based on a unique thin-client technology that for the first time enables true VoIP over 3G networks. fring™ dynamically adapts itself to the network and handset characteristics while enabling seamless roaming VoIP on multiple networks. The dedicated three-side P2P network architecture has been developed to support near telco-grade voice quality and network efficiency.

fring™ leverages the internet connectivity traditionally used for mobile Email retrieval and web browsing to provide mobile-VoIP communications so users can talk and instant message for free!

Truphone

BY Martin Geddes


For those who know PhoneGnome, Truphone does a similar function for mobile phones instead of landlines, just 100% in software. For those who don’t know PhoneGnome, read on. This is a goodie.

The company is Truphone, a UK-based venture-backed company. They’ve done what has been anticipated for a long time, and made it (relatively) easy to turn your cell phone into a wifi phone. But with a few big twists.

Provisioning isn’t Skype-easy, but it’s pretty easy still. Send a text message, get one in return, click, install, set up WiFi access points.

You then dial as normal. Press the green button. NO SPECIAL APPLICATION UI. If the other user is on another Truphone device, you’re through. If they’re on a landline in 40-odd countries (or US/Canada cell phone), you’re also through. Note: you’re not a penny poorer. The price of PSTN calls is effectively zero now — official death of the metered minute, full report at noon. Want to call Timbuktu, a high termination fee mobile or non-geographic or premium service? Deplete your pre-paid Truphone balance. Or just pay the usurious mobile rates if you insist — instead of pressing the green button, make a charitable donation to telco shareholders using the menu.

This is fixed-mobile convergence alright, but on the user’s agenda. If this takes hold and 5% of users adopt this within say 24 months, the pricing pressure will be evil.

There are some wrinkles. It currently only works on selected Nokia E series phones, targeted at enterprise users. (These have the necessary SIP stack, CPU, flexible Symbian UI, codecs, etc.). Mass-market Nokia N-series phones are in the pipeline. No doubt other manufacturers would like to have the “free phone calls” feature, too — you can imagine it being somewhat of a disadvantage not to. (Understatement is your friend.) They’re also addressing the consumer market, because their investors want that — yet the sweet spot is probably small and medium business (who can afford the devices and will gladly bypass carrier handset distribution channels).

Speaking of which, I’ve been consulting to anonymous handset vendors on this topic. The day of reckoning is coming for the operators. The handsets are creating the incremental value, not the networks. The value of subsidy in effecting lock-down is decreasing. The price of handsets is falling, the number of SIM-free unlocked handsets being sold is rising. The operators will resist these devices entering their channels. Bad luck, folk will just buy them tax-free in the airports as they go on holiday. You can’t compete against your customers.

Not a good day for investors in mobile operators, I’d say. Truly buggered by Truphone. Oh, and did I mention that their vision involves eroding some of those juicy SMS margins, and deploying features like voicemail to email that the operators refuse to roll out? No wonder they all look rather pleased with themselves.

Raketu new P2P VoIP app takes on Skype
By Tom keating

Is Raketu the next Skype killer? Raketu today launched a new VoIP client that also offers information and entertainment services. Raketu’s communications features include dialout calling (rakOut) to landline/mobile phones, Instant Messaging (supporting Raketu, Yahoo, MSN, AOL, ICQ, Google and Skype), SMS-text messaging, and file transfers/sharing.

Raketu's information features include news, sports, weather, stock feeds, and an advanced internet and travel searching facility. Raketu’s entertainment features include a podcast reader/player, games, and a full featured multi-media player with karaoke. Raketu supports click-to-call, click-to-im, click-to-sms, global online presence, and enhanced social networking features. I even noticed the software supports plugins, including games such as the classic Battleships game. I guess this would be Web 2.0 meets VoIP 2.0. Still missing some of my "cool" features for the perfect unified communications client.

In any event, Raketu’s peer-to-peer (p2p) technology allows high quality VoIP calling and they claim the highest call-completion, without the security issues associated with supernodes (i.e. Skype) and other traditional p2p technologies. And unlike other p2p communications providers, Raketu does not use your computer for other users’ communications.


According to Raketu, "More exciting services are planned for the coming months when Raketu will launch in-bound calling, advanced voice mail, large conferencing, and more. Today, Raketu provides the most comprehensive integrated communications, information and entertainment tool available, providing more options, more control, and more personal activity features than any other service."

Raketu runs over any internet connection including dialup.

Rebtel

By Aswath


How Rebtel works: For $1 a week, you get two numbers, one local to your calling area and another local to your designated buddy. When one of you want to call the other, that person will dial their local number allocated by Rebtel; in turn Rebtel will ring the other person's phone number. The called party can hangup within 30 seconds and dial their local access number allocated by Rebtel to be connected to the buddy and they can talk for free as long as they desire. One can easily infer the business model. Some time back Alec had pointed out that it costs about $1 per month to rent a telephone number. Since Rebtel's charge amounts to $4 a month for two phone numbers, there is a clear profit in just allocating the phone numbers. But the beuaty of the scheme is in recognizing that the two legs are incoming calls to Rebtel, unlike Jajah where it makes two outgoing calls. This way, Rebtel not only incurs any charges but can look forward to additional revenue in the form of settlement charges.

But Jeff had suggested (http://pulverblog.pulver.com/archives/005238.html)after a trip that Rebtel is more than the obvious “mobile arbitrage pay”. He doesn't give any more clues – to be more accurate I didn't pick anything else, till this morning. Andy(http://andyabramson.blogs.com/voipwatch/2006/09/rebtel_raises_a.html) points to a storyin B2.0(http://business2.blogs.com/business2blog/2006/09/with_a_rebtel_y.html)that claims that Rebtel is funded to the tune of $20 millon. Of course, they do not require this kind of money to execute the publically announced service. It is for those that are “going on under the hood that meets the eye”, as Jeff puts it. So the question is what could they be. Andy quotes his pal Sanjay jaawar of Bridgeport Networks. So this must be something to do with IMS.

Now for some nitpicks. I don't agree with Andy's characterization that Rebtel is like Mint Telecom. As far as I understand, Mint makes an outgoing call to the far end – it is not made up of two incoming calls. So they have to levy a per minute charge to their customers. Also Rebtel can further simplfy the user experience and extend the service. But I don't think that is what they are focusing on right now.


The reason for my trip to Stockholm: Rebtel
Jeff Pulver

Friends of mine introduced me to Rebtel a couple of weeks ago and after hearing about who they were and what they were up to, I decided to fly to Stockholm and spend the day with them and get to know the company from the inside out.

Rebtel is one of the most exciting startups that I have met in 2006 and they have the potential to become bigger than Skype. While on the surface they may appear to be a “mobile arbitrage play”, there is a lot more going on under the hood that meets the eye. The team of people at Rebtel are both revolutionaries and visionaries and they fully understand the value of “voice as an application” and underlying power of SIP.

From their perspective, “Communication is not between desks and desktops. It’s between people. And peoples’ preferred way to communicate is their mobile phone – their connectivity to the world.”

When I asked their CEO Hjalmar Winblah to describe the immediate opportunity that they going after he said: “The world’s mobile operators are going quickly to big bucket plans because they have to compete. If we do this, mobile operators say, we can defend our ARPU and achieve fixed to mobile convergence – get people to abandon their fixed line phones – and move out the long tail. It’s a defend-and-extend strategy – and that’s what Rebtel is going to leverage with our services that go directly after their remaining margins: international calls and later roaming.”

While Skype is all about peer-to-peer, Skype generally requires the use of a computer, broadband over WiFi or Ethernet, and an operating system that allows you to run third party software. Skype, with its amazing success on PCs, had intentions to go mobile. But it is the wrong technology for it; and the wrong business model. Since the launch of Skype, their success is legendary in the world of IP Communications and Skype showed the world that they can execute and as a reward they were acquired by eBay. In contrast, Rebtel is a different play and a much different bet. Rebtel’s current go-to-market strategy is to leverage what’s out there today: 2 billion standard mobile phones that consumers know how to use. No special downloads required. The catch? leverage “bucket pricing plans“ by getting consumer to place local phone calls in order to speak with friends and family members across national boarders. While “distance is dead” in the world of the Internet, Rebtel has removed distance from the consumers of mobile phones.

Skype leveraged the PC and broadband. Similarly, Rebtel is leveraging mobile phones and operators’ local minute bucket plans. And they’ve built a 100 percent SIP infrastructure prepared for the future evolution of mobile/wireless communications.

The team at Rebtel is trying to build something that lasts; something that drives fundament change. Just like their wireline cousins, mobile operators are best positioned to be wireless access providers. Wireless operators should not be able to claim ownership of someone or their devices. They should be happy just being access providers.

However, these days most mobile operators look at themselves as more than access providers and look to milk their customers for as much money as possible every day. It could be said that some mobile operators believe they have the right to maximize their customer relationships and nickel and dime people..because they can. While mobile operators have benefited greatly during the past six years due to “wireless conversion” they have failed in everything but one thing: getting people to pay too much to talk to their friends.

Look for Rebtel to be a major change agent here. Rebtel is in the business of liberating people from their forced relationship with their carriers. Let the revolution begin!

DEMO: Jajah debuts simple mobile VoIP

You've probably heard of Jajah, one of about a zillion free/cheap VoIP carriers. From the start, Jajah's twist was using the Web to set up calls: customers would go to a Web site, enter their number and the destination number, both phones would ring and the Jajah would act as a bridge. Well, today, at the famed and exclusive DEMO conference, Jajah is debuting a wireless VoIP play that essentially is the same thing on your cell phone, but pretty much without the browser. Symbian phones will be able to set up Jajah calls natively; phones with Java will need a small downloaded app; those with only a browser will be able to use Jajah that way, and other phones will be able to set up calls through SMS. At the desktop, Jajah can interoperate with Outlook or the Mac Address Book and can easily set up large conference bridges. As of today, Jajah pretty much steps to the front of the pack.


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Jajah, the Internet voice company that helped pioneer the mobile VoIP communications market, on Tuesday announced a mobile suite that allows users to make free or inexpensive long-distance calls from their cell phones.

The company, which launched in February and moved its offices from Austria to Mountain View, California, surprised the telecommunications world by announcing a series of products that freed VoIP from its PC-centric hold.

The company’s newest mobile service allows callers to use their cell phones to make inexpensive VoIP calls without involving a PC in the process.

Jajah said it was working on a mobile product back in April, at a time when few companies were talking about rerouting mobile calls via the Internet (see Jajah Dials Cell Phones). The business of rerouting mobile calls via the Internet has since exploded.

Mobile VoIP Explosion

A number of startups, including Rebtel, MINO, and Switch-Mobile, have announced products that use varied means to route mobile calls via the Internet. Rebtel received $20 million in a first round of VC funding Tuesday (see Rebtel Snags $20M First Round).

Even traditional phone companies are either working on or are already offering products that allow users to switch their mobile phones to IP when they are in reach of a broadband connection via Wi-Fi technology.

British Telecom has its Fusion service for consumers and Corporate Fusion for its business customers, allowing both groups to save money by routing mobile calls via broadband connections (see BT Dials Phone Convergence).

Jajah’s system works like this: first, users go to Jajah’s web site and choose one of four software options that would be appropriate for their type of mobile phone. The software and instructions are then sent via text message to the user’s phone.

The software program, which supports Symbian and browser systems among others, integrates itself into the mobile phone’s operating system.

When users dial a number, say an international number, the user hits a green button and the call is rerouted via the Internet.

“The default setting is that all international calls are Jajah calls and all local calls go the normal way, but users can change the setting to their personal needs,” said Jajah co-founder Roman Scharf.

“The only difference between a normal mobile call and a Jajah call is the user hears an announcement saying ‘please hold while Jajah connects your call,’” Mr. Scharf explained.

International calling rates from mobile phones in the United States have remained particularly high. The average is about $0.70 per minute, and can reach as high as $2.00 per minute.

Jajah calls to Europe are $0.02 per minute, but for registered Jajah users calling each other, the cost of the call is free.

Jajah Mask

Jajah has made a concerted effort to mask the complexity of its underlying system. While the audio message is playing, the phone dials Jajah and hangs up.

Lawsuits Perhaps?

Since the announcement, a number of firms have gone a similar route. Some use the mobile connection to work around the operating system.

Others spoof the mobile network, and still others use the mobile data service to route mobile calls via the Internet.

“This proves that we were on the right track because people don’t copy bad ideas,” said Mr. Scharf.

“But some are imitations that are infringing our patents,” he added. “We will look at this for a few more months and we will come to a decision on whether we will take action.”

Mr. Scharf said that many of Jajah’s competitors have come up with different permutations on Jajah’s idea, but he does not see any real competitors among the independent startups.

“Rebtel came up with a different angle and that’s good… It means the market is growing in diversity,” he said.

“Rebtel’s service is interesting but it is too complicated,” he added. “They have to improve on how they communicate and not add more complex features at this stage.”

The system calls back and the phone answers. That entire process is masked by Jajah’s “please hold…” message.

“The killer aspect here is that it does not need a data connection so it cannot be obstructed by the mobile carrier,” said Mr. Scharf.

In February, Jajah surprised the industry by announcing a series of products that freed the VoIP market from its broadband hold. Unlike Skype and Vonage, Jajah was not tied to the broadband connection. Someone with a dial-up connection could use the service.

Skype banned on California campuses...

Skype's got lots of fans, but network admins are not among them. The trouble is, Skype is a pretty pure peer-to-peer software implementation, which means it steals (excuse me, "borrows") bandwidth from every computer it runs on--and does it in a proprietary and hard-to-detect way. Admins hate "proprietary and hard-to-detect" traffic on their networks. A number of businesses have banned Skype, and now some universities are following suit. San Jose State University is the latest California campus to ban Skype, following UC Santa Barbara and Cal State Dominguez Hills. It might not be unreasonable to argue that the schools are just protecting their lucrative telecom franchises, except all three campuses apparently are still allowing non-grid VoIP apps like Gizmo free access.

read this article on Ars Technica
http://arstechnica.com/news.ars/post/20060924-7814.html

Skype Moves Towards a Secure VoIP Software for Corporate World


The general perception on Skype is that it disrupts corporate networking. In its new mission, Skype is releasing a beta version of its software to provide secure and enterprise-friendly VoIP related services.

This report says that the redesigned Skype for enterprises would empower systems administrators to use standard Windows management tools to set how the Skype software connects to the Internet, or to disable any of half a dozen functions, including file transfers.

As noted earlier, a quarter of the Skype customers use the Skype services for business purposes despite the fear that the p2p technology is hard to block connection protocols. It causes security problems in some business.

Michael Jackson, Vice President of telecommunications and Skype for business said that there was a rumor that we disrupt networks to get around things.

He added,

Because we design things for consumers so they work in any network environment. The back end of that is, it works in any network environment.” But that makes it difficult for companies to block the software.

With a view to develop enterprise-friendly software, Skype has teamed up with Intel. They would bring out a proxy server approach that would enable the companies to cut off the software’s network access if a security problem is traced out.

BlueNote Marries VOIP and SOA

BlueNote Networks on Sept. 25 introduced a pair of new software offerings that marry voice over IP with service-oriented architectures.

The startup's aim in the new SessionSuite SOA Edition and SessionSuite Desktop products is to allow voice to be embedded within SOA applications running on IP networks.

"SOA brings to an IT organization a distributed application framework that allows them to develop a suite of software services that can be reused and shared among complementary business applications, instead of building monolithic business applications.

"There are easier ways to embed voice into business applications than using [Computer Telephony Integration] interfaces from monolithic PBXes," said Sally Bament, vice president of marketing at BlueNote Networks in Tewksbury, Mass.

The two programs allow BlueNote to expose to developers the ability to make a phone call as a Web services application program interface, and at the same time allow other business applications to affect how to handle a phone call.

For example, a CRM application could initiate a phone call, route incoming phone calls to a user's mobile phone, or make a Web services request to another business application like a calendar and based on the result determine how to route an incoming call.


Another benefit in embedding voice services into applications is the ability to streamline the interaction of Web-shopping users and call center agents.

From a click-to-call button on a Web site, a call can come into the Session Suite software, which tags the call with contextual information from the user's activity on the Web site.

It then forwards the call to an agent while populating that agent's screen with the information on the user's shopping activity.

Such information can include the Web page the user was on when the user made the call or shopping cart information.

"We're keen on a lot of the functions that API provides us. I believe it'll be a big hit for us next year," said early customer Mark Ustik, CEO at The Hudson Group in Boston.

"Our customers are clamoring for some of the integration features, the biggest being integrated call detail reporting," added Ustik, whose company markets hosted applications for the ground transportation industry.

Ustik, who believes SessionSuite SOA Edition is unique in the completeness of its functions, sees the new software as a means to drive additional services revenues, and he believes his customers will benefit by being able to reduce the number of call center agents they must have on staff.

The software implements a call manager function, is a full Session Initiation Protocol server, and provides registration proxy, call forwarding, call transfer, integrated voice response, Quality of Service prioritization and more.

It can also interface with existing PBXes, and it provides Network Address Translation and firewall traversing tools.

BlueNote is targeting the new SessionSuite SOA Edition at business analysts, application developers and independent software vendors (ISVs) to allow them to add Session Initiation Protocol-based voice and video services to applications, Web sites or business processes.

SessionSuite SOA Edition's Software Development Kit hides the underlying communications infrastructure complexity from the developer by exposing the communications services through Web services APIs.

Within the SDK is a session lifecycle API that allows an application to initiate a phone call, control it, terminate it and tag data with the phone call.

The session manage API provides access to provisioning information for users, calls or services in the system. A session plug-in framework allows an application to impact how to route a call.

SessionSuite Desktop is a companion program to SessionSuite SOA or BlueNote's existing SessionSuite Enterprise that allows users to add voice functions to Windows clients.

It allows end users to make calls from within Windows applications using a single mouse click.

It also provides companion phone control that allows users to have a choice of which handsets they want to use, whether it's a VOIP phone or mobile phone.

"We separated the control of a phone call from the media – the actual speaking – so that users can continue to use their phone of choice and control the operation from their PC," said Bament.

SIP (Session Initiation Protocol) has certainly gained momentum, and it doesn’t look to be slowing down anytime soon.

By Erik Lagerway


In the past year we have seen the major IM providers like Yahoo! and MSN adopting SIP as their chosen signaling protocol for VoIP. Google has openly professed that they intend to support SIP in their new IM service. Even Apple’s iChat supports SIP for VoIP and video. SIP is the torch lighting the way for VoIP; there is no doubt about it. The thrust seems almost unstoppable with everyone racing to support open standards. For consumers, this is a very good thing as it presents choices. Networks that interoperate are very important and, if providers want any hope of peering with other networks, they will need to use open standards. SIP does all of this and more. So, what, if, anything could slow the adoption of SIP? Firewall traversal? No, we have pretty much solved that one with use of STUN, TURN, ICE, and some intelligence built into the SIP end point. What about Security? Yes, this could be a problem in certain networks that choose to ignore the issues. SPIT — SPAM for Internet Telephony — is not yet a huge problem, but it has had an impact on some providers who found themselves shutting down certain services because of it. You don’t hear much about it because the providers would rather not admit their networks are vulnerable. So what are the main threats? From a consumer’s point of view, there are two main concerns: protecting the identity of the user and protecting the content of the calls.



Identity Theft

In one case, a large VoIP provider in the U.S. had enormous problems with users spoofing other SIP IDs and making calls to sex lines in another country. The local carrier was appalled when the peering provider sent them their bill. This resulted in blocking service to that particular country. Today, most SIP-enabled devices and registrars support Digest Authentication and TLS. Together, these methodologies can be used to thwart ID theft in a SIP network. It’s too bad that TLS is not widely supported when connecting from SIP to the PSTN (Public Service Telephone Network), making it difficult to implement across the board. S/MIME (Secure MIME — Multipurpose Internet Mail Extensions) is another mechanism used to encrypt individual messages, much like PGP (Pretty Good Privacy). It was primarily developed for e-mail, but it certainly can be applied to SIP for use with VoIP and IM. What about DoS attacks? A few months back, another U.S. provider was hit with a DoS (Denial of Service) attack that consisted of many SIP requests overwhelming the network and castrating the service. The provider had to interrupt service in order to fix the problem, cutting off thousands of users in the process. Often in a VoIP network, a user does not know the person who is calling them and may want to decide whether or not they want to take the call before answering. The IETF is close to finalizing a new security method for VoIP, called “SIP Cert,” which is meant to provide a secure way to make sure that callers are, in fact, who they say they are.

Securing the Media — Encryption

Eavesdropping is an issue not only in VoIP, but also in traditional telephony. To think that someone could overhear your conversations is disconcerting to say the least. Today, VoIP networks using SIP generally use RTP (Real-time Transport Protocol) for the actual voice and video that we see and hear during a call. Encrypting this data can be done by implementing Secure RTP, or SRTP. Since SRTP was built to be very efficient, it uses up little additional bandwidth and CPU. The bottom line is that SIP is the best standards-based protocol for building VoIP and Video services on the Internet today. The SIP community is rich with IP Communications leaders and the support from this network is tremendous. This protocol will could very well mature into a complete IP Communications framework. The battle for VoIP has been won; now what about IM? Stay tuned as SIP wages war on closed IM protocols and takes on the enterprise with SIMPLE (SIP for Instant Messaging using Presence Leveraging Extensions).

The Facts about SIP
By Interactive Intelligence

After years of refinement, the international SIP standard continues to become more integral to the movement toward converged voice and data communications. Unfortunately, some legacy vendors are still questioning various aspects of the open SIP standard. Here are a few of the more common objections going around, and the Interactive Intelligence responses to them.


MYTH: SIP offers only basic functionality and needs further development before companies deploy it.

FACT: This objection represents a fundamental misunderstanding of SIP and its role in
communications. As its name makes clear, Session Initiation Protocol is simply a protocol, not an application. Interactive Intelligence products are mature application suites that have been developed and enhanced since 1996. They make use of SIP for a very circumscribed part of their operation — establishing audio and video sessions between various IP end points, such as IP phones. SIP provides everything needed for this function.

In fact, SIP provides everything that legacy protocols such as ISDN provided, but in a much simpler, text-based formulation that is well designed for IP networks. As software applications, Interactive Intelligence products including the Customer Interaction Center® (CIC), Enterprise Interaction Center® (EIC), and Communité™ provide rich application functionality for automatic call distribution, multimedia queuing, unified communications, and other advanced interaction management features.


Sources voicing this “further development” objection likely have viewed immature products designed exclusively for SIP, and incorrectly blame functional limitations on the protocol rather than the application.

MYTH: SIP lacks interoperability.

FACT: This is an odd claim to make when vendors such as Interactive Intelligence are
installing heterogeneous systems combining SIP phones from Cisco® and Polycom® along with SIP gateways from AudioCodes®, Cisco, etc. Interactive Intelligence has tested its SIP-based communications system with SIP voice trunks from MCI® and SIP proxy servers from companies including Microsoft® and Cisco.

Trade shows such as VON also regularly feature various interoperability demonstrations. Because of its relative simplicity, interoperability is much easier with SIP than with many older protocols such as ISDN that are infamous for compatibility problems.

Perhaps this misguided claim arises in reaction to product offerings from Cisco, Avaya, and others in which vendors insist that customers install only vendor-provided components. Such a situation represents an attempt by vendors to lock customers in, and is certainly not an inherent fault in the SIP protocol.


MYTH: SIP introduces new demands on networks and raises security concerns.

FACT: Actually, it is voice over IP (VoIP) itself that places new demands on data networks and introduces new security concerns. Such issues stem from routing a new type of traffic (voice) over the network and connecting new end point devices (phones) to the network. These same issues must be faced regardless of whether the protocol used to set up VoIP calls is H.323, SIP, or something else.


Increasingly, however, network equipment is taking account of the momentum of SIP. Almost all newer routers and switches support quality of service for prioritization of voice traffic. With QoS and a well-designed switched network, any LAN can be used to reliably handle VoIP.

SIP also has added mechanisms for authentication and encryption to greatly improve security. Interactive Intelligence fully supports the RFCs dealing with security, and the products from Interactive Intelligence have recently passed with flying colors the DOS (denial of service)attack tests performed by Miercom. SIP authentication prevents hijacking, redirection, and man-in-the-middle attacks. Encryption is mainly being done at the transport layer with industry standards such as IPSec and TLS (transport layer security – successor to SSL). Such encryption prevents eavesdropping, tampering, and message forgery.

With properly implemented security, SIP-based VoIP is actually more secure than traditional telephony, where anyone with a butt set could listen to calls!


THE INTERACTIVE INTELLIGENCE OPINION

In short, SIP is being increasingly regarded as a well-designed and mature foundation for voice over IP. This is why it has been adopted by every major vendor including Microsoft, Siemens and many others, and why SIP is recognized as an international standard. With mature, feature-rich interaction management products such as those from Interactive Intelligence, SIP is indeed ready for prime-time.

The evolution of SIP- and IMS-capable mobile handsets

By Dean Bubley Disruptive Analysis




Disruptive Analysis’ new report, “The evolution of SIP- and IMS-capable mobile handsets” is the most comprehensive review of the market for next-generation IP-connected cellphones, offering detailed analysis and recommendations.

The noise level around IMS is huge. But up until now, almost all attention has been focused on IMS network deployments and SIP applications. The phone itself - the "User Equipment" in IMS industry jargon - has almost been ignored. Once again, the telecom industry seems to have under-estimated the complexity of getting the phones "right" before investing billions on new infrastructure. Not only that, but the gap is likely to be filled by "open" or "naked" SIP-enabled mobile phones, which will enable 3rd-party providers - such as Internet VoIP and IM specialists - to exploit a huge mobile user base with their own on-handset software applications.

This ground-breaking study provides extensive argument and rich quantitative forecasts of the future of IMS- and SIP-enabled mobile phones. It assesses operator and "disruptive" usage cases, looks at a wide range of handset types and software architectures, and considers practical issues around standards, user experience and integration. It profiles the leading suppliers of IMS and SIP handset software.

The market size and segmentation data covers handset type (eg smartphone vs featurephone), SIP/IMS implementation (eg "Naked SIP", "Closed IMS" etc.), regional market sizes and wireless bearers (cellular, WiFi, UMTS, CDMA, WiMAX etc).

Based on a huge research effort spanning 300+ interviews and meetings with key providers of handsets, software, network services, semiconductors, operator infrastructure and corporate networks, the study provides actionable, substantiated insights into the "End User Battleground" for SIP and IMS services - the handset itself.


Highlights

Although many operators are deploying IMS networks now, it is highly likely that IMS phones will be late-to-market, and suffer from poor user experience.

There is little consensus on the answer to “What exactly is an IMS phone?”

Existing standards are too protocol-focused and are insufficient to define how IMS phones should “behave”, e.g. how IMS & non-IMS applications interact.

Handsets with basic IMS capabilities (often operator-proprietary) will start to ship in small quantities in late 2006 and 2007, although it will be 2009 before 20%+ massmarket penetration is reached, with more standardised handsets.

Concepts of handset-based “combinational services” and downloadable IMS applications are not yet practical. User interface design and interoperability between multiple vendors’ phones require much more effort & development.

A good IMS user experience will need handsets capable of full multi-tasking – something which is outside the capabilities of most current phones.

These problems should be overcome eventually. In 2011, it is forecast that there will be almost 500m IMS-capable phones shipped globally.

Despite hype around fixed-mobile convergence, WiFi will be present in less than 10% of IMS-capable mobile phones by 2011; most will be cellular-only.

SIP – an essential basic subcomponent of IMS – is much easier to implement than a full IMS software framework. SIP-capable phones are already shipping.

There are many interesting non-IMS applications of SIP on mobile phones, such as VoIP, Internet IM, enterprise IP-PBX access, or interactive games.

In total, 787m SIP-enabled mobile handsets will ship in 2011, of which 40% will be smartphones. Europe will account for 50%+ of SIP handset volume shipments until 2010, although Japan and Korea lead, in penetration terms.

SIP will be adopted more slowly on CDMA handsets than GSM/UMTS ones.

“Naked SIP” phones, on which 3rd-party applications can exploit the SIP stack, will grow rapidly in importance, with 48m shipping in 2006, more than 220m in 2008 and 500m+ in 2011. This is a huge threat to mobile operators.

Naked SIP will be enabled by smartphones OS’s, virtual machines like Java, and the inclusion of “exposed” SIP in many featurephone platforms.

Although some devices will support both naked SIP and operator-oriented IMS applications, there will be 1 billion more naked SIP handsets shipped, than operator-only “closed IMS” phones, between 2006-2011.

Internet brands, enterprises, 3rd-party developers and competing service providers will exploit the opportunities from the 1.6 billion “naked SIP” phones that will ship between now and 2011, using on-handset software clients.

Some operators will attempt to block “parasitic” 3rd-party SIP applications, by “locking” handsets or intercepting network traffic. These attempts will seem clumsy and vindictive, and will likely drive churn and customer disloyalty.

Mobile operators’ visions of stopping threats from Internet-based brands, 3rd-party VoIP and other services, by locking-down IMS networks and only permitting customers to access IMS-based services, are totally unrealistic.

A large number of specialist firms are emerging to supply SIP and IMS software for mobile phones. However, many will struggle with integration of their clients into existing OS and application software on handsets.

SIP-capable mobiles - minimising fragmentation
By Dean Bubley


A couple of conversations I've had recently have warned me that SIP-capable mobile phones may already be starting to suffer from the interoperability & fragmentation problems that plague other handset software areas.

My understanding is that the implementation of SIP on phones at present is haphazard, with performance (and conformance) highly variable. While there are some moves to harmonise this, it seems likely that this will pose problems for developers in the short term. This may give some operators a respite from the threat of competitive 3rd party applications exploiting "Naked SIP" - but may also hamper their own efforts to deploy new applications in advance of "full IMS" becoming a reality.

It's also driving a number of the competitive mobile app guys to use their own proprietary protocols instead, rather than SIP. Obviously Skype is the most apparent, but I've met others in the Mobile VoIP space that are also going that route. (This also has the benefit of avoiding any nonsense with operators' SIP proxies or SBCs playing silly games with external SIP sessions, I guess). Some others use their own SIP stack within their applications, irrespective of whether there's another one in the OS.

Bottom line is that "it's not that easy" (is it ever?) to develop SIP-based applications or services that will work across a broad range of phones.

Now, the other market where this occurs is Java, where there has been huge fragmentation in the implementation of J2ME on phones - different extensions, different access to the underlying phone capabilities, different performance and so on. There's already a cottage industry of companies like Tira Wireless that offer porting solutions, taking a lot of the work out of customising device-specific variants of software. Others do testing, or other tools to help.

It strikes me that there are quite a few opportunities for doing the same with SIP. At the very least, a company with a database or a simulation tool of different phones' SIP implementation would be valuable. A porting tool would be more difficult given the range of OS's, but it ought to be possible to go part of the way.

Obviously, it would be best to get a standard way of doing handset SIP - but I suspect that this is likely to be operator-centric if and when it occurs. In the meantime, the best solution might be to assume fragmentation will occur, and look as soon as possible at creating ways of easing the pain.

Flash In The VoIP Pan


Adobe Systems, the San Jose, California-based software giant, has been the real catalyst for the ongoing online video boom, thanks to the near ubiquitous Flash software that plays back everything from stupid pet tricks to the amazing theatrics of LonelyGirl15. YouTube and hundreds of online video sites are using the Flash software to build businesses, some valued at over $1.5 billion. (Adobe acquired Macromedia, maker of Flash technologies in April 2005.)

And now Adobe Systems wants to replicate its success in video space in the Voice over the Internet (VoIP) arena, making it easy to embed voice into web applications. GigaOM has learnt of a secret start-up project currently being incubated by the $1.9 billion in annual sales software giant. Some members of this startup come from the Macromedia Breeze (now called Acrobat Connect Professional) conferencing group. (Breeze is a Flash based web-conferencing system, much like WebEx.) Though less than a year old, the start-up has started to attract some serious VoIP talent.

Sources say, Dr. Henry Sinnreich, generally known as “The Godfather” of SIP (the Session Initiation Protocol) is helping the team, though we have no details about his role within the project. He was most recently the chief technology officer of Jeff Pulver’s VoIP greenhouse, Pulver.com, and prior to that worked for MCI.

The Adobe start-up team faces quite a few challenges. For instance, it would have to support multiple VoIP protocols, and it will also have to figure out how to keep the overall size of the Flash client size small. Sources say that touching Flash Player is like messing with God inside Adobe, and the start-up team needs to figure out how to embed a SIP stack inside the player without making it bloated.

The charter for the start-up is to enhance “Flash” and add support for various voice-over-IP protocols including SIP. They have to come up with ways to make Flash-based-voice work with some of the commonly used signaling systems. These are huge challenges, but if they can overcome all these issues, they could be onto something big. For starters, they could enable web based calling, and prevent the technical hell that comes with many soft phones of today.

If they can make the technology work with the Mobile version of Flash, then the Internet-enabled smart phones can be used to initiate and terminate calls via the mobile browser or special Flash-lite based apps. But these are the most obvious use-case scenarios. Flash Games with VoIP could be another use case scenario. It could be the first step in giving web developers (Flash experts, at least) ability to add voice to whatever mash-ups they can dream off. SIP, XMPP, Jabber and Flash – put them in the blender and you could see some magic.

The gulf between the voice geeks and web developers is one of the biggest challenges facing the Voice 2.0. The vibrancy of Flash developer community, and open source projects such as Asterisk could become a new font of “Voice 2.0” innovation

The possibilities of Flash with VoIP built in can be seen in a new web-based app called Pronto, which has integrated VoIP, messaging, email, calendaring, and contact management. Communigate has just launched this service. It is not the first VoIP application to leverage Flash. Many of you might remember Gtalkr, a Flash-based Google Talk client, handiwork of Carr brothers that was acquired by Google last year before it got to show us its true voice potential.

Adobe Flash Goes VoIP
By Tom Keating Chief Technology Officer and Executive Editor


Om broke the news about Adobe’s secret VoIP start-up project. I knew about Adobe’s top-secret VoIP plans since June of this year. I interviewed Bhanu Sharma, Entrepreneur in Residence, back in June and got some interesting insights into Adobe’s plans for adding VoIP to their Flash player and how it could impact social networking sites, such as Myspace.com. Bhanu invited me to be an advisor to Adobe due to my experience in the VoIP industry to discuss the architecture and plans for what they are working on – but under NDA. Thus, they asked me to keep quiet about their plans. Now that Om broke the news, I can reveal what Adobe is up to in the VoIP space.

First, I should point out that the Flash player has had VoIP capabilities since March 2002 and the live video capabilities are activated primarily by the existence of a server in the middle called Flash Media Server making it a client-server solution. It uses H.263 codec from Spark made by Sorenson. Many social networking sites and video sites are already using the Flash player, including Myspace.com, YouTube, MSNBC, and many more. Unlike Microsoft Media Player, Macromedia Flash has better cross-platform support, which is why YouTube, MSNBC, and many other sites use Flash. In fact, MSNBC recently switched from only supporting Media Player to now supporting Flash and Media Player, which enabled Windows Firefox users as well as Linux and Mac users to watch videos.

Here is my interview with Bhanu from June 2006, which gives some really interesting insights to what Adobe is up to and how this could have a huge impact on the VoIP industry:

Tom: Is it easy to add VoIP to your existing client? Is your code modular?

Bhanu: We’ve done phenomenal engineering to keep the file size small. We've had it for the last 4 years as you know, so our VoIP capabilities need an upgrade. Back in 2002 VoIP wasn't happening as much as it is happening today. So we definitely want to update the Flash player and our other capabilities including Reader and other clients to make sure realtime communications becomes part of suite of products. So our developers can use Flash and other products to build all sorts of interesting things on top.”

Tom: With this client would you ever thing about becoming an ITSP (Internet telephony service provider)

Bhanu: There is no intent to be a phone service provider - rather we just want to be a platform where service providers build all sort of clever applications. As long as the inheritance capabilities on the player and the platform are as good as any softphone and developing a workflow and a graphical interface & services and apps - which is where hopefully they will make money some day, instead of PSTN replication, which is what the whole industry is doing right now.

Tom: Are you supporting SIP?

Bhanu: Currently we are only supporting H.263. In the future we’d like to support things like SIP. As you know we’ve brought Dr. Henry Sinnreich, onboard.

Tom: Yup, the father of SIP

Bhanu: Henry works on my team and he’s definitely going to be helping us on understanding how SIP applies. So we’re definitely serious about this space which is why we’re putting this rock solid team together

Tom: You mention the video uses a client-server architecture. Does this mean your future VoIP plans also require a server?

Bhanu: You know, frankly I don’t know what value a server will brings long-term in IP communications. That’s the debate we’ve always had – there’s other ways to monetize. Frankly, we’ll never charge for the Flash player, so we have to sell something as a company to make money and feed our engineers. Thus we sold tools, servers, and boxes. Is that the right model moving forward? Who knows? If technically you don’t need a server, why would you push a server down the throat of your customers? We’re not the kind of company that is going to try and make a monopoly out of it. So our customers insist on no server capabilities and essentially just need the client.

Tom: So if it’s just a client, if you go to Myspace.com, how are you handling the negotiating of their username or IP address if there is no central registrar. Are you doing some sort of P2P technology?

Bhanu: We’re obviously thinking of that, but I won’t say at this stage we’ve made a decision on that. There is no standards way of doing it unfortunately, which is why you see proprietary implementations that have done a good job and kudos to them for doing that.

Tom: (probably a reference to Skype (News - Alert))



Just imagine the impact of this “future” Flash player having VoIP and video capabilities. I can envision a user going to a Myspace blog, seeing who else is “in the room,” and then initiate a 2-way, 3-way, 10-way, etc. audio/video conference using the omnipresent Flash player. All the Myspace user has to do is add a small piece of HTML code to their Myspace blog to enable this and the user has to have the Flash player. Of course, the beauty of the Flash player is that it is a very small client, which is why it is very popular with techies that hate bloatware, and of course the video/audio performance is rock solid. It will be a tricky engineering feat to add a SIP stack, presence information, P2P autodiscovery, and other pieces of code to enable you to have a seamless audio/video conferencing experience simply by visiting a website. But if anyone can do it, Adobe can. So keep your eye on Adobe.