SIP - Enriching the future communications

Sip9 & Humanity

2011-10-22T05:47:00.001+02:00

Making bridges between People.

Session delivery networks

2011-08-26T08:57:00.002+02:00

Web based SIP

2011-08-11T22:15:00.003+02:00

http://www.doddlephone.com/
http://mizu-voip.com
http://sippy.com
http://java.net/projects/jain-sip-applet-phone/
http://mjsip.org
http://www.pernau.at/kd/voip/bookmarks-sip-stacks.html

Offering Free VoIP Calls to Haiti

2010-01-24T14:16:00.004+01:00

T-Mobile isn’t the only contributor in the Haiti situation. Google, Vonage, Skype are offering Free VoIP Calls to Haiti. Ricky sent us in an interesting post on the Google Voice Blog this morning. It looks like Google is doing its part as well providing free calls to Haiti for the next two weeks using Google Voice.
“It’s impossible to watch the aftermath of the Haiti earthquake without wondering how one can contribute in helping the thousands of families who lost everything in this disaster.
Google set up a disaster relief page, which includes information and resources for anyone interested in helping out, and the Google Voice team also wanted to respond in our own way.
Particularly in the wake of such a catastrophe, for people in the US who have family in Haiti, it is critical to stay in constant contact with your loved ones, provide moral comfort, and offer daily updates to friends and family in Haiti.

To help those families, we’re offering free calling to Haiti through Google Voice for the next two weeks. To place a call using Google Voice, use the Click2Call button on the website, the Google Voice mobile app, or dial your own Google Voice number and press 2 to place an outbound call.”
Now you don’t need to have a BlackBerry or other phone compatible with the mobile application. The services can be used online if you follow the given instructions on how to make a call. If you need to get in touch with someone in Haiti, feel free to use Google Voice. Don’t forget T-Mobile’s offer as well where you can donate money as well by simply sending out a text message. In addition, Vonage, Skype are also offering free 10minutes call to Haiti

Head to GoogleVoice's blog, Vonage and Skype as below for more information

GoogleVoice'blog
Vonage's twitter
Skype's sharing

Verizon Business Expands Unified Communications and Collaboration Consulting Services

2010-01-23T19:48:00.002+01:00

Verizon Business is expanding its suite of UC&C Consulting Services to help enterprise customers more effectively integrate unified communications and collaboration with their business processes.

As part of this initiative, expert consultants from Verizon Business will collaborate with customers to define their objectives; assess their current infrastructure; look at where in an organization investment in these communications tools can make the biggest impact; create user profiles to pinpoint areas in which business processes can be improved; and help organizations overcome other hurdles to deployment and acceptance of these new communications tools.

CA Inc. is one of the customers that has used the Verizon Business professional consulting services, which it used both for its enterprise and contact center operations.

For more on the latest developments in unified communications and collaboration, TMC invites you to attend this week’s ITEXPO event in Miami, which includes a UC track. For specific information on what’s available, visit the track on her website

Avaya's Plan: Migrate Nortel Customers to Aura SIP-Based UC Platform

2010-01-23T19:30:00.001+01:00

Putting to rest any fears about what the acquisition of Nortel’s assets will mean for hundreds and thousands of users, Avaya today unveiled a roadmap that details how it will integrate the products and services from the recently acquired Nortel Enterprise Solutions unit to create a new standard for business communications.
Avaya will speak in greater during this week’s ITEXPO show in Miami, held Jan. 20-22, about its new roadmap, and what it means to customers, resellers, carriers or developers.

“We’re talking about two worlds coming together,” Lawrence Byrd, director of UC architecture for Basking Ridge, N.J.-based Avayatold TMCnet. “The roadmap is not about ‘A’ versus ‘B.’ It’s about ‘C.’ ”

For the past year, that “C” has symbolized the emergence of an open-standards, enterprise-SIP architecture. In March 2009, Avaya unveiled the first fruit of its labors –Avaya Aura – aimed at next-generation communications for enterprises.

In further building a “better Avaya Aura” -- a SIP-based communication platform that unifies complex communications networks -- Byrd told TMCnet it will not use a “rip and replace” strategy. Instead, it will fuse the best technologies from Nortel with its own presence/messaging technologies.

“We have all these customers and all these systems and we have to merge them together,” said Byrd, adding that roughly 6,000 employees of Nortel’s have been fully integrated into Avaya.

Detailed within the larger roadmap are Avaya’s plans for growing its unified communications, contact center and other core businesses.

As Avaya Aura comprises the centerpiece of Avaya’s UC portfolio with its proven abilities to reduce infrastructure costs and easily deploy centralized collaboration and customer service applications, it provides a natural upgrade path for all existing Nortel voice and UC customers.

The next level of Avaya Aura, therefore, will benefit from Avaya (formerly Nortel) Agile (News - Alert) Communications Environment technology. ACE uses service-oriented architecture and Web services to facilitate the rapid development of communications enabled applications.

Avaya Aura will also be at the center of Avaya’s contact center portfolio. With this foundation, contact centers will move to a SIP-based, service-oriented architecture, simultaneously supporting multiple modes of communication. The new contact center solution will be based on a new and collaborative work assignment model that will connect customers, agents and information. The new capabilities will help businesses and organizations to stay ahead of growing customer care demands for tighter integration and lower costs.

“The integrated roadmap we’re outlining today supports Avaya’s commitment to providing customers with the advanced communications capabilities they need to manage and grow their organizations,” said Alan Baratz, senior vice president and president, Global Communications Solutions, Avaya, in a press release. “In developing the portfolio, our guiding principle was to deliver the most compelling and cost-effective systems and services to customers and partners.”

Original source

Unified Communications Comes of Channel Age

2010-01-22T19:19:00.001+01:00

One of the historic problems with unified communications is how disjointed this set of technologies has really been. Most customers are really still unsure what it means to unify their communications and the various components of unified communications solution don’t always work so seamless together.

But at the same time, interest in increasing the productivity of the existing workforce through the better use of IT is on the rise. The question is how can solution providers best leverage unified communications to meet the needs of customers while simultaneously advancing their own business goals? The answer to that question is to start thinking more like a carrier that is delivering set of services that have been customized for a specific customer.

We all know that every major carrier is looking to expand their footprint in unified communications. But the issue these big carriers will have over time is their inability to customize these services. As unified communications evolve in the enterprise, customers are going to want to see these services delivered inside the applications they use every day. That creates an opportunity for solution providers to provide custom unified communication services as a managed service to the customer.

To pull that off, however, solution providers are going to need to deploy a suite of products based in industry standards such as the Session Initiation Protocol (SIP) that by definition creates an open architecture. For example, it’s the maturity of SIP that enabled Siemens to replace Tandberg with Polycom as its preferred partner for the video conferencing component of the OpenScape unified communications platform. That same approach has also allowed Siemens to make a case for layering its software in on top of existing Nortel infrastructure rather than just assuming those customers will become customers of Avaya in the wake of the latter company buying the enteprise assets of Nortel, said Michael Garland, head of the North America channel for Siemens Enterprise Networks.

As time goes on, SIP will be extended to integrate with a host of service-oriented architecture (SOA) services, as promised this week separately by Avaya and IBM, that will allow unified communications to be delivered seamlessly across public and private cloud computing implementations.

In the meantime, unified communications represents one of the best opportunities for solution providers to inset themselves inside the business processes of their customers. And as we’ve said before, whoever controls the business process controls the customer.

original source

How is Unified communications

2010-01-21T07:43:00.004+01:00

Magic Quadrant for Unified Communications

2010-01-21T07:33:00.001+01:00

Unified communications offers the ability to improve how individuals, groups and companies interact and perform tasks. Enterprise planners and managers should review how this emerging generation of communication software and systems can improve their business operations and processes.

Unified communications (UC) offers the ability to significantly improve how individuals, groups and companies interact and perform. UC also enables multiple communication channels to be coordinated. In some cases, separate servers may be consolidated, but, more frequently, UC adds functionality to existing communication applications. Key technologies include Internet Protocol (IP)-PBX, voice over IP (VoIP), presence, e-mail, audioconferencing and Web conferencing, videoconferencing, voice mail, unified messaging (UM), instant messaging (IM), and various forms of mobility. Another key capability of UC is that it offers a method to integrate communication functions directly with business applications; Gartner calls this capability "communication-enabled business process" (CEBP).

Although there is significant interest in UC from many enterprises, it remains a daunting and confusing topic. As a result, many enterprises find it difficult knowing where and how to start. One approach is outlined in "Developing an Enterprise Unified Communications Road Map." This research advises enterprises to review their inventories of communication equipment and business partners, then develop a vision for where their communication could be in five years. This plan can be accompanied by developing a UC center of excellence; this group brings individuals together from multiple areas, including IT operations, business applications and the line of business. This group then provides broad guidance and direction for the plans.

No vendor product adequately addresses all of an enterprise's UC needs. As a result, planners should not expect their UC requirements to be met by one vendor's products: UC solutions require vendors' products to be interoperable. This evaluation focuses on enterprise premises solutions, and considers how well vendors can work with other vendors and with hosted solutions. Enterprises should consider interoperability as an important criterion. Gartner publishes separate research that reviews UC-as-a-service (UCaaS) solutions.

The term "unified communications" sometimes is misused. This results in confusion. Users should be aware that some products that are labeled as "unified" cannot be integrated with other vendor products into a full portfolio. These mislabeled products are capable of being used only in a stand-alone and nonintegrated manner.

Many leading enterprises are developing UC road maps and plans, and some have trial or phased deployments under way, but few enterprises have a fully integrated communication environment. Adoption should continue to increase through the next several years as UC technology solutions mature and enterprises update their installed bases of communication infrastructures.

For more info, visit Gartner's core research

Users spurn traditional calls for Skype

2010-01-21T07:05:00.002+01:00

FRANKFURT (Reuters) - Users wanting to call home from abroad are increasingly turning to Skype's Internet telephony service to the detriment of international carriers, new data showed.

"Skype is now the largest provider of cross border communications in the world, by far," said Stephan Beckert, analyst at research firm TeleGeography on Tuesday.

Skype's technology allows consumers to make practically free long-distance calls over the Internet on fixed lines. It is mostly used on desktops but Skype has made the move into mobile too and it now comes pre-installed on some cellphones.

According to the firm's data, over the past 25 years, international call volume from telephones have grown at a compounded annual rate of 15 percent.

In the past two years this growth has however slowed to only 8 percent, rising from 376 billion minutes in 2008 to an estimated 406 billion minutes last year.

By comparison, Skype's on-net international traffic between two Skype users grew 51 percent in 2008, and is projected to grow 63 percent in 2009, to 54 billion minutes.

"The volume of traffic routed via Skype is tremendous," said Beckert.

In general, TeleGeography said, "demand for international voice has been remarkably robust, but it's clearly not recession-proof."

Traffic to Mexico, the world's largest calling destination, declined 4 percent in 2008 for example, and aggregate traffic to Central America declined 5 percent, data showed.

Established in 2003 and based in Luxembourg, privately owned Skype has more than 520 million registered customers who use the free Web service for voice, video or text communication.

But despite its size, its revenue is relatively modest -- at about $551 million in 2008 -- as the company has had a difficult time getting users to pay for its largely free services.

Skype aims to nearly double its annual revenue to $1 billion in two years.

(Reporting by Nicola Leske)

Securing SIP Trunks

2010-01-21T02:25:00.003+01:00

Security plays an important role for your networks and secure SIP trunk is one of these important jobs. SIP trunking provides organizations with a cost-effective, reliable method of connecting the internal network and telephony systems with external VoIP and traditional phone systems over the IP network. SIP tunking is quickly replacing traditional PRI and analog circuits for enterprise communications. Typically, SIP trunking involves an IP PBX, however Microsoft Office Communications Server 2007 R2 introduced the ability to do direct SIP trunking to the OCS 2007 environment and eliminate the IP PBX. Microsoft only provides direct SIP trunking with two VoIP providers, but Evangelyze Communications developed SmartSIP which lets organizations leverage their existing telephony hardware and employ direct SIP trunking with virtually any VoIP provider.

SIP trunking has many business benefits, but also introduces some additional security concerns. Internal security policies and controls will most likely differ from the security policies and controls of the SIP trunk provider. Connecting with the SIP trunk provider may involve opening ports though the firewall or NAT device, modifying the IP PBX (if present), changing private IP addressing or numbering plans, or other changes to the unified communications infrastructure. The organization must also maintain control over signaling and media secuity as well as call-routing policies.

Organizations have to understand the risks and implement appropriate security measures to ensure the availability of their unified communications network, the integrity and confidentiality of voice and data communications, and the overall compliance with regulatory requirements the organization might be subject to.

SIP Trunking is Key to Accelerating Unified Communications Deployments

2010-01-21T02:18:00.003+01:00

Companies today are undergoing a significant transformation to a more global Anywhere Enterprise™. Unified communications (UC) is a crucial
component in this evolution and organizations look to collaborate better with an extended enterprise (see Exhibit 1 on the next page). UC
has the power to help companies lower the overall cost of communications, bring worker productivity to new levels, enhance corporate green
initiatives and completely redefine the way we work by becoming part of our application infrastructure.
However, the deployment of UC is not without its challenges. Too often, organizations go down the path of deploying new technology with old
technology principles in mind and UC is no different. Many of the early adopter deployments of VoIP and UC were designed exactly the same as
the old systems, severely limiting the overall value of UC, which is a highly flexible, IP-based solution.

Migrating from traditional trunk lines to SIP trunking is a very simple, cost-effective change to open doors to other advanced services that can
enhance a UC deployment. SIP trunking will allow companies to recognize the following benefits:
• Dramatically lower the overall cost of communications
• Extends UC to software-as-a-service (SaaS) applications and other cloud-based options
• Accelerates UC deployments through the simplification of network design
• Easier migration to other advanced services such as mobile integration and MPLS networks
This report introduces the reader to UC, outlines the benefits of UC and highlights some of the challenges in deployment. The report also
introduces the reader to SIP trunking and how it can play a key role in accelerating the deployment of UC as well as provides recommendations
on how to evaluate SIP trunking providers and some thoughts for initial deployment.

For more information, visit infoworld

SIP vs XMPP or SIP and XMPP?

2010-01-20T17:44:00.005+01:00

History
SIP was invented to provide rendezvous for session establishment and negotiation on the Internet. XMPP (or Jabber) was invented to do structured data exchange such as synchronous or active presence and text communication among group of people. XMPP evolved from instant messaging and presence, whereas SIP evolved from Internet voice/video communication. Later, XMPP added support for session negotiation using the Jingle extension, and SIP community added extensions such as SIMPLE to support instant messaging and presence.

Technically comparing SIP and XMPP is like comparing apples and oranges because the core protocols serve different purposes: session randevous/establishment vs structured data exchange. On the other hand, because of the extensions invented in both the protocol worlds, SIMPLE and Jingle, they now have overlapping functions, and can be compared. When one compares SIP vs XMPP, actually the comparison is SIP/SIMPLE vs XMPP for IM and presence and/or SIP/SDP vs XMPP/Jingle for session negotiation. Even though the goals of the two sets of protocols are converging, there are fundamental architectural differences that I will enumerate in this article. There are other articles on SIP vs XMPP [1, 2, 3].

Differences: SIP vs XMPP
The following table lists the crucial differences between the two sets of protocols.

There are many other differences, e.g., the way a URI is represented, or how authentication is done, or what kinds of messages are supported. I will not go into details of those since they tend to become too specific for the kind of application and we miss the important points. From a developer's point of view 'ease of programming' is very important.

Ease of programming
Both SIP and XMPP are easy to implement. My 39 peers project has modules for both in few thousand lines of Python code. Although the basic protocol is easy to implement, a complete system such as a collaboration client with audio/video and messaging/presence support is very complex.

Because of the way these protocols have originated, they are well suited for certain kinds of applications. For example, if you want to build an audio/video communication system, it is better to start with SIP. Features such as interoperability with other VoIP phones, incorporating any-cast call distribution, or using existing VoIP provider for trunking are easy and readily available using SIP. If you want to build an instant messaging and presence client, it is better to start with XMPP. Features such as friends roster, group chat, blocking a user, storing offline messages, etc., are readily available using XMPP. Any advanced communication or collaboration system needs to include both these kinds of features.

XMPP has solved the application's problems and has defined mechanisms for several commonly used features in an instant messenger-type or shared state-type application, e.g., group chat, visiting card, avatars, etc. The emphasis is on application design, use cases, and practical solutions.

I think there are two main reasons for SIP's difficulty among developers: (1) the emphasis of SIP is on interoperability rather than application and feature design, and (2) the emphasis in SIP community is to have one protocol solve one problem, which requires implementing a plethora of protocols for a complete system. Let me explain these further.

When a new VoIP features is implemented by one phone, it must interoperate with another phone or VoIP service provider. Hence most SIP extensions focus on wire-protocol and interoperability mechanisms. Although specifications of several SIP extensions are available, there are no evaluation or open reference implementation on how they fit in the overall design. More recently efforts have been made, including my RFC 5638 (Simple SIP Usage Scenario for Applications in the Endpoints), to simplify the specifications for certain types of SIP applications -- those endpoints that want to work in web and Internet world without the legacy of the traditional telephony systems.

Secondly, SIP community tries to keep one protocol to solve one problem. Some extensions deviate from this guideline, but they are exceptions. The problem comes when this design principle involves implementing several distinct protocols just to get a complete system. For example, a SIP system incorporates other external mechanisms such as STUN, TURN, ICE, reverse-connection-reuse and rport-based symmetric request routing to solve the NAT and firewall traversal problem, and still does not guarantee media connectivity in all scenarios unless HTTPS/TCP tunnel in used. Implementing instant messaging and presence involves implementing several RFCs and drafts related to Event, PUBLISH, CPIM, PIDF, XCAP, MSRP, and still the application does not have all the features of commonly available XMPP client. In summary the SIP community has created numerous extensions for solving several problems in a way that scares away a new developer!

As you can see, both these reasons (emphasis on interoperability and one-protocol-one-problem) are ideal in theory. So what is wrong? The practice. To solve these problems, (1) IETF working groups should not proceed with a draft without an open-source and simple reference implementation, (2) IETF working groups should build reference applications combining several protocols for different kinds of applications and evaluate (a) consistency and (b) ease of programming.

Consistency indicates whether the new extension is consistent with existing guidelines, best practices, protocol format, as well as design principles. For example, if an extension incorporates a new processing in the server which could have been done in the endpoint, then it is against the principle of intelligence in the endpoints. Such extensions should be marked as such so that developers know the trade-off. There are only a few good design principles, hence creating a consistency matrix of extensions against principles should be easy.

Ease of programming is determined by three things: (1) how easy it is to implement the set of protocols, (2) how easy it is to build a real application using those protocols, and (3) how easy it is to build the real application using existing platforms and tools. The first is usually available as a software library, the second as an application and the third is re-usability. It should be easy to not only build the library but also use the library to build a usable application. Every new extension adds new things to the library, which cause more interaction in the application and hence more complexity. When a software project is started, usually the interoperability is not the highest requirement, but the re-usability, short development time and real prototype application are crucial requirements. Once the project is started on one path, it is very difficult to change the path by changing the core communication protocol. If there are reference implementations then not only they help you get started quickly but it also becomes easy to see how much additional complexity a particular SIP extension brings to the application. An important programming quote: less is better than more!

The flexibility of SIP also comes with its limitations. For example, SIP is flexible to support both UDP and TCP transport. However, UDP is treated as a second-class citizen by many programming languages or libraries even today, e.g., Tcl didn't support built-in UDP socket when it came out, and Adobe ActionScript does not have built-in UDP sockets for Flash Player even now. This prevents a developer from building a complete SIP stack as Flash application, for example. However, if you peek further, you would expect that if UDP is not supported then the platform is not suitable for real-time communication anyway. However, this does not prevent web-style developers to implement XMPP in ActionScript, and perhaps tweak it to support signaling of media sessions as well. The result is a broken or non-interoperable software application.

Reviewing the evolution of SIP vs XMPP specifications, I think XMPP has defined an architecture that allows adding new extensions easily and hence reduces the application complexity, whereas SIP extensions have focused on interoperability and wire-protocol without much needed attention to application design. While application design may seem unnecessary for protocol specification, it is very important in the short term. Consider a developer who uses some data structures for representing protocol elements. If a new extension is defined in XMPP, and it reuses the existing XML format that gets readily mapped to the data structures, it becomes very easy to incorporate this new extension in his source code. If a new extension is defined in SIP or SDP, which re-uses an existing mechanism of another protocol for which there is no real implementation available, then the developer will first have to implement that other mechanism, then integrate it with SIP or SDP. The mechanism may have its own formatting which needs to be incorporated in the data structures. Essentially the developer will have to spend more time implementing such an extension. In the end, the actual format of the message whether text-based or XML-based is not terribly difficult once you have a library for message formatting and parsing. However, if an extension uses a different format, connections, sessions, etc., that are not readily available in existing libraries and tools, complexity arises. For example, adding ICE to SIP/SDP created custom format whereas ICE in XMPP/Jingle re-used XML. Another example is how an particular endpoint is identified in XMPP vs SIP. In XMPP the URI itself is extended to include the resource, e.g., "user@domain/resource", whereas in SIP new extension such as globally routable user-agent URI (GRUU) is defined which is, well, more programming effort!

Scalability and performance
SIP is inherently a peer-to-peer protocol whereas XMPP is inherently client-server. Tasks that are easy in client-server systems such as shared state, roster storage on server, or offline messages on server, are well accomplished with XMPP. On the other hand, one of the primary goal of SIP is to keep the intelligence in the endpoint. Ideally, a SIP proxy server does not even maintain the session state for the SIP dialog. Few messages in SIP such as REGISTER and PUBLISH are intended for client-server communication. In XMPP, server is a must and all signaling communication goes through the server. There are message semantics defined for the types of messages, e.g., client-server information query, client-server-client message sending, client-server event publishing and server-client event notifications. Clearly client-server applications are limited by scalability and performance of the server. For example, an instant messaging session need not go through the SIP server saving bandwidth and processing at the server. But that means you lose the offline message storage feature at the server. In real SIP applications today, servers have become an integral part of the system and hence the scalability difference diminishes. In fact, the bulky message format of SIMPLE makes it less scalable than XMPP for presence updates that go through the server. Note also that although P2P-SIP is possible, a P2P-XMPP is not easy because XMPP is inherently client-server.

Once we know this, we understand that SIP and XMPP systems solve two different problems, are designed for two different architectures and have evolved with two different guidelines. From here, you can do two things: either try to incorporate/translate all the features of one system to the other and eventually give up, or try to design your system that uses best of both worlds.

Interworking and co-location
There have been interworking attempts to inter-operate SIP/SIMPLE and XMPP, especially the IM and presence part [draft-saintandre-sip-xmpp-*, draft-veikkolainen-sip-voip-xmpp-*]. The first reference shows how to implement a gateway to connect between SIP and XMPP networks, and the second shows how to implement a client that can support both SIP and XMPP and co-relate the two protocol messages if the user is connected to both servers by the same provider. The popular OpenSER (now OpenSIPs and Kamailio) SIP server has a Jabber module to inter-work with XMPP network. People have developed clients that can understand both SIP and XMPP. Interworking is complex, and not all features can be completely translated or used from one protocol to another, unless the protocol is changed a lot with custom hacks.

Conclusion
Industry experts predict that both SIP and XMPP will stay for a long time. Rather than arguing about the differences or trying to mend the protocols to be like each other, one could build systems that use both these protocols for what each is good at. XMPP is good at creating application level streaming/secure/client-server pipes that can be used for shared state, one-to-many message delivery and publish-subscribe-notify-type use cases. SIP is good at rendezvous of session establishment and negotiation of session parameters for a separate session establishment.

To interwork between XMPP and SIP, you could (1) use a gateway at the server to translate the basic functions, (2) learn or send SIP parameters over XMPP message from a client, or (3) use SIP to establish XMPP chat session with a client. For example, a multi-protocol client of user alice@example.net may be talking to bob@home.com over SIP, and discover that both clients support XMPP, and then add each other in XMPP roster or start an XMPP chat session. Alternatively, if they are chatting over XMPP and discover that the other supports SIP as well, then they initiate a SIP session to do multimedia call. Implementing both the protocols in the client is better than in the gateway for scalability and robustness. There are other interworking architectures possible, e.g., having two XMPP servers use SIP to communicate with each other or talk to a trunking provider, or having an integrated SIP-XMPP server that allows both SIP and XMPP users to seamlessly communicate with each other. These modes, however, are not interesting from a P2P point of view. This article is from Kundal's blog

SIP in the view point of Google, Microsoft, Yahoo

2010-01-20T17:35:00.003+01:00

In our last post, we discussed how GTalk2VoIP brings SIP capability to your instant messengers. Continuing our discussion with Ruslan Zalata, co-founder or GTalk2VoIP, today we will talk about SIP strategies of Google, Yahoo, and Microsoft. The interview extract follows (AS – Alok Saboo; RZ – Ruslan Zalata):

AS. GTalk2VoIP is essentially filling a gap by bringing SIP services to the ubiquitous instant messengers. Why do you think, Google, Microsoft, or Yahoo are not providing such a service themselves?
RZ: It’s about four years have passed by since we started our service. During all this time we cannot help hearing that Google, MS or other Biggie “just almost ready to deploy” a SIP/VoIP service, which will definitely kill ours and make whole consumer VoIP business seamless. Perhaps one day one of the biggies will finally come up with such a killer service, yet there are some obstacles I believe different to each company, preventing them from starting a killer VoIP service. Here are my personal thoughts:

Microsoft. It’s a huge software development company. All their revenue comes from selling desktop (and very few mobile) software. It’s no secret that Microsoft has made a number of attempts to enter VoIP business or online business game, all of them failed or are about to fail. I conclude that Microsoft is just not a proper company for online business, mostly because they got used to super-profits which are hardly achievable on the net due to stiff competition from small startups and low cost of switching for users. This company is no competitor to any online service.

Yahoo. It’s strictly opposite to Microsoft – an online business only company. The problem with Yahoo is that they are incredibly greedy control freaks. Yahoo’s Voice service is pure SIP except that it was intentionally protected with a nonstandard challenge algorithm. Why did they do this ? Because they don’t want their users feel freedom and power of choice. And this is very right decision from marketing point, otherwise they would lose all their customers long time ago (if they have not lost them yet).

Google. It’s very different. Google is damn big and damp popular and this is their Achilles’ heel. They can do virtually anything or start any service. Yet, even Google has problems. Why do you think they don’t offer Google Voice outside of US ? The answer is very simple: because VoIP is very much regulated almost in every other country (esp. when you deal with phone numbers or do offer PSTN termination). Obeying to regulators drastically increases expanses on running such service and the bigger the company the more interesting it becomes for regulator to milk it, sue it, ban it, etc. You have to remember that Google, unlike Skype which is hiding in Liechtenstein, has official offices in many countries (including such VoIP haters like China and UAE). Another thing is, 100% of Google’s revenue comes from advertising on the Web. It’s still unclear how VoIP can fit into this business model. And this, I believe is one of the reason why Google Talk is a dead project inside Google – they don’t know how to monetize it, neither directly nor indirectly. And is also the reason for appearing GoogleTalkGadget and Gmail Voice and Video plugin. Web is their everything, pure VoIP is nothing but a headache.

It is known that inside Google they use a version of Google Talk which supports SIP and allows PSTN calling, but they were not able to make it publicly available because they smashed against this uncertainty of “regulation” problem. There are no evidences that it can be solved in near future.

It’s interesting that there’s a trick to access Google Voice through our GTalk2VoIP service from any country and we have some users who are very happy with this possibility. I’m not going to describe it publicly because I don’t want Google to ban this feature :-) .

AS: I just had a post earlier about how Google Voice perfectly fits in to Google’s strategy. Google gets user data and can then use it to target its services better. May be all the theory is wrong, but if it is the case, would your response about Google change?
RZ: How much personal data does an average Google Voice user does share with Google beside a bunch of phone numbers ? How much of that data can be used for contextual advertising ? None I believe. The only way for Google to monetize on any VoIP service (beside selling minutes) is to build a massive Web service around VoIP service, somewhat like Gmail for SMTP, then enforce users to share a much of their “sensitive” data as possible during usage of this hypothetical Web service, then sell contextual ads. The problem, as I already mentioned, is cost. Gmail, search and others have nearly zero cost of running and have infinity possibility for scaling (cheap commodity hardware). VoIP does not! VoIP is damn expensive to run en mass! VoIP requires special costly hardware and it cannot be easily scaled like web stuff, it requires regular payouts to service providers and telephone companies. So, I believe, it can be a very hard way for Google and it’s partially off of their mainstream conceptions and traditions – it’s very different from web :-) .

AS. Now that Google is getting serious about the Voice business (there have been talks about Google incorporating SIP in GTalk), do you foresee any changes in Google’s strategy? How does it impact you?
RZ: Regarding SIP in Google Talk, please see my previous answer. There will be no SIP in Google Talk. Google Talk is dead project.

Regarding Google Voice, I don’t think Google is getting very much serious about Voice. They are just experimenting, blindly firing in all possible directions, I would say, in search of a new business model to diversify their Web-only revenues. It’s still very unclear how Google is going to make big buck on offering free Voice services. In-call advertizing will not work for sure, there had been many such offerings from different startups and even from well established telcos, all of them are dead now. Because free VoIP attracts people without money, it’s totally impossible to make them pay for advertized goods. We have had a similar negative experience with our Freeringer project so we shut it down. Charging for VoIP calls could give them some new revenues, but is not in Google’s practice or style.

Regarding Google’s future strategy towards VoIP. As I’ve said, Google is playing. I’m sure their strategy will change, they might even acquire one of two more small VoIP companies or startups, like ours, with a non-mainstream solutions to expend their current offerings, thus attracting more users. You have to remember that unlike Web, VoIP is not free to run. Google pays a lot for their Google Voice DIDs, for PSTN termination service, for SMS delivery service, for telco interconnections, etc. This is no way cheap for them. So, I think, if they won’t find a way to monetize VoIP in more or less profitable ways in next two years, they will just shut it down, same way they did to many other services. They also have to solve “regulation” problem as well as the problem with big telcos whose anger is growing fast. It’s also possible that AT&T will just sue Google to death for Google Voice and thus this service disappears in vain. Something similar already happened to Vonage, AFAIK. If Google finally manages to solve all these issue, they will create not only an exceptionally good service, but will bring whole VoIP to a new level. Let’s hope for the best!

Regarding future of GTalk2VoIP. We are strong believers in everything mobile. We believe that more and more people (esp. in developing countries) will use their mobile phones to access to more and more online services including VoIP and here we can help them solve their communication needs. During past few years we are constantly developing and improving our mobile VoIP (mVoIP) and IM chat software called Talkonaut, which is a kind of Google Talk for mobiles. It’s based on same XMPP/Jabber protocol and implements Jingle for calling (this makes it 100% compatible with Google Talk). Unlike ascetic Google Talk, Talkonaut is very feature rich, it supports very impressive list of XEPs (XMPP Extension Protocol) and has a seamless integration with our GTalk2VoIP service. It runs on Symbian S60 and Windows Mobile platforms. Recently we released Talkonaut for Android and are about to release it for iPhone. For these few years we obtains many devoted Talkonaut users. Some use it as a simple mobile SIP client, some use it as mobile Google Talk (cause it looks very like Google Talk and supports gmail accounts), some use it as a cheap PSTN calling solution. We believe and hope this segment of our user base will grow in future and more people will find Talkonaut useful and turn themselves into customers :-) . This is the direction we’ve chosen for near future of our small business.

talkonaut-logoAS: Is Talkonaut similar to Nimbuzz or Fring? Even if it is, there is enough room and you can definitely improve on their features.
RZ:Yes, Fring and Nimbuzz are main competitors to Talkonaut. Both are building proprietary solution using their own home-brew protocols, while Talkonaut is completely based on open standards and can be used to connect to any standard XMPP server directly. Hence Talkonaut can be used in intranets and on proprietary networks which neither Fring nor Nimbuzz can do. Talkonaut does not require to register new accounts, you can use any of your existing Jabber account including gmail.

Another interesting thing about our competitors’ business model is that both Fring and Nimbuzz are building their business around Skype userbase – people using Fring as a “mobile Skype solution”. Recently Skype release a fully capable application for iPhone which allows VoIP calling over 3G and merely today they released Skype app for Symbian S60 platform (I have to say that most Fring users come with Nokia’s smartphones). Availability of native Skype app for Symbian is dreadful for Fring (Nimbuzz has their userbase slightly more diversified). I think Fring is in agony right now :-) .

Thanks are due to Ruslan for his great, bold, and honest comments about some tough questions. If you have any further clarifications or you do not agree with any of the above comments, do not hesitate to drop a line. Look forward to the last part of our discussion where we will talk about ChromeOS and SIP…

P.S. While the official SIP support may or may not come from these companies, you can always use hacks to get such benefit, e.g., read our tutorial on how to enable SIP support in Google Voice.

VoIP 2010: Simplification through Integration

2010-01-20T17:13:00.002+01:00

In addition to the product trends the communications industry has experienced over the past year – all of which will be on display in the exhibit hall and topics of conference sessions atITEXPO ( News - Alert) East in Miami – there has been a significant shift towards integration of products and services between vendors as they seek to provide more complete solution sets for their resellers and end users.

The idea is to provide them the very luxury today’s communications solutions are designed to provide for end users – the ability to focus on their core businesses, in this case, selling product. Naturally, they are able to leverage the latest feature and product enhancements, from mobility to HD and more, but by providing a pre-integrated set of solutions, driven by the adoption of common standards, like SIP, the vendor community effectively simplifies the entire sales process, benefitting the entire value chain.

For more on how vendors like AudioCodes (News - Alert) are working more closely with integration partners, as well as how these and other trends are likely to play out at ITEXPO, I spoke with AudioCodes’ director of market development, Alan Percy.

Erik Linask (News - Alert): What are your reflections of 2009, at best, a challenging year for most businesses, yet one that brought success to many in the telecom industry?

Alan Percy (News - Alert): It was a year mixed with some incredible challenges and surprises. Clearly the economic slowdown that took hold in the first few months of the year created great uncertainty for the entire industry and re-evaluation of business plans, but these challenges were buffered by what we see as some very positive signs. The growth of SIP Trunking, Unified Communications, HD VoIP, improved interoperability and solution integration were all successes that brought new business that countered the economic slowdown.

EL: What is/are the most significant trend(s) to come out of 2009 that will drive the IP Communications space into the next decade?

AP: I’m starting to see a shift in business models that will affect us for the next few years. Looking back a couple years, it seemed our industry was following the PC/IT world business model with separate community of software application developers and then a complementary group of hardware vendors, leaving it to the channels and VARs to integrate the various components.

However, while some were up to the task, it required significant technical knowledge of a number of different vendors’ products and technologies, significantly limiting their ability to deliver solutions quickly and without a lot of integration costs. Lately, it seems there’s renewed interest in pre-integrated solutions, having the software and hardware vendors performing the integration work and dramatically simplifying the solution. The end customer still ends up with a “best of breed” solution, but the VAR didn’t have to go through a science project to get the solution working.

EL: Will 2010 be the year HD audio takes off? If not, when?

AP: This year will, indeed, be the year that HD VoIP does finally get some legs. We’re seeing HD-capable hardware maturing, software applications that support HD, and both enterprise and service provider pilots around the globe. Once you hear it, you will be hooked, too.

EL: How do you view the impact of the growth of the open source community on the overall communications industry?

AP: Open Source has enabled many new a creative applications to come to market at very affordable price points; expect the trend to continue. SIP has been a tremendous enabler for the open source developers, allowing interoperability between open source and commercial products that would be otherwise very difficult. For AudioCodes, open source applications have created opportunities for our media gateways that are needed to connect these applications to the PSTN, user desktops or other existing equipment.

EL: How will your company build on the momentum of the past decade to continue on a growth path into 2010 and beyond? What new and innovative products are in store for your customers?

AP: An important new product initiative for AudioCodes is our BusinessPLUS bundled solutions. This effort brings together the AudioCodes hardware products (gateways, MSBGs, server modules and IP Phones) and combines it with our partners’ software applications to create ready-to-install solutions for the market. As hinted earlier, we’ve solved one of the market limitations and dramatically reduced the complexity for VARs to respond to their customer requirements. As an added bonus, we’ve solved a business model challenge that makes the platform more profitable for both the software partner and AudioCodes.

EL: What is the most important reason your booth is a “must visit” for ITEXPO attendees?

AP: At this ITEXPO, we’ll be showing the first two of our BusinessPLUS solutions, including one based on Digium’s (News - Alert) Asterisk open source platform and another based on Nortel’s Software Communications System, or “SCS.” From what I can tell, this is the first time in our industry that you can buy a complete hardware platform from a single vendor and then choose from a number of pre-integrated software products.

EL: What are you looking forward to most at ITEXPO East 2010?

AP: I always find the opportunity to meet with our customers and partners in a face-to-face setting at ITEXPO exhilarating. With air travel becoming so time consuming and costly, the opportunity to meet with a number of our partners in one comfortable setting is both a wise financial and excellent time investment.

EL: What is the one product you are hoping to see from the communications industry in 2010?

AP: For this next decade, I’m really looking forward to seeing us achieve “Universal Broadband,” basically the IP equivalent of the Rural Electrification Act of 1936, bringing affordable broadband to rural communities and the rest of our great country. Not only will this create business opportunities for our industry, but it will also raise educational opportunities, open up new tele-worker job opportunities, and allow more businesses and governments to go 100 percent electronic and eliminate costly printed forms, catalogs and booklets.

original article

Top Communications Trends in 2010

2010-01-19T03:51:00.005+01:00

As a result of economic conditions in 2009, many companies also sought speedy deployment of converged communications in search of faster ROI. As we move into 2010 and beyond, Avaya’s top 10 communications trends are:

1. Proactive customer communications will enjoy a resurgence in popularity.

The falling cost and increasing sophistication of contact centre communications technologies such as voice and speech recognition systems, live and virtual agents, SMS, e-mail, presence, call routing, etc will allow businesses to deploy strongly beneficial outbound contact centre applications.

2. As the lines between devices and media continue to erode, analysis of employee communications will give companies greater flexibility at all levels.

Businesses will follow privacy standards, but also will increasingly track the phone calls, instant messages and e-mails of their employees to better predict work needs and behaviors. To meet that need, the communications industry will provide greater consistency across multimodal interfaces, whether through iPhones, standard telephones or the Web.

3. Social media will revitalize the contact centre.

Customers will initiate more company interactions via social media tools like Facebook. More businesses will embrace mining of the social network, capturing new opportunities to increase customer loyalty and brand preference.

4. Corporate spend on mobile phone communications will be tamed.

SIP-enabled environments will control mobile phone expenses by consolidating trunk lines and using the corporate IP network to route long-distance and international mobile phone calls at the cost of local calls, drastically cutting roaming charges and other fees.

5. Contact centers will utilize real-time analytics and iterative process re-engineering, making the business more agile.

By tracking customer interactions across a range of media, contact centre representatives will be able to make real-time decisions, allowing them to adapt solutions to trends. This will cover all in- and out-bound communications, including dialogue in self-service applications and instantaneous coaching of call centre agents.

6. Flexible working practices will soar, driven predominantly by business need.

The possibility of hiring workers part time at short notice, and without provisioning offices space, will gain ground. Retired employees may find a second working life as businesses tap into their experience and availability to increase business agility and customer service as the downturn ends.

7. Unified communications (UC) will shift to a three-click experience, massively bolstering adoption rates.

In 2010, with SIP and session management, companies will build, deploy and support applications easily. In three clicks, and in three seconds or under, workers will have access to many more resources and applications—using almost any device they choose.

8. The first true multivendor communications networks will appear.

SIP-enabled communications will help to ensure disparate products from different vendors will effectively work together. Communications will truly become brand-agnostic.

9. Communications overload will cease to be a problem.

At a time when many businesses are struggling with the volume and breadth of communications systems that need to be managed, new technologies will help reign in the complexities to bring greater levels of operational and cost control.

10. Contact centre queues will be made to work for the company.

Businesses will aggressively find ways to make every aspect of their clients' interaction more productive in the call centre environment. A major part of this will be to make richer on-hold experiences the norm: customers will have multiple options for using on-hold time, including ways to move transactions forward before the call centre representative arrives.

source: informationweek

SIP Trunking is Much More Than a PSTN Replacement

2010-01-19T03:41:00.006+01:00

We have reached another tipping point in the telecommunications industry. SIP trunking is the fastest growing service in our space right now and we all have an opportunity to capitalize on this trend, but we must be smart about our approach.

SIP Trunking’s growth presents a new revenue opportunity, but only if the trunk offers services above and beyond PSTN quality voice. If a service provider simply provides VoIP connectivity, they will see their revenues erode. SIP Trunking offers service providers a tremendous opportunity to deliver valuable services to enterprises by providing new communication services in demand by the enterprise market. Enterprises are becoming more educated on SIP Trunking. Practically every large enterprise has read a case study that demonstrates how an enterprise can reduce their trunks by 30% – 40%, which is obviously a negative revenue proposition for the service provider. So service providers must develop a comprehensive managed service offering to enhance and complement their SIP Trunking service.

There are several market trends, which are driving adoption of SIP Trunking by enterprises. Over the next 5-years:

* Enterprise workforces will become increasingly mobile

* Video calling will be widely adopted

* High-definition voice will be the new standard for voice communications

* PBXs will migrate to unified communications

* Enterprises are demanding comprehensive business continuity capabilities

We can easily talk about how SIP Trunking is enabling new revenue opportunities for service providers of each of these trends, but in this post we will focus specifically on the prospects with Unified Communications.

ABI Research recently issued a report “Vertical Market Opportunities in Unified Communications,” which predicts that the unified communications solutions market will reach nearly $4.2 billion in 2014 – a sharp increase from 2008 when the market reached around $302 million.

We all know that Unified Communications (UC) is the integration of varied communication options, like voice, video, email, instant messaging and conferencing, on a single IP platform. The primary benefit of UC is the ability to speed the rate of communications, keeping everyone more closely connected and improves collaboration among employees.

Another capability of UC is the greater control it provides a user over their communications options. With a single Web-based account, individual users can decide when, where and how they can be reached — and users can define these parameters without the need for IT support. (See last week’s Broadband Ignite post: VoIP’s Success in 2009 and Beyond, which focuses on the user experience…and a smart network).

SIP Trunking enables the delivery of Unified Communication capabilities now, from the “cloud, offering service providers an immediate, new revenue opportunity versus shifting that revenue opportunity to the PBX manufactures. With a “cloud-model”, enterprises no longer need to purchase additional equipment to have a full-featured UC solution.

BroadSoft has tightly integrated our BroadWorks product with business applications from Microsoft, IBM, Google, Zimbra and Counterpath in order to enable service providers to offer “UC as a Service.” When service providers host UC in the cloud, enterprises can purchase UC on an as-needed basis. This approach eliminates the need for businesses to spend valuable CAPEX dollars to purchase on premises-based systems, and creates a new revenue opportunity for service providers.

Another business model for service providers is Aastra’s Clearspan product, which integrates BroadSoft’s BroadWorks platform. Clearspan enables enterprises to collapse PSTN connectivity from multiple sites into a single centralized SIP trunk while allowing for a graceful migration of enterprise users to UC without the immediate need to replace costly PBX phones. Service providers can deliver and manage these platforms for enterprises, regardless of their communications environment, PBX, IP or a hybrid.

And the benefits of UC to an enterprise are no longer just a gut feeling. Aberdeen Group recently conducted a study that quantified the benefits of UC to an enterprise, which are:

* 35% increase in knowledge sharing

* 35% increase in workplace flexibility

* 25% improvement of an organization’s competitive performance

* 16% increased of collaboration for decision making

* 11% accelerated speed of conflict resolution

All of these factors present a perfect storm for service providers to increase revenue opportunities, versus creating a commoditized service. They are well positioned to offer advance communication services today through SIP Trunking.

We are hosting a webinar tomorrow, December 2, 2009: Capitalize on SIP Trunkings’ Momentum where we will expand on the other revenue opportunities available with a comprehensive SIP Trunking solution strategy, including:

o Unified Communications

o Fixed mobile Convergence

o Video and Telepresence

o Teleworker Solutions

o Disaster recovery

o Customized communication applications with BroadSoft Xtended

BroadSoft’s goal is to enable service providers to deliver value with SIP Trunking, not simply enable the replacement of circuit switch calls. We will all lose if that is the strategy for SIP Trunking.

original source

Twitter Meets up with Unified Communications

2010-01-19T03:33:00.007+01:00

The convergence of social networking and unified communications was on display this week at the VoiceCon conference in San Francisco.

The Siemens Enterprise Communications Group used VoiceCon as a venue to show off some forthcoming integration between the Twitter social networking service and Siemens’ OpenScape unified communications software.

With all the people using various services on the Web to interact with each other, it’s only a matter of time before these types of services become tightly integrated with unified communications software.

Siemens is encouraging developers to extend OpenScape to these services using a set of published application programming interfaces that it is making available via a cloud computing service. This “sandbox in the cloud” gives developers a way to experiment with OpenScape integration without having to purchase their own infrastructure for testing.

That integration includes the ability to send a message about, for example, going to lunch on Twitter and having Siemens unified communications software automatically change your status to being unavailable and then have all your calls immediate forwarded to voicemail.

Siemens plans to make Twitter integration a standard part of OpenScape in the first quarter of 2010.

original source

Siemens unified-communications adds social networking

2010-01-19T03:14:00.004+01:00

At San Francisco’s VoiceCon 2009, Siemens Enterprise Communications Group previewed Twitter integration with their OpenScape Unified Communications application while running in an Amazon EC2 (Elastic Compute Cloud) environment. Siemens also says they plan to release the SDKs to allow customers to create their own social networking and communications “mash-ups” and test them in a scalable instance of OpenScape.

“Social media tools have rapidly moved from being the preferred communication method of millennials to the standard by which enterprise workers and customers can quickly and freely connect,” said Mark Straton, senior vice president of marketing, voice and application solutions for Siemens Enterprise Communications Group, in a prepared statement.

Siemens also says they plan to integrate other social networks like Facebook and LinkedIn, with all social functionality going live in Q1 2010.

Rival networking companies such as Cisco Systems have aggressively touted social network at channel partner events. Is this all a gimmick or do solutions providers actually think customers want social media blended with their unified communications applications?

Predictions: VoIP Trends in 2010

2010-01-19T03:07:00.003+01:00

By all accounts, 2010 should be a banner year for IP Communications and VoIP in general. A recent report we wrote about saw VoIP as the 'industry of the decade' and then went on to project VoIP as the industry that will see the most growth over the next decade!

With the economy still on shaky ground VoIP will most likely continue its slow and steady growth seen at the end of the decade with companies realizing the cost savings of the technology and looking for ways to invest in upgrades. Upstart technologies like Internet voice and mobile VoIP will be at the mercy of the Net Neutrality rulings while Videoconferencing will probably continue to see growth in line with rising energy costs and the demands to make working cheaper for companies and employees.

Read more: http://www.fiercevoip.com/special-reports/predictions-voip-trends-2010#ixzz0d1J1Bpgd

SIP in Unified Communications

2010-01-19T03:04:00.001+01:00

Do you fire up your Gmail client to send an email to another Gmail user? And would that email fail to arrive if you used Outlook instead? Of course not and it's all down to the standard email protocol called SMTP. Every email client supports it so you don't have to worry what the other person is using. Likewise, HTTP delivers web content regardless of the browser being used.

This has always been the promise of SIP (the Session Initiation Protocol). Designed to establish, manage and 'tear down' communication channels between any media device, anywhere. Sound good?

SIP in Unified Communications

I'm not going to give you a definition of Unified Communications (UC) here, just hit Google and wade through all the interpretations that pop up. I do however, want to talk about how SIP 'underpins' UC.

It does this by ensuring that all elements of a business UC solution can establish connections to work together. It also allows for 'mid session' information to be passed between these elements to enhance the communications experience. A good example of this is starting up a whiteboard application with a colleague to give your existing conversation another dimension.

SIP/SIMPLE

Now UC isn't just about Instant Messaging, but I need to start somewhere.

The IETF SIMPLE (SIP for Instant Messaging and Presence Leveraging Extensions) working group focuses on the application of SIP to Instant messaging (IM) and Presence services. Invariably their work will affect any UC solution.

Now you may not know this, but when you use IM you become a 'Presentity'. That is, you are an 'entity' with a presence state i.e. online, not available etc. So now, if someone wants to see your presence state, their IM client sends a SIP SUBSCRIBE message to your IM server to 'follow' you. If you change your presence state i.e. off to lunch your IM client will send a NOTIFY message to your 'followers'. All of this is defined in RFC 3265, overseen by the SIMPLE working group.

They have also overseen the development of a number of RFCs that cover how presence information is coded in XML documents. These documents are carried in the body of a SIP message and the use of XML here can be taken advantage of by applications to facilitate interoperability and automation of business processes.

SIP/SIMPLE in Action

So, in the background SIP/SIMPLE is delivering (to you) presence state updates for the people you are following. Click on a 'buddy' to start a chat and SIP will carry the message. If you now want to speak with your friend, 'click to call' and SIP sets up the channel through an INVITE. Whilst you are talking you can fire up your Webcam and SIP will establish this connection for you without affecting your voice connection. Don't forget you can bring another person in on the conversation by initiating a conference; SIP can take care of that as well. In effect, it's SIP that connects everything together though it's important to remember that it's the applications being used that are making people more effective in their work.

Open Sesame

Because SIP is an open standard, all UC solutions should be able to work with each other seamlessly. Unfortunately, although everyone has their own presence capability there is no way to 'federate' (or join) them.

Today, if you are on a Cisco system you can't see who is present on an Avaya system. Maybe they'd just like you to buy their solution.

The push for Federations

But hold on! Microsoft and IBM announced (at Voicecon Orlando) that they will support Intra-domain federation so that a company with UC solutions from these two vendors will be a step closer to full UC support regardless of platform and client. Will this 'shake up' other manufacturers into doing the same? And how long will it be until SIP gives us the same 'transparent' interoperability that SMTP and HTTP do? Only time will tell.

Unified communication to grow to $30B market

2010-01-19T02:41:00.001+01:00

Two separate studies on unified communications from Wainhouse (News - Alert) Research reveal the UC industry will grow to a healthy $30 billion market just five years.

UC service providers should see rapid growth due to the complexity of premises-based UC solutions, Wainhouse Research reported. In fact, the research hub anticipates that UC services revenues, which are far smaller than product revenues today, will eclipse those of traditional conferencing services by 2013 or 2014.

For premises-based providers, the telephony market will rebound but the growth is constricted by a reduction in prices per user, and lingering uncertainty in the global economy. Team workspaces and on-premises Web conferencing servers are expected to see double digit growth. Video providers are predicted to see a 17 percent growth over the forecast period.

“In the premises-based UC solutions market, we expect to see declining revenues for PBXs, audio bridges, and unified messaging solutions; however, these will be offset by increasing revenues in presence/IM servers, Web conferencing servers, team workspaces, and video endpoints and infrastructure,” saidBrent Kelly ( News - Alert), senior analyst and partner at Wainhouse Research. “In addition, we are seeing significant interest and now some large deployments in the UC services market, and we expect managed/hosted UC services to grow tremendously.”

UC’s growth will no doubt have a positive impact on equipment manufacturers, as well as service providers.

VoIP phones manufacturer snom AG, for one, has seen growing interest in its 870 phone, which loaded with UC capabilities for business users who need to do a number of things with a simple desktop gadget.

The 870 desktop phone’s features include touchscreen technology and high-definition TFT color display, qualities which make the phone much more user-friendly than other models. According to snom, the 870 allows user to easily handle even complex applications like swapping of calls, or establishment of conference calls for up to five participants via simple “drag and drop.”

Google's next target: Unified communications

2010-01-19T02:04:00.004+01:00

Unified communications has been a technology specialty of networking vendors for years, but Google Inc.'s recent forays into voice communications and collaboration could drastically upset the competitive landscape.

It's not as if Google Voice and Google Wave, which are both launching later this year, will kill related efforts by companies like Cisco Systems Inc., Microsoft Corp. and others that are heavily involved in unified communications, but Google seems to have the competition scrambling already.

Witness today's comment by Doug Dennerline, Cisco's senior vice president of collaboration software: "Google Wave validates what we've been doing for two years [with the Webex Connect collaboration offering]," Dennerline said during a webconference with reporters and analysts. "We are going to invent and reinvent. You'll see cool things from us."

Anybody who has followed the computer industry for long knows that when a vendor says another company has "validated" them, it really means, "Yes, they are clearly in our living room, and we are making sure they don't move in permanently." Dennerline was careful to imply that Cisco is up to the Google challenge and would "invent and reinvent" to stay competitive.

While Google Voice and Wave seem more focused on consumers, with tools for instant messaging, e-mail and social networking, Dennerline was quick to point out that "social networking is important to enterprises, too."

Zeus Kerravala, a Yankee Group Research Inc. analyst who was on the call with Dennerline, said Google Voice and Wave so far are not a threat to Cisco, Microsoft and voice-switching vendors like Avaya Inc. or Siemens. However, he added, "long term, Google will have a significant role" in the voice and unified communications markets.

The main reason is Google's size. "Google has the mind share and capital resources that it can be as big a threat as it desires to be," Kerravala said.

Google could pose a more serious threat to Cisco than it does to other companies, since Cisco has a dual mission of keeping its traditional customer base of enterprise IT operations and service providers happy and well-supplied with networking gear, while also seeking to serve consumers, especially with video technology.

Cisco in March announced plans to buy Pure Digital Technologies Inc. in order to acquire its Flip handheld camera technology. Cisco has also said that it is developing a consumer version of its telepresence system.

"We think video is going to be very key in driving the next level of collaboration ... with Internet video, desktop video ... and consumer telepresence," said Padmasree Warrior, Cisco's chief technology officer, who also participated in today's conference call.

So while Cisco clearly sees its audience as both business users and consumers, a more pertinent question is whether Google intends to go beyond the consumer market with its Voice and Wave products, taking both services into business settings.

The answer to that question, in a sense, is: It doesn't matter. The reason is that Google clearly sees multiple markets, all blended together, where consumers are also workers. Consider this: If Google's Voice service -- which is designed to link all of your phones to one number and features add-ons such as the ability to turn voice mail messages into text -- can be offered to millions of users for free, isn't it likely that people will use it at work? Small businesses could use it and not care if Google is taking some of the information gleaned from users and selling it to advertisers.

Large businesses might not want their employees to use Google Voice or Google Wave tools on the job, but who would stop anyone from doing so, and how? It's the same concern that was raised two years ago with the first-generaton iPhone, which was so attractive to some people that they used iPhones at work even though their IT shops issued security warnings about the device.

Today, Cisco's Warrior said that the company will offer virtual voice service, probably through its service provider customers. That could be interpreted as Cisco's most direct response to Google Voice, even if Cisco officials won't admit it directly. That's because nearly every major wireless or wired service provider sells to both large companies and consumers, and no service provider is going to want a cloud-based service like Google Voice to come along for free and take away paying customers.

So, Cisco's virtual voice in the cloud could give a service provider the ability to tell its own customers, "See, we have our own version of Google Voice, but you can offer it to your customers, complete with Cisco security so there are no worries about their loss of privacy."

Kerravala said he has no doubt that virtual voice from Cisco will compete with Google Voice. "Oh, yeah, its gotta be competitive," he said. "Google Voice is really just cloud-based voice, so that's very competitive with Cisco's telco clients."

Those Cisco clients include many of the major voice and data carriers. The market battle between Google and many companies in unified communications may be quiet so far, but it is still very much a battle. Cisco says it is up to the challenge from Google's Voice, Wave
more info visit computerworld.org

Google and Unified Communications

2010-01-19T02:02:00.000+01:00

Deciphering Google's involvement in unified communications is difficult simply because the company moves so fast and has so many balls up in the air.

That said, it's clear that Google is a significant UC player, whether the definition is broad and generic or the narrow one used by the folks in the business

Investopedia and No Jitter offer, between them, three articles that together form a good backgrounder that provides context for Google and its UC initiatives. Investopedia's James Brumley starts with a nice summation of some of the relevant moves made by the firm during the past couple of years. These include pushing the development and use of the Android open source mobile operating system, the launch of Google Voice, and the acquisition of Gizmo5.

Brumley concludes that Google’s goal is to give away telecommunications services in order to sell advertising. However, a lot of the steps taken to do this also put in place the infrastructure for an expansive UC offering. Dave Michels focuses specifically on UC in a two-part series at No Jitter. Part one is here and part two is here.

Michels profiles myriad Google services, including Docs, Wave, Talk, Blogger, Picasa, Gmail, Gears, HTML5, Latitude, Maps, App Engine, GO and Chrome. He says the company’s search capabilities are the glue. The list is not exhaustive and doesn’t mention the Nexus One, the phone that HTC is creating for Google. Michels then offers an important critique:

Google is in the UC business today, though because of service gaps, integration holes, and a limited channel, [its] solutions are not comprehensive for business. But the technology is there and getting better.

Google, the most powerful force in contemporary communications, is a player in unified communications – using either a broad or narrow definition. It is difficult to understand how everything links together, however. Indeed, it is easy to get the idea that Google doesn't quite get it either. To begin making a name as a player in the UC business, Google has to do a better job of explaining itself.

Google Wave: The Future of Collaboration, Unified Communications and Business Intelligence

2010-01-19T02:00:00.000+01:00

Lars and Jens Rasmussen of Australia, the creators of Google Maps, have done it again.

Google Wave is an open platform and open set of application programming interfaces (APIs) that integrates multiple collaboration techniques into logical, flexible and powerful virtual shared conversations, or "waves." You can "jump in" at any point in a wave's existence, play back parts you missed, and determine whether everyone or only certain people receive whatever you decide to share. Waves can feed blogs with minimal coding. Web sites can be wave-enabled with relative ease. You can access and participate in waves from mobile devices. Waves enable consolidated content collaboration and discussion - no need to choose between, for example, an e-mail thread and a wiki.

There's a whole bunch of other cool stuff in Google Wave, but there's no way I could do it justice here - at least not until I download and become conversant with the APIs and relevant other tools. Which isn't happening - not this week, anyway. You should go to http://wave.google.com and check out the hour-plus presentation and demo, take a shorter "sneak peek" or learn more about the Wave.

And you should expect to be as surrounded by waves as Australia, Tasmania, or your ocean-based land mass. The growth of public, private and hybrid computing clouds is very likely to be mirrored by the growth of public, private and hybrid waves supporting every type of business communication, collaboration or relationship. Which means waves will quickly become essential tools in the service of those pursuing more and better business intelligence (BI).

Why am I so confident? Partly because of what's happened to and with Google Maps - zero to near-ubiquity as the enabler of geographic content and features in Web-based applications in almost no time. Partly because of what's happened and is happening to and with Google Docs & Apps. But mostly because of all of the above, plus it's Google. And because I can't imagine any type of size of business that can't improve communication, collaboration and/or outreach to clients, prospects and partners with the current and likely forthcoming features of Google Wave.

The open APIs and protocols, along with Google Wave's native HTML 5.0 foundations, mean that integration with other online and traditional applications is coming sooner rather than later. And I'm sure that tools for analysis of feature and content access and use patterns are also coming soon. Heck, someone's probably working on direct integration with at least one open source BI tool even as I write this, let alone by the time you read it.

Frankly, I'm hoping to encourage development of more features and integrations among all interested in Google Wave. I am fervently convinced that the delivery of customizable and flexible consolidations of content creation, collaboration and sharing can lead almost directly to greater BI - and more intelligent businesses. And Google has demonstrated its ability to develop and deliver powerful, flexible and open enabling technologies. So I, for one, expect a tsunami of support for Google Wave, and for must of that support to result in new and useful options for those seeking powerful and flexible BI solutions.

How Gtalk2VoIP brings SIP capability to your instant messengers?

2010-01-19T01:56:00.002+01:00

We talked about GTalk2VoIP earlier as wonderful solution to SIP-enable your plain vanilla instant messaging client. GTalk2VoIP is an interesting service, bringing SIP service to your instant messenger. If you have not tried GTalk2VoIP, I would strongly urge you to try the service. While the service is great, we were not sure how the folks at GTalk2VoIP manage this feat. However, I was recently able to speak with Mr. Ruslan Zalata, co-founder of GTalk2VoIP, and he was kind enough to share some insights into the service and other technical details about how GTalk2VoIP brings SIP capability to your instant messengers. The interview extract follows (AS – Alok Saboo; RZ – Ruslan Zalata):

GTalk2VoIP_logoAS: For the benefit of our readers, can you introduce yourself?
RZ: My name is Ruslan Zalata, I’m a co-founder of GTalk2VoIP, Inc., one of the developers of GTalk2VoIP service and a piece of mobile VoIP (mVoIP) software called Talkonaut.

AS: You provide a wonderful service in Gtalk2VoIP where you bring SIP capabilities to instant messaging clients. Can you provide more details about the service?
RZ: Sure! We developed GTalk2VoIP service in early 2006 soon after Google had made their Google Talk with Jingle Audio publicly available. The main idea was and still is to provide Google Talk users with an access to outer VoIP worlds like SIP and phone calling. Despite our deep knowledge and large experience of running VoIP (mainly on Ciscos) it was a great challenge for us: first, we had to reverse-engineer Google’s Jingle implementation (it was not published at that time), second, we had to write our own soft-switch from scratch implementing basics of H.323 and SIP stacks. The first version of the service we released in April 2006 had only one feature: a possibility to make paid for phone calls through one of our VoIP termination partner. We started to receive lots of feedbacks and feature requests, mostly asking for adding SIP support, which we shortly did. In a month, we added SIP gatewaying which allowed Google Talk users to make calls to SIP URIs and and vice versa (i.e. SIP users to make calls to Google Talk through a mapped SIP URI). Then we added a possibility to define users’ SIP accounts to make phone calls through them (Betamax clones such as Rynga, ActionVoIP, and SmartVoIP are very popular among our users). Later in 2006 we reverse-engineered MSN voice protocol and added it to our soft-switch and to the service. In 2007 we cracked Yahoo’s SIP challenged which allowed us to add Yahoo into the set of supported IM messengers. In early 2008 we added AIM/ICQ voice support. Despite the large set of supported IMs, Google Talk users are still 90% majority of our service, so we keep focusing on them, adding more and more features for Google Talk.

GTalk2VoIP - SIP capability to your instant messengersAS: Why should users subscribe to Gtalk2VoIP when they can access some of the services through a SIP client?
RZ: “Subscription” is not a proper word for our service as we do not ask users to create new usernames, fill-in endless profiles with personal data, etc. What users have to do to be able use our gateway is just to add service@gtalk2voip.com buddy into their list of contacts (roster). Our gateway auto-accepts such invitation then automatically creates all the necessary records in our database. So, once service bot contact is added user may use all the services we provide right away (except some paid for services which require adding credits before usage of course). Un-subscription is same way easy – just remove service bot contact from your roster.
To answer why people should use our service, I can tell you that majority of Google Talk users never knew what SIP and/or SIP client was, they are using our service because they find it convenient for them to solve their communication needs, they don’t need to install complex, often over-bloated SIP software, which only highly skilled expert can properly configure. GTalk2VoIP service is damn simple: add contacts, make calls. No usernames, no passwords, no extra software to install.
There is one more thing. Our GTalk2VoIP service is very popular in countries and on mobile carrier networks where regular SIP and Skype are blocked. This is a clever hack to access SIP over Jingle.

AS: Can you provide some insight into the technology that you are using to bring SIP capability to IM clients?
RZ: Technically our service is highly scalable, it runs a cluster of distributed FreeBSD servers and satellite VPS relays. The soft-switch part was written in C/C++ from scratch. We also developed our own FSM framework in Perl5 to run FSM for call processing logic, most service features are merely FSM scripts. Billing stuff partially written in Perl5 partially in PL/PgSQL (we run PostgreSQL 8). There’s some brief description of the technology on https://www.gtalk2voip.com/gtalk_technology.shtml.

AS: What is your business model? I see you have some premium services, but I was just wondering if a user just uses the basic free services, how much does it cost you?
RZ: Our business model is straightforward. We charge customers directly (there’s another story how difficult it is with all the credit cards fraud!) for premiums services. The most popular are: calling to PSTN, selling DIDs and User-defined SIP which allows users to use their existing SIP accounts in Google Talk and Talkonaut for small monthly fee. We run a lot many free services as well, including gatewaying Google Talk/MSN/Yahoo/AIM to SIP and back, gatewaying calls between these voice capable IM messengers, gatewaying to SIPBroker and free calling to 1-800 (toll-free) numbers. It’s no surprise that in early days of our service more than 90% of the traffic was made by people who used free services only. Due to the nature of our service, we are forced to relay all the signaling and media traffic through our servers, which hurt us a lot. It took a great effort and more than two years for us to leverage balance between free and paid-for traffic to 60/40 and this is where business comes to the scene. I cannot tell exact figures due to known reasons, but I can tell you that we have a slightly more than 1 million of “registered” users, 10% of them are active, so you can calculate how much traffic it is and how much does it cost (the average freemium user makes 2500 minutes a month). Since November 2008 GTalk2VoIP service is profitable business. The company itself is profitable since the very start as we made quite some revenue on selling our technology.

AS: Who is your biggest threat/competition?
RZ: Well, almost any regular VoIP/SIP provider can be considered as a competitor to us as they may and always do steal our customers attracting them by cheaper PSTN calling rates (like endless Betamax clones). But, as this is not the only paid-for service we offer, so it’s not a much threat to us. What threats us more is that Google may one day block our gateway or stop distributing their Google Talk or decides to shutdown entire XMPP service they are offering for gmail.com accounts. This will damage us a lot for sure and this is another reason why we direct our steps towards our own mobile VoIP solution – it brings independence and opens new markets.

Thanks Ruslan for your responses!! We will continue our discussion with Ruslan in upcoming posts, where we talk about why Google, Microsoft, or Yahoo may not provide SIP service in the future and integration of the SIP protocol in the operating system. Keep looking…

original source

SIP Services Offer Cost Saving Options: Communications Provider

2010-01-19T01:54:00.001+01:00

As more companies understand the benefits that Session Initiation Protocol, or SIP, services provide, adoption of the technology is on the rise. A recent report from Infonetics Research found that SIP trunking will likely become the second most commonly deployed service this year.
While widespread adoption of SIP services is still a ways off, businesses are looking more at the communication solution simply to save costs, according to a Cincinnati, Ohio-based communications provider. As signs of an economic recovery increase, companies still have cost savings solutions at the top of their list to keep their operations functional.

“Companies, I think, are looking to SIP to save money,” Matt Ziebro, vice president of enterprise sales at PowerNet Global Communications, told TMCnet in a recent podcast. “The traditional TDM solutions require an install of a dedicated TDM connection – some type of WAN connection to their other offices. SIP goes over the regular data connection... through the public Internet, or you can actually create private line connections between customers and ourselves.”

The ability to combine voice and data on one line and have it integrated gives companies more flexibility and lower costs, Ziebro said. What’s more, other features can be offered with SIP services than were traditionally only available under some more advance networks carriers offered.

PowerNet Global (News - Alert), for instance, offers companies cost savings with its recently launched SIP lines service. Rather than offering pre-packaged SIP features, PNG’s á la carte service lets businesses of all sizes enjoy feature-rich advancements at a fraction of the cost, officials said.

To hear the podcast in full, click here. For more about the PNG, or its new SIP Lines service, visit the communications provider channel on TMCnet.

Can a VoIP Solution Lead Us Into the Future?

2010-01-19T01:51:00.001+01:00

Voice over IP is the process of an IP network to carry telephone voice signals as IP packets over the internet or other data networks. IP packets are small packets of digital bits of information which are routed through the Internet infrastructure. The bit sequence of the packet (field structure) usually is arranged to convey the destination address in addition to the data that is already being transported along with other data such as the packet originator and error protection bits.

To implement a VoIP Solution all you need is a microphone for instant free voice communication in a chat room. By uniting the traditional phone system and the advanced communication capabilities of using a single data network for all communications, it is possible to reduce overall costs of communication for end users.

VoIP solutions can be utilized anywhere a broadband modem is connected to the Internet, or with the combination of a wireless network card, it may be connected over a wireless local area network. A wireless enabled device can access the Internet any time it has access to network signal.

This type of technology determines the proper system that will be used for a communication session, automatically sets the relevant parameters of the communication at both ends of the transmission and then manages call transfer and call termination. In addition, SIP (Session Initiation Protocol) allows users to initiate and receive communication from any location, networks are also capable to identify users, wherever they are. The information packets are transmitted over any open line rather than a dedicated line making communication much more effective.

The mass amount of simultaneous communications can be significantly expanded using packet switching over circuit switching. VoIP is a cutting edge technology that has the potential to completely redesign the entire world’s communication infrastructure. The benefits of Wireless VoIP over cellular connections has lead the majority of interest in this method of communication thus far.

VoIP solutions have the potential to fulfill all your communication needs whether it may be telephony, data transfer, video conferencing… There is no limit to the possible benefits that may be offered.

Many large companies and industries are now discovering the huge advantage of implementing VoIP to establish a better communication systems, often at a lower cost than traditional telephone service. Also, companies making these claims concluded that VoIP rendered no communication delays when speaking on a VoIP Phone System.

This modern day communication innovation offers a variety of different applications and features providing clear and affordable conversation all over the world. Call detail records currently provide a reliable means of safely routing data to call accounting or telemanagement systems. Call record buffers have long been largely accepted as the chosen storage device as a safeguard against incidents of delayed call collection or communication failure. VoIP possesses many benefits that will surely have an a advantage to dominate a large proportion of the communication market for both individuals and business in the near future.

The present business sector has seen a huge growth in real-time and interactive business traffic. Prompt response requires a layer of application-aware control that will actively link the performance requirements of a growing mix of converged data. VoIP solutions and video applications combined with available network resources must be blended in a manner to achieve a heighten user experience.

Your business can celebrate the benefits by implementing a comprehensive Internet network in your offices so they may communicate better at lower cost. The primary objective of a VoIP service provider network is to establish private communications amid individuals.

When consolidating a single data network internally within offices and between employees your business can largely reduce operating costs, increase productivity and efficacy, plus enhance communications and customer services. When you operate a business wireless network that’s local, you will discover VoWiFi (Voice Over Wireless Fidelity) to be a great benefit plus extremely cost effective.

Issues with network connectivity has prompted the urgency towards a Hybrid VoIP phone systems. Hybrid VoIP solutions will need to combine two or more systems that would have the ability to process multiple types of signals, physical channels, and services at an affordable rate to the end user. Before selecting any provider for a VoIP Hosted PBX service make sure they are furnished with the proper Network Management System (NMS) for their VoIP deployments, one that provides their Network Operations and Field Technicians to locate and troubleshoot VoIP problems before a complaint is ever registered from the customer. Cutting edge network monitoring systems now employ advanced diagnostics linked to an online knowledgebases that automatically provide network technicians with troubleshooting solutions.

VoIP overall is still the best solution that the communication world has to offer. With a seamless telecommunication ambience of where reduced limitations of time and distance would be the ultimate objective behind any new born invention in the communication spotlight. Therefore, making a VoIP solution efficient, yet effective communication technology will lead us all into the future.

original source

Compare VoIP and Traditional Phone, It's Free

2010-01-19T01:44:00.000+01:00

In a family with several children, it's not very good to compare them with each other. Oftentimes this will result to a feeling of enviousness and rivalry. They would also try to compete for their parent's attention. Comparing can be very harmful to the kid's relationship towards each other.

Comparing things can be useful if you're trying to compare a VoIP service provider. And this goes true even to different businesses offering products and services in order to attract customers; they want to compare their products/services against a competitor. Whichever provider that comes out best will get the fairer share of the market.

This is one reason why all business owners have developed ways to attract prospects, including all VoIP providers. But do you know what VoIP is?

The main technology responsible for internet phones is the voice over IP, more popularly called VoIP. As its name implies, this new technology allows any person to make voice conversations over the IP network.

Through VoIP, a P2P session is established by optimizing protocols as a group. This will result to an uninterrupted stream of digital data carried through the net. The two VoIP protocols which are now widely accepted are the open standards and the session initiation protocol or SIP. Voice signals are continuous in nature, and so it requires this session.

Now, three internet phones are being used, namely: PC softphone , VoIP handset, and the ATA or the analog telephone adapter. The first one is a software that makes use of the computer's sound card, speakers, and microphone. The VoIP handset relies on your personal computer and is more complex than the first one. The ATA will not require you to have a personal computer because the high-speed internet connection is directly connected to it. These internet phones also have different softwares.

VoIP providers use any of the three phones and software. So if you have preferences especially with the hardware, you have to compare different VoIP providers. By comparing them, you will be able to choose one that has almost all your needs.

The VoIP provider is responsible for the initiation of P2P sessions. Aside from that, they also associate the telephone numbers to different IP addresses, and route VoIP data to other personal computers, to the mobile number or landline number that you're trying to call.

Although VoIP providers differ in their hardware and software being used, the process of making a call is still the same. They only differ on the hardware's processing.

Not all people are ready for this new technology. They still compare it with their long-trusted conventional telephone. In comparing these two, the biggest difference lies on the costs. With VoIP, you can surely cut down your monthly phone bills. VoIP is often associated with 'free' calls.

VoIP service is well-known for free calls for promotional or trial basis. However, when you sign up for their service, you will be asked to pay low-cost plans to enjoy the service. But this cost is lower than that of traditional phones.

Since VoIP is still new in the market, other people want to rely on their traditional phones for reliable connections and exceptional sound quality.

In doing a comparison between a traditional phone and voice over IP, you can see their advantages and disadvantages over the other. If you try count each of their good and bad aspects, you can come up with a good decision. It's either you settle for VoIP or remain loyal to your traditional phone.

You alone can compare these two services. If you have a regular phone, why not settle first for a free trial period? Look for a VoIP provider which offers such service, you can find a lot of them on the internet. After the trial period expires, make the decision. Trying it yourself can give you a clearer view of how VoIP works.

If for some reason, you're not satisfied with the result, then continue with your traditional phone service.

Comparing VoIP and the regular phone service will only be effective if you know what you need in a phone service. Most people who go for VoIP are the ones which want added features on their phone, one that only VoIP providers can provide.

source articespawner

How An IP Phone Controls

2010-01-19T01:43:00.001+01:00

Something that everyone ought to bear in mind of is how an IP phone works, specifically because of the multitude of benefits that you are offered by an IP phone and phone service. An IP phone works by using VoIP technologies that permits telephone calls to be made over the Net instead of through the ordinary PSTN system.

The IP phones use protocols like Session Initiation Protocol and Skinny Shopper Control Protocol, and can be merely software-based Softphones or purpose-engineered hardware devices that seem abundant additional like an normal telephone.

There are varied options when it comes to buying an IP phone for yourself, the most fashionable version being the wireless IP phone as a result of then you do not have any wiring or cable to fret regarding and it will be much easier and additional convenient for you to use. There are different sizes, designs and brands that you’ll select from, therefore opt for from a large choice and find one that’s simply right for you.

Advantages

There are a ton of different benefits that you will receive if you begin using an IP phone, one amongst the biggest being that there can be a simplified infrastructure. You will also cut back operating prices as a result of a VoIP exchange is based on software instead of on hardware and therefore it’s a lot of easier to change and maintain.

Another of the most important advantages is that there’s added flexibility. A Virtual Personal Network is an allocated amount of bandwidth on the general public Web where public access is prevented through encryption.

You can make all the same calls only much quicker, easier, and for less money. Everybody needs to save lots of money wherever they will, and not enough individuals are aware of just how reasonable the VoIP service is and why we have a tendency to should all be switching over to it.

Simply create positive, if you are considering hunting and creating the switch, that you first make yourself a lot of a lot of educated and aware on the services and the way it all works thus that you are feeling comfy and knowledgeable and are really going to be able to create the foremost of it. If you’ve got determined that you wish to switch over to the VoIP service, you’ll find everything you need to induce started and get going at general phone corporations, and therefore the Internet is also a terrific resource that you’ll use to seek out everything you would like to urge started with VoIP calls.

SIP Trunking 2.0 - The Service Provider Perspective

2010-01-19T00:57:00.000+01:00

In the world of business communications, SIP Trunking was one of the strongest trends in 2009. For both businesses and hosted providers, the basic rationale is evident, particularly in a weak economy where everyone is looking for ways to cut costs. Unless you’re an incumbent telco who likes the status quo, SIP Trunking makes the value proposition of IP telephony stronger by reducing telecom costs in a few basic ways. Aside from allowing even more voice traffic to be routed off the PSTN, SIP Trunking reduces the need for costly PRIs and, in some cases, can eliminate them altogether.

These benefits are tangible, and in many cases are reason enough to adopt SIP Trunking. However, they essentially replicate TDM services, and beyond the cost savings, offer little in the way of sustainable competitive advantage. In our view, this is SIP Trunking 1.0 – cost effective, but short on innovation that enables differentiation. With SIP Trunking becoming more mainstream now, we see the market being poised to go beyond this in 2010. SIP Trunking 2.0 speaks to the fuller potential of end-to-end IP, which builds on voice, but extends to other modes such as video, mobility, conferencing, presence and high definition – both audio and video.

To better understand and define the SIP Trunking 2.0 opportunity, a variety of service provider customers were interviewed at our seventh annual executive users’ conference, BroadSoft Connections 2009: Voice & Vision. Their confidential insights provided a wide range of perspectives, not just for the opportunity in front of them, but for their customers as well. This article highlights the key themes coming from these discussions, and provides a preview of what we can expect to see in 2010.

The Opportunity for Service Providers
SIP Trunking appeals to service providers and businesses alike, but for different reasons. Herein will be the start of our discussion. First, it must be noted, that there are many types of service providers, and each will have a distinct set of drivers. Examples include mobile operators, national CLECs, regional CLECs, incumbents, and cable operators.

Across the board, however, most service providers face the same challenge – their customer base is still primarily TDM – which means declining revenue for core services. All the growth is coming from IP, which they are uncertain about how to monetize. Cost conscious customers want to keep using their PBXs for as long as possible, making it a challenge to transition them to newer IP-based services.

Why should service providers look at SIP Trunking here? In short – customer retention. They still need to provide TDM and support legacy equipment, but there is a continuous shift occurring to IP. In time that will mean fewer PRIs, T1s, E1s, etc., which means reduced revenues. Most carriers are willing to accept this tradeoff to keep the customer, and eventually sell them on IP services, which tend to be high margin. For now, this is a SIP Trunking 1.0 story, but without the customer, there is no 2.0 story to tell.

To build the SIP Trunking 2.0 story, service providers are increasingly partnering with BroadSoft for hosted solutions. Most providers do not have a native SIP infrastructure, and a platform such as BroadWorks addresses many of the challenges they face to keep their business customers happy. One key challenge is ensuring QoS and proving to customers that IP is every bit as good and reliable as TDM. Without this, they will not be able to introduce 2.0-style services, especially those based on wideband codecs that deliver an experience that TDM cannot match.

Another challenge that hosting addresses is support across a wide range of PBX (News - Alert) vendors. Interoperability has long been an issue, especially for multi-site enterprises, who typically have several PBX systems. SIP-based interoperability standards have advanced to the point where hosted solutions can support service across multiple vendors and allow the carrier to automate the provisioning process for their customers. This is a real value-add, since the enterprise can benefit from SIP Trunking despite having multiple systems, and without needing to replace them.

On a more practical level, a SIP Trunking partner makes for better management of network resources. The greater the reliance on SIP Trunking, the less need for TDM switches, which in turn frees up space in the Central Office and reduces power consumption. This is of value to carriers in a couple of ways. First, it is easier to enter new markets. Less investment is needed to set up COs and with the scalability of SIP Trunking, carriers can ramp up service quickly in new markets.

SIP Trunking also makes their overall cost structure more competitive. This is important since most operators do not own the last mile. Direct access to the customer will always be costly, so they have to make their money based on the cost efficiencies of their networks. Another example would be wireless operators who want to serve business customers. They generally do not have PRIs of their own, so SIP Trunking is the ideal way to go. In cases where their customers have MPLS networks, SIP Trunking is a natural add-on, paving the way for wireless operators to capture a larger share of the overall voice business.

Without these points of entry, there is no basis for SIP Trunking 2.0 services. They can choose to just focus on lower cost telephony and deliver a good 1.0 story. However, by thinking more broadly about the full potential for SIP-based services, carriers can position themselves higher up the value chain as solutions providers. This is where the growth is for services, as well as the new revenues that will replace the declining TDM business.

VoIP 2010: Simplification through Integration

2010-01-19T00:54:00.000+01:00

Citrix Online Now Offers Convenient Conferencing Shopping

2010-01-19T00:52:00.000+01:00

Organizations and their managers looking for alternatives to increasingly wretched, expensive, and productivity-killing and environment-degrading business travel for their staff no longer have to spend countless hair-pulling hours shopping around for options.

Citrix Online, the Web conferencing division of Citrix Systems (News - Alert) now offers a range of audio conferencing services along with its Web conferencing suite. Business of all sizes can now experience a variety of high definition audio and Web conferencing services together. With this comprehensive offering, customers benefit from enhanced convenience, performance and affordability.

Through Citrix Online (News - Alert), customers can choose a standalone audio solution with HiDef Conferencing or HiDef Corporate, as well as integrated toll-free, toll-based audio or VoIP with Citrix GoToMeeting and CitrixGoToWebinar services. With the addition of an integrated toll-free option, participants in the same GoToMeeting session or GoToWebinar event can, for the first time, select the audio option that is best for their needs.

HiDef Conferencing is distinguished by broadband capability that significantly enhances the quality of voice transmission compared to other audio conferencing providers. Users dialing into conference calls experience crystal clear, rich and natural audio that is more like a face-to-face meeting. HiDef Conferencing services combine simplicity with powerful functionality.

“To meet the market potential and our customers’ needs for greater choice in audio conferencing technologies, we are bringing innovative solutions to market faster than anyone else,” Ben Lilienthal, general manager of Citrix Online Audio, LLC, said. “Customers, who have long trusted Citrix Online for its innovative and reliable Web conferencing services, can also now experience the convenience of one-stop-shopping for audio conferencing that is unsurpassed in value for the richness and clarity of the audio communications. Through continuous innovation and close attention to customer needs, we are well-positioned to succeed in this market, which offers tremendous growth opportunities.”

Citrix Online’s strategy addresses the strong growth potential for high definition and affordable audio conferencing services. According toWainhouse ( News - Alert) Research, an independent firm that provides insight on the global markets for this sector, Citrix Online is well-positioned to meet the demand.

“The market opportunity for Web and audio conferencing volume is approximately two times larger than the current one, providing ample opportunity for Citrix Online to accelerate its growth,” Marc Beattie, managing partner andCSP ( News - Alert) practice manager, said. “The unattended audio conferencing category alone, which is where Citrix Online Audio focuses – is expected to surge – growing 93 percent, and almost doubling in volume between 2008 and 2013.”

Larry Blumberg & Associates, a hotel development and management services company is one such firm that is pushing up those numbers. After trying other vendors it discovered Citrix Online’s all in one approach to Web and audio conferencing when they purchased HiDef Conferencing and GoToMeeting.

“This convenient approach meets our need to deliver online meeting and conference calls to distributed teams with different requirements,” James Snellgrove, the firm’s director of financial analysis and risk management, said. “The convenience and simplicity of the products, coupled with the level of service, allows our company to operate more efficiently.”

Orange Business Services Optimizes Microsoft Unified Communications Delivery

2010-01-19T00:50:00.000+01:00

Orange Business Services has announced the expansion of its consulting and integration services for delivering Microsoft (News - Alert) Unified Communications.

The company provides customers with consulting services whether integrated with IP telephony or as a stand-alone Microsoft solution.Orange Business Services ( News - Alert) will assess business and technical requirements for designing and implementing Microsoft-based unified communications solutions.

The Microsoft-based unified communication solutions can be integrated within the customer’s IT and real-time services such as enterprise telephony, mobility, voice, video and conferencing.

The major benefits offered by Microsoft-based unified communication solutions include greater collaboration within and outside the company, reduces voice costs, pricing, reduced travel and better workflow thereby increasing worker productivity. Orange can customize solutions according to the requirements of the client while integrating on- premise or managed and hosted data center with UC as a service or a hybrid solution.

According to a press release, Orange is working with Microsoft for testing, validating and integrating Microsoft Office Communications Server Enterprise Voice with Business Talk Global, a global voice VPN service.

“Orange Business Services provide an impressive combination of messaging, IP voice and mobility expertise,” said Betsy Frost, general manager MicrosoftUnified Communications ( News - Alert) Marketing, Microsoft Corp., in the release. “When combined with their global network reach, flexible hosting capabilities and Microsoft Gold Partner status, this represents significant value for large enterprises wanting to implement unified communications.”

Customers will benefit from Orange consulting services in designing and implementing Microsoft unified communications like Microsoft Enterprise Voice and Conferencing based on Microsoft Office Communications Server 2007.

The latest versions of unified communication and collaboration offerings including Microsoft Exchange Server 2007, Microsoft Office Communications Server 2007 and Microsoft Office SharePoint Server 2007 are supported by Orange Business Services. These services are supported with ITIL standards-based service management for secure and reliable performance and enhanced end-user experience, said company sources.

“Multinational corporations need help removing complexity when integrating communications within their organization to reduce costs while improving business processes,” said Didier Jaubert, senior vice president, Global Services, Orange Business Services.

Jaubert also said that Orange consulting and integration services help customers unify their communications for supporting business growth.

Avaya, Cisco Contribute to Unified Communications Market Growth in Third Quarter: Report

2010-01-19T00:43:00.000+01:00

In the recently released Group Enterprise Telephony Quarterly Report, Dell'Oro Group reported that the Unified Communications (News - Alert) market grew in the third quarter of 2009. More than 70 percent of the reported vendors contributed to the market’s quarterly growth.

The top two Unified Communications vendors, Avaya and Cisco, have each posted double digit revenue gains over the second quarter of 2009. According to Alan Weckel, Director at Dell’ (News - Alert)Oro Group, the second half of the year is typically stronger for the Enterprise Voice market, and the third quarter of this year was no exception for the Unified Communications segment.

Weckel said that even though vendors were expected to experience strong fiscal year-end results in the fourth quarter of 2009 and 2010, Dell’Oro Group did not expect vendors’ fiscal year-ends to be as strong as they were prior to 2008.

The report said that IP Phones grew 10 percent during the quarter with contributions from all IP Phone (News - Alert) segments. These include IP conference, IP desk, IP soft, and IP WLAN phones. Although third-party IP desk phones continue to gain traction, the top eight IP desk phone vendors continue to be the top eight PBX vendors as well. These eight vendors are Aastra, Alcatel-Lucent, Avaya, Cisco (News - Alert), Mitel, NEC, Nortel, and Siemens.

The Dell’Oro Group Enterprise Telephony Quarterly Report offers complete, in-depth market-level and detailed vendor market share information on the markets ofPBX ( News - Alert), Telephones and Enterprise Voice Applications. The report contains tables covering manufacturers’ revenue, average selling prices as well as line (analog/digital and IP) and unit shipments for PBX (IP, Hybrid, Traditional PBX, Traditional Key) and Telephones (IP and Legacy Business Phones). In addition, tables that cover manufacturers’ revenue for Unified Communications and Enterprise Voice Applications are also present.

The Dell’Oro Group report also tracks PBX and Telephone segments by region, PBX vendor shipments by region, Unified Communications market and the Unified Communications vendor market share revenues by segments.

The company provides in-depth, objective research and analysis that enables component manufacturers, equipment vendors, and investment firms to make fact-based, strategic decisions. The in-depth market information from Dell’Oro Group helps clients keep abreast of current market conditions and take advantage of future market trends. from tmcnet.com

Bringing VoIP to Social Networking with IVR Technologies' Talking SIP

2010-01-19T00:08:00.000+01:00

Social networking Web sites have become very popular in the past few years. For the most part, these sites are geared toward consumers rather than business users. And, while most let people stay in touch using text-based communications, voice capabilities to date has not been a major focus in the social networking market. But, this is changing.

In relatively recent research, The Yankee Group (News - Alert) predicted that subscribers to pure-play VoIP services will reach 6.4 million by the end of 2011 (up from about 2.8 million at year-end 2006). Meanwhile, Nielsen/Netratings estimated that use of social networking sites is experiencing almost 50 percent growth annually. Tie these two together, and it seems obvious that voice integrated with social networking is a huge market opportunity.

For next-generation service providers, adding voice to a social networking system is a new and potentially very lucrative revenue stream opportunity. How to take advantage of the opportunity, though? This is whereIVR Technologies comes in. The company specializes in RADIUS billing replacement and SIP server technologies, and its Talking SIP platform can be used by service providers to integrate voice with social networking.

Talking SIP is a turnkey platform that can be integrated into a social networking portal, bringing voice capabilities to that setting. Once integrated in this way, a variety of methods can be used to initiate voice conversations.

Some of the network configurations that could be used to provide voice services to social networking users include Web page-enabled softphones, mobile phones, IP phones,WiFi phones and traditional phones. One advantage of Talking SIP for this type of application is that it can be set up to protect the privacy of users. Embedded links for initiating voice conversations can disclose as little or as much connection info as the user prefers.

For example, let’s say a social networking site subscriber is contacted by a visitor who wants to initiate a voice conversation. That visitor can click on a link to start a call, and the two parties can be connected, without either disclosing their identifying information (real name, phone number, etc.) to one another. This type of setup could be used in a variety of settings, such as on a dating Web site where protecting the privacy of users is paramount.

With Talking SIP, service providers also get a flexible set of billing tools. These capabilities can be used to create a variety of different billing models, such as packages, fee-based, flat rate and unlimited. Such models can be custom-tailored for specific portals and specific subscriber bases.

For next-generation service providers, Talking SIP offers an opportunity to drive higher margin revenues by offering differentiated applications to subscribers. Using Talking SIP to integrate VoIP with social networking is a way to attract new subscribers and retain existing ones. It also affords the opportunity to cut network operating expenses and administrative costs by consolidating and streamlining network management. for more information visit www.tmc.net

A Look Ahead at 2010

2010-01-19T00:01:00.001+01:00

Wireless Internet - with the increasing use of handheld smart phones, wireless Internet usage will soar and with it will come increased congestion, reliability issues and eventually usage fees. The "all you can use" wireless Internet is going to get expensive.

Apple will end exclusive arrangement with AT&T - with competition from Google Android (the operating software for the "Droid") and ever increasing network congestion and performance issues on the AT&T network (see above), Apple will be forced to finally end their exclusive relationship and expand to other carriers. At first to other GSM carriers (T-Mobile) and then CDMA carriers (like Verizon and Sprint). However, an iPhone for CDMA networks requires a different radio in the iPhone which means there will end up being two different phones and users will never be able to jump from GSM to CDMA carriers without buying a new phone.

Google will aggressively exploit Apple / AT&Ts performance issues (see above) and become a strong competitor in the mobile device market with both GSM and CDMA carriers (still with different devices).

HD Voip - the pieces are all there, and the first few carriers are readying their product launches. Your wireline telephone is about to become obsolete. Once you hear it - you will understand.

Fiber-to-the-home - will grow as fast as they can bury the conduit. Whether it comes from Verizon with their FiOS offering or your local cable operator, coax and twisted pair copper wires are dead. Blazing speed Internet, HDTV and video on demand over IP will drive a wooden stake in the heart of the old copper infrastructure. Long live FTTH!

Video over IP - I've seen the future and it's unbundled video over IP services like Netflix, Blockbuster, and Amazon.com with their streaming or digital downloads to your DVR. They completely obsolete video on demand with it's limited choices and start times. Also, why would anyone go into a video rental store ever again?

Blu-Ray - with the HD-DVD / Blu-ray battle ending two years ago and sufficient HDTVs in people's homes, it's time for Blu-ray to take off. But, only if Blu-ray disks also include a DVD or digital copy for the kids room/car/portable player. We've got too many other DVD-only devices around the house to limit playback. But, Blu-ray will have to compete with Video over IP services and will never be as ubiquitous as DVD rentals were 10 years ago.

Twitter is in Twouble - it's 15 minutes of fame are over and it will basically settle in as a replacement for PR news wire services with content by the people and for the people. Much like web pages, email, YouTube and blogging, social media in general will find a home in the suite of communications tools. The bottom line is that talented people will always be needed to create interesting/funny/insightful content to keep your attention - otherwise it's just noise.

Morse Code - sorry to see you go, but when the ARRL and FCC dropped it from the Amateur Radio operating requirements, now there is absolutely no reason to keep it around. --. --- --- -.. -... -.-- .

Jobs recovery - Green energy, Biotechnology and security are the growth areas for the next decade. Kids - make sure you prepare yourselves for these industries / careers.

Energy/Fuel prices - with our sputtering economy taking a long slow recovery, I suspect we will enjoy relatively low fuel prices at least for the first six months. This will give time for the auto manufacturers to figure out how to make a plug-in-hybrid for the masses. Hopefully it will be ready when the prices of fuel take off again.

10 years, how digital technology will change?

2010-01-14T08:36:00.000+01:00

10 years, the world of technology has been dominated by the big power of Google and social networks, the surprising resurgence of Apple as well as a decrease gradually from Microsoft and Yahoo. So the world of technology will change how the next 10 years?

It's hard to believe the first decades have occurred largely as predicted by technology experts from 10 years ago. Concerns about the Y2K problem likely to happen yesterday when the world experienced breathing phào relief fluctuations first century and this continued to worry about people completely unfamiliar , once the details of our life on the social networking site Twitter or Facebook. Worry serial worry but what continues in the next 10 years. Many technology experts wonder that, in 10 years, whether Google would still dominate the field of Web? Microsoft will still exist? It is possible and here are the predictions that the humor but not without basis for the technology world for 10 years.
Social networking boom strength:

Will be 11.5 billion people all over the planet to register for the social network and Facebook will open up a social network for the race without borders. However, users will have to worry much about security and privacy on it.
Life 3 (Third Lives):

People born and raised, in school, married, older, death will come but we have to thank the creative new incarnation of the real life inside the virtual world. Microsoft is working on a new system to create the actual incarnation of appearance of the user within the virtual world. That is entirely possible and we should hope.
Google phone to interact directly to the brain:

The implant gAmygdala (located in the center of the brain where processing triggers emotions in humans) has been waiting a long time. That will create a wireless link between the brain Android users with their mobile devices. Finally, users can test this function but the service cloud the brain (Brain Cloud) Google will be maintained in a trial in seven years.
The brain is super:

Microsoft will create a version of its competitive gAmygdala IE17 with the browser application Bing Live Brain Implant 1.0. Two weeks later, the airline will issue a firmware update (firmware) to patch holes caused exploded head.

The film illusion.

The latest version of James Cameron, Avatar 6 (the revenge of those Na'vi) will appear in 3-D theaters worldwide give users the experience of fantasy-based world vivid 3D.
Ending litigation legal form of computer tables and thrive:

Michael Arrington will eventually win lawsuits against manufacturers equipment JooJoo table (formerly called CrunchPad). However, the tablet in the future be used to surf the web, thin as paper and capable of connecting broadband 6G will appear in supermarkets for about $ 4.99 USD.
Y-holes:

Department merger between Yahoo and AOL will announce a major reform of CEO after the first 37 - Miley Cyrus on its inauguration. No one except Billy Ray will report that information.
Amazon:

Will swallowing Netflix, Blockbuster, ABC / Disney, Comcast / NBC and any independent film today. Amazon announced that the company controls all media content that is not owned by the booth iTunes. In the future, all chosen its video on demand will still meet most users.
Twitter:

The most common primary blog in the world will celebrate the number reached 1 billion billion (1018) tweet of them by sending the first message is 140 characters to and from Uranus (Uranus).
iMortal Apple:

Near the end of next decade, Apple will announce iMortal, a new application for iPhone 7GSS. Device will allow users to store all the memories took place in their life for generations of future children can access them. And perhaps the first to deserve the "marvels" iMortal, of course, is Steve Jobs.
United States of Google:

2020, Sergey Brin and Larry Page will the U.S. presidential candidacy. Official action will be their first overhaul the education system by requiring all elementary school students gAmygdala chip installed. Monday's action is to change their motto U.S. currency by altering the phrase "E pluribus unum" (a country of many ethnic groups) to "I'm Feeling Lucky" (I'm Feeling Lucky) - clusters from being used on Google to choose the first result of the command search.

10 năm tới, công nghệ số sẽ thay đổi như thế nào?
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10 năm qua, thế giới công nghệ đã bị chi phối bởi sự lớn mạnh của Google và mạng xã hội, sự trỗi dậy đáng ngạc nhiên của Apple cũng như sự giảm sút dần dần của Microsoft và Yahoo. Vậy thế giới công nghệ sẽ thay đổi như thế nào trong 10 năm tới?

Thật khó tin khi thập kỷ đầu tiên đã diễn ra hầu như theo dự đoán của các chuyên gia công nghệ từ cách đây 10 năm. Những lo lắng về sự cố Y2K dường như mới chỉ xảy ra vào ngày hôm qua khi cả thế giới thở phào nhẹ nhõm trải qua biến động đầu tiên của thế kỷ và đến nay lại tiếp tục lo lắng về những con người hoàn toàn xa lạ, lần theo những chi tiết trong cuộc sống của chúng ta trên các trang mạng xã hội Twitter hay Facebook. Lo lắng tiếp nối lo lắng nhưng điều gì sẽ tiếp diễn trong 10 năm tiếp theo. Nhiều chuyên gia công nghệ tự hỏi rằng, trong 10 năm tới, liệu Google sẽ vẫn thống trị lĩnh vực Web? Microsoft sẽ vẫn tồn tại? Điều đó là có thể và dưới đây là những dự đoán có phần hài hước nhưng không phải là không có cơ sở cho thế giới công nghệ trong 10 năm tới.
Mạng xã hội bùng nổ mạnh:

Sẽ có tất cả 11,5 tỷ người trên hành tinh đăng ký tham gia mạng xã hội và Facebook sẽ mở ra một mạng xã hội cho các cuộc chạy đua không biên giới. Tuy nhiên, người dùng sẽ phải lo lắng nhiều về vấn đề bảo mật và tính riêng tư trên đó.
Cuộc sống thứ 3 (Third Lives):

Con người sinh ra và lớn lên, đi học, lập gia đình, già đi, cái chết sẽ đến nhưng chúng ta phải cảm ơn sáng tạo mới của việc hóa thân sự sống thực sự bên trong những thế giới ảo. Microsoft đang làm việc trên hệ thống mới để tạo ra những sự hóa thân thực tế về ngoại hình của người dùng bên trong thế giới ảo. Điều đó là hoàn toàn có thể và chúng ta nên hy vọng.
Điện thoại Google tương tác trực tiếp tới bộ não người:

Sự cấy ghép gAmygdala (nằm ở tâm của não là nơi xử lý các yếu tố gây cảm xúc ở con người) đã được chờ đợi trong một thời gian dài. Điều đó sẽ tạo nên một sự liên kết không dây giữa bộ não của người dùng Android với các thiết bị di động của họ. Cuối cùng, người dùng có thể thử nghiệm chức năng này nhưng dịch vụ đám mây não bộ (Brain Cloud) của Google sẽ duy trì ở dạng thử nghiệm trong 7 năm.
Những bộ não siêu nhỏ:

Microsoft sẽ tạo ra phiên bản cạnh tranh gAmygdala của hãng như trình duyệt IE17 với ứng dụng Live Bing Brain Implant 1.0. Hai tuần sau, hãng sẽ phát hành một bản cập nhật chương trình cơ sở (firmware) để vá lỗ hổng gây nổ tung đầu người.

Những bộ phim ảo tưởng.

Phiên bản mới nhất của James Cameron, Avatar 6 (sự trả thù của những người Na'vi) sẽ xuất hiện tại các rạp hát 3-D trên toàn thế giới mang lại cho người dùng những trải nghiệm về thế giới tưởng tượng trên nền 3D sinh động.
Kết thúc kiện tụng pháp lý về máy tính dạng bảng và phát triển mạnh:

Cuối cùng Michael Arrington sẽ chiến thắng vụ kiện chống lại nhà chế tạo thiết bị bảng JooJoo (trước đây có tên là CrunchPad). Tuy nhiên, các tablet trong tương lai được dùng để lướt web, mỏng như giấy và có khả năng kết nối mạng băng rộng 6G sẽ xuất hiện trong siêu thị với giá khoảng 4,99 USD.
Y-holes:

Vụ sát nhập giữa Yahoo và AOL sẽ công bố thêm một sự cải tổ lớn nữa sau khi CEO thứ 37 - Miley Cyrus của hãng lên nhậm chức. Không một ai trừ Billy Ray sẽ thông báo thông tin đó.
Amazon:

Sẽ nuốt chửng Netflix, Blockbuster, ABC/Disney, Comcast/NBC và mọi trường quay phim độc lập hiện nay. Amazon công bố rằng, hiện hãng kiểm soát mọi nội dung truyền thông mà không thuộc sở hữu của gian hàng iTunes. Trong tương lai, mọi sự lựa chọn video theo yêu cầu của hãng sẽ vẫn đáp ứng được hầu hết mọi người dùng.
Twitter:

Tiểu blog phổ biến nhất trên thế giới sẽ kỷ niệm con số đạt được 1 tỷ tỷ (1018) tweet của chúng bằng cách gửi đi những tin nhắn đầu tiên có 140 ký tự tới và từ sao Thiên Vương (Uranus).
iMortal của Apple:

Gần cuối thập kỷ tới, Apple sẽ công bố iMortal, một ứng dụng mới dành cho iPhone 7GSS. Thiết bị sẽ cho phép người dùng lưu trữ toàn bộ những kỷ niệm diễn ra trong cuộc đời của họ để các thế hệ con cháu trong tương lai có thể truy cập lại chúng. Và có lẽ người đầu tiên xứng đáng với “tuyệt tác” iMortal, tất nhiên là Steve Jobs.
Nước Mỹ của Google:

Vào năm 2020, Sergey Brin và Larry Page sẽ ứng cử tổng thống Mỹ. Hành động chính thức đầu tiên của họ sẽ là đại tu hệ thống giáo dục bằng cách yêu cầu mọi học sinh trường tiểu học cài đặt chip gAmygdala. Hành động thứ hai của họ là thay đổi phương châm tiền tệ Mỹ bằng cách thay cụm từ "E pluribus unum" (một quốc gia của nhiều dân tộc) thành “I'm Feeling Lucky” (tôi cảm thấy may mắn) - cụm từ đang được sử dụng trên Google khi muốn chọn kết quả đầu tiên của lệnh tìm kiếm.

SIP Trunking 2.0 - The Service Provider Perspective

2010-01-13T22:55:00.000+01:00

Google launches Nexus One smartphone

2010-01-06T11:29:00.000+01:00

Google has launched its much-anticipated debut smartphone, the Nexus One, confirming speculation that it will sell the handsets both via carrier partners and independently via an online e-tailing strategy.

The GSM-based Nexus One is now available in Hong Kong, Singapore, the US and the UK. Google is selling an unlocked version from $529 (€368), and is also approaching carriers to stock the phone.

From today Google is selling the handset via this new direct online sales channel. The online sales channel will be the host for a range Android phones to be progressively added to its site. “The Nexus One is the first in a series of phones to be brought to market through this store,” Google said.

In the US, T-Mobile USA has agreed to sell the Nexus One for $179 on a two-year contract, and Verizon Wireless plans to follow suit in the future.

Vodafone will sell the handset to its customers in Europe, Google said but did not reveal how long that period of exclusivity will last.

Google product management VP Mario Querioz said “we expect to add more operators, more devices, including from Motorola and other handset manufacturers as well as more countries to this program - we will bring the web store to more countries.”

He also confirmed that Vodafone in Europe will also be added to the Google store site and to the new model from spring 2010. “We’re working as quickly and hard as we can to ensure the store is ready for business but also to make sure we offer the different flavors of the phone through these different operators later this spring.”

Just as the market rumors indicated, the handset has been developed by HTC to Google's specifications. It is powered by Google's own Android OS, and includes a 1-GHz Snapdragon chip.

"The Nexus One belongs in the emerging class of devices which we call ‘superphones,’ with the [onboard] chipset making it as powerful as your laptop computer of three to four years ago,” Google VP of engineering Andy Rubin said.

The GUI includes shortcuts to Google services such as Google Maps, Gmail, YouTube and Google Voice. The handset sports UMTS, HSPA, GSM/EDGE, Wi-Fi and Bluetooth connectivity.

Rumors of a Google phone have been circulating for years, but some observers had expressed doubt that Google would risk alienating handset makers at a time when the Android OS is just beginning to gain traction.

Selling the handsets unlocked, and therefore removing the element of handset exclusivity, might also discourage operators from subsidizing the phone. But these appear to be risks Google is willing to take. For more information visit telecomseurope.net

GoogleVoice with Nexus One

2010-01-06T01:39:00.000+01:00

We're excited to see today's launch of Nexus One, particularly since Google Voice is one of the pre-installed apps on the phone. This is the latest version of the Google Voice mobile app we released for Blackberry and Android-powered phones last July.

Google Voice comes pre-installed on Nexus One phones sold in the United States. Existing Google Voice users can log-in and access their voicemail messages right away, while new users can set up Google voicemail as an alternative to their carrier's voicemail.

With the Google Voice mobile app, you can receive free voicemail transcriptions and play messages in the order you want. A karaoke-style interface lets you easily replay any part of a message, without needing to listen to the whole voicemail. You can also place international calls via Google Voice and enjoy Google Voice's low rates from your Nexus One.

Last but not least, the app is synchronized with the web, so you can access and listen to all your voicemail messages from any computer by simply logging in to www.google.com/voice.

If you're already using Google Voice and have a Google phone number, you can display this number as the caller ID on outbound calls. Additionally, you can send and receive text messages using the Google Voice app for free.

To learn more about the app, visit m.google.com/voice. And for more on the Nexus One, visit google.com/phone. As always, we welcome your feedback on how to improve Google Voice. For more information visit http://googlevoiceblog.blogspot.com

Almost 1.2 Billion SIP Users by 2012

2010-01-06T00:21:00.000+01:00

Session Initiation Protocol (SIP) services will become the norm after 2010 and will rapidly begin to dominate the world’s telecom markets. By 2012 almost half of all telecom users will be using at least one SIP-based service, but more than likely will have many services from multiple devices able to communicate with other users and services across the Web and between enterprise and public networks. This will generate over $150 billion in service revenue annually with cumulative infrastructure capital expenditure of over $10 billion by that date.

A move to all-IP networks is the chosen path for introducing new services, with NGN the ultimate goal.

“The path to this all-IP goal is complicated: migrating existing services onto IP networks while retaining resources until they can be taken out of service is not a straightforward process,” says ABI Research analyst Ian Cox. “Mobile operators’ voice services are already optimized to reduce network traffic, and the move to VoIP is not an easy choice until the introduction of LTE or HSPA. One method that has gained some momentum is to use an IP overlay based on SIP. This allows new services to be designed and launched using a well-supported standard that also opens the way to bring Web services, service delivery platforms, and ultimately IMS into the network.”

Cox further comments that, “Using SIP, telephony becomes another Web application, which can be integrated into other Internet services. It allows service providers to build converged voice and multimedia services.”

By 2012, ABI Research expects almost 1.2 billion VoIP users to be active, most users also subscribing to several forms of messaging and video sharing driven by the interest in user-generated content. Additional services supported by SIP will include presence, click-to-dial, buddy lists, email and Web access which are assumed to be “core” services and will be included as standard in any service offering, and bundled with broadband access. A portion of the VoIP users will also be connected to a fixed-mobile convergence service.

”The Worldwide SIP Services Market” reviews the world market for SIP services. It includes forecasts for the market potential to 2012, including users (consumer and business), services revenue and capital expenditure. It forms part of three subscription Research Services: Mobile Operators, and Fixed-Mobile Convergence, and Wireless Infrastructure.

Founded in 1990 and headquartered in New York, ABI Research maintains global operations supporting annual research programs, intelligence services and market reports in broadband and multimedia, RFID & contactless, M2M, wireless connectivity, mobile wireless, transportation, and emerging technologies. For information visit www.abiresearch.com

Google Universal Voice in 2010?

2010-01-06T00:16:00.001+01:00

And if Google Voice in 2010 became a real competitor to Skype, the most popular software for Internet telephony? It will require removing the restrictions that surround it, namely its accessibility to only U.S. territory, by invitation, and positioning more focused on unified messaging and numbers than actual telephone conversations to PC PC or PC to phone.

And this is what seems to be the ambition of Google executives, cited by TheNextWeb site. They claim that after the acquisition of start-up specializing in VoIP Gizmo5 last month, they have specific tools and know-how to incorporate mobile devices in computers and for the communication to be accessible from any connected device. Spitting image of Skype!
One can imagine the upcoming release of a new version of Google Voice, which operate entirely on Internet protocol, without need of a physical issue at the base. What would the Web abolishing borders to internationalize this service. If this were the case, Skype would be useful to introduce innovations in its software to meet this new competition. (EP)

interview David Bryan on P2P technologies

2010-01-06T00:14:00.002+01:00

Telecosm Interview: David Bryan on Peer to Peer Technology

I am pleased that David Bryan agreed to be interviewed for the first of what I hope will be a regular feature on the Telecosm blog: interviews! I am very interested in peer to peer technology and its many applications, and was thrilled when David agreed to help me launch this part of the blog. David Bryan wears two very big hats:

* He is Chief Executive Officer of Sipeerior Technologies, a leading provider of standards-based, OEM software solutions designed to enable serverless implementations of real-time communications applications.
* He is co-chair of the Peer-to-Peer Session Initiation Protocol (P2PSIP) working group at the Internet Engineering Task Force (IETF), the standards-making group for the Internet.

I am particularly pleased the David was able to take some time for the interview from his busy schedule, as there is an IETF meeting underway in Philadelphia this week.

IKE: Peer-to-peer file-sharing applications such as BitTorrent get a lot of press, so when people hear peer-to-peer they could be excused for thinking only about file-sharing. What other applications could be enabled by peer-to-peer technology?

DAVID: I think that is a great question. Peer-to-Peer is one of those things where people not only immediately think file-sharing, but immediately think music piracy too, which is really unfortunate. Peer-to-Peer is a technology, in the same way that client-server – the mechanism used for everything from email to web servers – is a technology. I think as people get their heads around peer-to-peer you will see it used in more and more places. It really is just a way to take advantage of the tremendously fat pipes we have in place to our homes, and the powerful devices we have connected to those. It's a great way to leverage that power, for companies to offer services without the need for large data centers, and to be "green" while doing it, since the end device is being fully utilized.

People also think about only large, Internet-scale communities connected by P2P. I think that you will see many of these, but will also see P2P being used within an enterprise or even a home. Communications, in the form of VoIP or IM is obviously one area where this is very attractive. Small office systems can be server-less, and incorporate all the wonderful features that large office systems have today, such as IM and presence. These systems are running today. Consumer electronics is another area where you will see P2P being used. Some of this looks like file sharing – video on demand content shared between set-top boxes for example, but it can also be used to enable features such as communications between users of consumer devices or for gaming. Imagine being part of a community of users with digital home hubs, where I can post content, look at others’ content, and then communicate and notate their content. Today that would require a big back-end commitment from the vendor. With P2P, much of that can be pushed to the edge.

IKE: Why is SIPeerior’s technology important?

DAVID: I think there are number of reasons what we are doing is really unique and important. We have focused on working on a standards-based approach to P2P. One of the problems P2P has today is that many of the applications that employ the technology are either "black" or at least "grey" in the eyes of ISPs and system administrators. They may not want to allow BitTorrent or Skype traffic on their networks. A big part of this is that it the technology isn't transparent. They don't know what they are dealing with. Similarly, it won't interface with existing devices, and it won't usually operate in a stand-alone, enterprise way.

What we have done is worked since day one to promote and open standard for peer-to-peer (P2P), and in particular P2PSIP. We are working to promote a documented, open standard for the P2P part, and using established, well known mechanisms for call signaling (SIP), NAT traversal (ICE), encryption (DTLS), etc. We offer software for building generic P2P applications, as well as very specialized software for P2PSIP. Both are based on open specifications, and "play nice" with other Internet applications. Our VoIP side products are SIP compliant in the sense that you can take a cluster of P2PSIP phones and connect in a few off the shelf SIP phones, gateways, or even servers if you want a hybrid architecture. You can do that with devices based on SIPeerior's technology today. I think the approach of being standards-based for both the P2P and the call portion is really unique.

We’ve also really focused on reliability and redundancy. Our customers aren’t building a file-sharing network where downtime is acceptable. They need things to work in real-time, and P2P can be a very good way to handle reliability, when done correctly, and I think that is what we bring to the table – a robust, embeddable software library that is open and allows vendors to eliminate their servers.

IKE: How could peer-to-peer VoIP applications improve on the current, widely-deployed client-server VoIP applications like Vonage?

DAVID: I think this question really depends on who's perspective you are looking at this from. For a provider, particularly a hosted provider like Vonage, P2P can be a way to provide a very similar service at a far lower cost, using what we refer to as a hybrid architecture, where some servers are used but much functionality is pushed to the edges using P2P. Essentially, they can push P2P technology into each phone or edge box/ATA in the network, and offload all the basic call processing away from their core. No need to buy soft switch ports or data center capacity for them.

While doing this, they can still use centralized servers for billing, call termination, and even some advanced services, and if they are using P2PSIP, these servers and gateways can be the same ones they are using today. If SIPeerior software was being used in this type of deployment, this would all be secured using our certificate mechanism, so the certificates are used to authenticate the peers to one another, ensuring the overlay contains only paying, trusted devices, and also to provide their credentials for each centralized service. In many ways you get the same thing IMS offers, of allowing user-by-user access to each service and feature, but it is all handled in the certificates issued, making things much simpler from a deployment perspective.

For the end user, regular services look no different, which I think is a very good thing. However, the end user over time will see many features added that would just be too costly to add if the provider needed to back each one with a server to implement it. That is where the real magic in P2P comes in for the end user, since suddenly the only cost for a new application becomes developing it, even if it involves lots of data sharing. The end user's device and connection enables it, not the provider or vendor's data center.

IKE: The IETF meeting started on March 9th in

Philadelphia
. What is the current state of peer-to-peer SIP protocol development? What progress is expected at the

Philadelphia
meeting?

DAVID: I think the meeting in

Philadelphia
will be a very exciting one. The P2PSIP group actually won't meet until Friday this time, since it is our rotating turn to go last, but I think everyone is very excited. The P2PSIP work has actually been going on for a very long time. I introduced the first draft on P2PSIP to the IETF back in January of 2005, and we had the first official meeting as a separate working group (rather than under the regular SIP- related groups) in March of 2007. As a result, we have a number of rather mature proposals at this point. The hope is at this meeting that we will be able to select a proposal (one leading candidate is actually a merge of several earlier proposals) that will serve as the basis for a real standard going forward. At that point the battle isn't over of course, the standards process is like making sausage, but there will be a reference for folks to look to for their implementations, and a place where consensus is documented. It will be a big step for P2PSIP, and I think at that stage things will really take off and we will see many different interoperable implementations emerge.

IKE: Ebay’s Skype is often credited as having the most widely-deployed peer-to-peer VoIP application, though the application is not completely peer-to-peer. What pieces of the Skype application appear to rely on servers, rather than peer-to-peer client software?

DAVID: There was an excellent paper out of Columbia University that took a look at the Skype application a few years ago, which can be found at http://www1.cs.columbia.edu/~library/TR-repository/reports/reports-2004/cucs-039-04.pdf. The methods they used to figure out what was going on are fascinating, since Skype is a closed protocol and everything is encrypted, meaning they had to use experiments to figure things out. Basically, Skype seems to use a central login server, contacted each time a user joins, while search and relaying of messages for users is distributed. This type of architecture is what is generally referred to as a hybrid architecture. This is a bit different than what a similar approach would look like using our software or what the IETF P2PSIP group is currently proposing. We'd still have a central server, but it would only be a certificate server, contacted at enrollment time and periodically when certificates expire. All of the day-to-day operation would be strictly between the peers.

It is also very interesting that when an application says it is P2P, it can mean many things. When I talk about P2P, I generally mean reducing or eliminating the central servers so that as much functionality as possible is pushed to the edges and resources are shared. Some other applications appear to mean only sharing resources, but still using centralized boxes. The Ooma service comes to mind, where at least from what I can tell, they still use servers for their call processing, but what is P2P about it is that users share each other's phone lines to allow for low priced calls, taking advantage of local lines. In many ways, it looks very much like Free World Dialup from a few years ago. I think it is really important when looking at a P2P application to determine exactly what it is that is being pushed to the peers.

IKE: How will peer-to-peer technology change the Internet?

DAVID: I think we are already seeing this today. File sharing and Skype account for a tremendous fraction of the bandwidth used on the Internet. In the longer term, I think we will have many little sub-clusters of devices – perhaps every instance of a particular manufacturer’s device will form one and all the computers or phones of a group of people with a common interest will form a P2P cluster. I think this is really the part of the big picture we aren't seeing today, since right now most of the P2P applications are built around a service model, really, where each user comes together into a giant collection. I see having a standard as driving P2P technology into more devices, and that means that new applications will emerge and smaller and smaller groups will form. Think about it like your buddy list, only it might be a P2P community of interest, able to collaborate, share data, and communicate.

I also think that the Internet will be changed in far less visible, but equally important ways. There are very strong economic incentives to move services that generate little revenue or are orthogonal to your main application away from your machines and out to the edge. Quite frankly, telephone services are clearly one of these. VoIP has made the service so cheap that it almost makes no sense to keep it on centralized servers when you can push the cost and bandwidth to the users in the call. Companies that figure out how to offer unique services this way will be very successful, because the infrastructure costs to deliver them approach zero other than development. We all heard how YouTube was 10 guys in a garage and a whole bunch of servers at a colocation facility somewhere. That is the real cost of that service. The costs of YouTube may have been very low relative to what the company is valued at, but even those costs will be squeezed out by P2P going forward. The change will, honestly, affect nearly everything about the Internet.

IKE: Peer-to-peer protocols have been predominantly used in consumer VoIP applications, so far. What opportunities are there for using peer-to-peer protocols in enterprise VoIP applications?

DAVID: Well, I talked a bit about this in responding to question 1, but I think a few areas really stand out. Replacing the PBX or key switch with a cluster of phones is the obvious one. This can be done for an entire enterprise, or just a branch office that is trunked back to a central PBX. This can scale to very large scales, since each new phone brings more processing, but we are already seeing some vendors position P2P as only applicable for the small office. Particularly for P2PSIP, the fact these devices speak regular SIP makes a small office P2P offering attractive, since as they grow they switch over to being "dumb" SIP phones attached to a PBX, but there is no technical reason you can't run a ten-thousand person enterprise on P2P. Combining wireless with P2P, perhaps with a femtocell base station is another area I think is exciting. Having your cell phone "magically" turn into a PBX device when you are in the office is a very exciting prospect.

Using P2P between enterprises is another area I think is very interesting. If each enterprise incorporates P2P into their firewall/SBC/edge device, the enterprises can cluster among themselves, and essentially eliminate the carrier in the middle. I think you'll see this both with companies that have many small offices, for example retail players, and from hardware vendors who want to create communities of their customers. You may also see this from carriers as a cost reduction mechanism.

I also expect to see many hosted providers switch to P2P. It saves administration and cost if they can drop their central switches and push much of the functionality to the edge. They basically only need to offer SIP trunks. I think that these kinds of applications and the larger enterprise systems will be very popular, particularly as the mechanisms to troubleshoot and configure P2P mature in the next couple of years.

SIP is the future of telecommunications

2010-01-06T00:13:00.002+01:00

The 18th Internet Telephony Conference & Expo (ITEXPO) came to Los Angeles last week, and being the opportunistic sort, I decided to get myself on the press list. That’s one of the advantages of writing for ZDNet. Conference organizers will often let you attend their creation for free in hopes that you might write something about the products on display therein.

I’m not, however, one of those writers who likes to post a play-by-play of the happenings at this or that conference. I prefer to attend the entire conference, visiting all the classes and study groups which, to my mind, serve as the highlight of such things, only to figure out afterwards what new thing I have learned from the whole process.

I did note that this was the 18th ITEXPO…which means that there were companies thinking about a future where communication was entirely mediated by Internet-related protocols as far back as 1990. In 1990, the notion that a web of connections mostly linking academic institutions might serve as the foundation underpinning something as critical as voice communications must have seemed, to some, like so much technophile hyperventilation. Clearly, however, the Internet has been more than a little bit succesful. Furthermore, protocols designed to run over Internet connections seem to be reaching a tipping point with respect to their adoption by telecommunications companies. I’m not just saying that because I am still responding to the “reality distortion field” caused by the strange chemicals conference organizers insert into their continental breakfasts. Looking back, most of my career seems to have been spent working for one telecommunications company or another (though video-related work seems to have occupied most of the rest).

One thing that has already been rolling along for quite some time, but has become extremely apparent given recent developments in the industry, is that Session Initiation Protocol (SIP) isn’t just a protocol that is “popular” in Voice over IP (VOIP) environments. It is, for all intents and purposes, THE VOIP protocol, at least among traditional telecommunications providers. It is the protocol that networks choose when they want to IP-enable their largely SS7 environments (SS7 is the signaling protocol used in the global circuit-switched network used to communicate within and between almost every telecommunications company in existence). SIP plays a central role in the IP Multimedia Subsystem (IMS), a family of protocols which is supposed to define the architecture of next-generation mobile networks capable of streaming various kinds of text, voice and video data to mobile phone subscribers even as they roam between networks.

Though the people at the conference were clearly fans of an IP, and SIP-based, future, nobody believed that SS7 was going to go away tomorrow. The existing investment in SS7 is massive, and telecommunications companies aren’t going to replace overnight a technology into which they have sunk many billions of dollars in investment and which reliably serves the voice communications needs of the entire world. However, as we move beyond a network mostly built to cater to the needs of voice, SIP is simply a more flexible alternative.

Session Initiation Protocol (SIP) is, at least syntactically, a lot like HTTP. It confines itself to simple negotiation of data streams between endpoints, and doesn’t impose any kind of rules as to the nature of those data streams (though streams of data are usually packaged inside RTP, another publicly-ratified protocol). SIP, in other words, can be used to negotiate a voice connection, as well as exchnage text message of varying sizes and complexities, exchange presence information, and exchange any kind of data the nature of which can be described via MIME headers and the protocol details for which can be described via Session Description Protocol (SDP, which is exchanged as part of SIP negotiation). SS7, in theory, can be used to negotiate multiple kinds of data streams as well, but the history of the development of the traditional phone network means that most endpoints can, at most, handle basic audio encoded according to G.711 rules (a limitations of the vertically-integrated switch hardware that was the norm for most of telephony product history). Further, though SS7 is a simple signaling protocol as well, it doesn’t have the level of expressiveness and flexibility as a text-based protocol like HTTP or SIP.

HTTP has proven to be one of the most popular media exchange negotiation protocols in existence. SIP borrows many of those semantics, and makes media exchange negotiation more flexible still.

But if I don’t dive deep into a blow-by-blow description of the intricate details of the SIP protocol Right Now, knowledgeable readers will pillory me for being excessively vague. So, I will stop here, and note that you can find stacks of documentation about the SIP protocol on the Internet and / or books published by any number of publishers. That’s the magic of a public and officially-ratified protocol. There’s no real mystery about its details.

Of particular note at this conference, however, was the emphasis on the concept of “SIP trunking.” Traditional TDM trunking is often via a T-1 / ISDN PRI connection. T1 connections are ridiculously expensive ($500 / month is not uncommon, and it used to be much more), particularly when you consider that they only provide a maximum bitrate of 1.544 mbps over the 24 unique voice / data channels a T1 provides, each of which runs at 64 kbps (the European equivalent, E1, gives you only a bit more, at 2.048, with 30 voice channels). Given the bandwidth limitations of TDM trunk connections, few businesses would use a T1 or E1 link for anything but their voice communications.

This makes T1 connections an extra expenditure on top of digital data links. Further, there is a translation step for many businesses using modern business telephony products, many of which use SIP as a means to communicate between telephony endpoints within the business. A far cheaper and flexible way to achieve your office communications needs would be to dispense with that TDM connection entirely and use that data channel as conduit for all of your communications.

SIP trunking allows you to do that. By paying for a SIP trunk connection, you don’t need to have any kind of phone line hooked into your office. You just pay for a SIP connection from the SIP trunk provider of your choice, and your SIP-compliant software has all the telephony connectivity it needs.

One of the advantages of this model is that you aren’t tied to just a number in your local area or country. Given that SIP abstracts away the concept of where you are, it is easy to have numbers in multiple countries that feed back into your central office. This gives small companies the ability to seem like much larger companies, something from which my company, which is extremely small, could certainly benefit, but from which even home users can benefit. SIP trunking also allows you to escape reliance on the owner of the phone or cable lines that run into your home for voice service, moving beyond cable-based VOIP services, and even consumer services like Vonage, to allow a web of VOIP providers to compete for your home phone business. Though that wasn’t the emphasis of the conference, which still focuses on the needs of the enterprise and call centers, the writing is clearly on the wall. SIP trunk connections are likely to find their way into end-users homes once businesses - a group that serves as the canary in the mine for most new technologies - have worked out the kinks.

Tomorrow, I’ll talk about a product I saw at the conference that, at first blush, I wasn’t sure I would find interesting when first invited to speak to the responsible parties. That product just so happens to come from Microsoft…and no, it isn’t Office Communications Server (OCS).

John Carroll has delivered his opinion on ZDNet since the last millennium. Since May 2008, he is no longer a Microsoft employee. He is currently working at a unified messaging-related startup.

Siemens OpenScape offers tight integration of voice, video and messaging

2010-01-06T00:13:00.001+01:00

Siemens Communications is tightly integrating voice, video and messaging in its new software-based unified communications product.

The OpenScape Unified Communications Server is an IT application that runs out of a company's data center over an existing IP network utilizing existing data infrastructure. Mark Straton, senior vice president of enterprise systems marketing for Siemens, said the company has recognized that the days of voice as a standalone technology in the enterprise are over. To break out of that silo, Siemens is transforming itself into a software company.

"Our view is that voice as a standalone category will be eliminated, and it's really about delivering communications as a software and the services required to support that," Straton said.

Henry Dewing, principal analyst with Forrester Research, said Siemens' transition to a software-based communications vendor is not unique.

"Avaya and Nortel have sort of valid software strategies as well," Dewing said. "The key thing here is pushing what's been purpose-built hardware onto general-purpose computing platforms with software to drive the flexibility and functionality of the system. I think it's absolutely where the industry needs to be going."

The OpenScape server has three major applications:
More on Unified Communications Platforms
Cisco is top Unified Communications Choice, UC spending rises

Building a UC strategy

How service-oriented architecture enables unified communications

OpenScape Voice, which is really a renaming of Siemens' existing HiPath 8000 VoIP product, which has been rolled into the OpenScape brand.

OpenScape UC, a messaging infrastructure that runs on Linux and can operate in a standalone mode.

OpenScape Video, a new high-definition videoconferencing portfolio that can also run independently. The video portfolio includes room-based high-definition videoconferencing as well as lower-quality desktop video.

"The big deal for them in the announcement that they're moving to including video, mobility and other functionality such as unified communications that puts them a little bit ahead of some of their competitors," Dewing said. "Nortel or Avaya has to set up partnership deals in order to deliver those types of functionality."

The transition from hardware- to software-based UC gives enterprises a lot more flexibility, according to Brent Kelly, senior analyst with Wainhouse Research. Companies no longer have to rely on a single vendor for hardware and software, so they can avoid having their voice and other communications siloed within the company.

"It makes it easier for people to integrate this in with Web 2.0 and those kinds of things," Kelly said. "It makes it easier to integrate it with other line-of-business applications, and Siemens' architecture is really good for doing that."

Observers were particularly interested in the videoconferencing piece of OpenScape.

"It's tied into the unified communications server, where all the presence capabilities and the user interface you would have on UC softphones you will also have on the video side," said Nick Lippis, principal of The Lippis Report. "They did a nice job in having the same control interface for both video and voice, where you see a lot of companies offering separate video technology that's not integrated."

Siemens is hoping that the OpenScape launch will help it expand its North American market share. Observers agreed that the new offering puts the company on the leading edge in unified communications, but some wondered whether that will be enough.

"It just depends on how they execute on this,"Kelly said "The products are good. There's no question about it that they're good. But there's also a lot that goes into making these solutions work. All the supporting things that a company needs to get to market are all going to have to fire well in order to be successful."

Kelly said Siemens, a European-centric company, will face some challenges in North America. Recent news that the company would shed 6,800 workers through layoffs and a sell-off of factories will make some buyers apprehensive. Siemens said the workforce reduction is part of a restructuring plan that coincides with its transformation into a software communications company.

"It's clear … [the layoffs are occurring] because things aren't selling, and at the end of the day, that's what matters," Kelly said.

Siemens Launches OpenScape Unified Communications Server

2010-01-06T00:12:00.001+01:00

David Leach, Senior Public Consultant Siemens Enterprise Communications will be speaking on the "Microsoft's New UC Strategy:" panel at VON.x on March 18 at 11:00 AM

Siemens (www.usa.siemens.com) has launched the OpenScape Unified Communications (UC) Server, a new UC software platform. The company is marking this as a significant step as it transforms into a software-oriented company. OpenScape UC is designed to enable a comprehensive suite of UC application and initially includes the OpenScape Voice Application, OpenScape Video, and OpenScape UC Application V3.

OpenScape UC server, available for sale on April 30, 2008, enables presence, administration, session control and other shared services for the current and planned family of OpenScape UC suite of applications, which are designed to enable easy growth and expansion. New modular capabilities are enabled by simply activating the desired license keys on a user-by-user basis.

Unified Communications - think simple

2010-01-06T00:11:00.000+01:00

Business Every Time, Everywhere

Enhance business productivity by creating a Unified Workspace encompassing multiple applications, devices, networks, and operating systems. Improve integration of communications with business processes to ensure that information reaches recipients every time, regardless of their working environment or location.

* Mobility: Extend communications so that employees can do their jobs from everywhere using any device, wired or wireless.
* Collaboration: Share information via voice, video, or web conferencing.
* Security: Integrate comprehensive network security, from infrastructure through devices and applications.
* Choice: Use open standards to facilitate integration with industry-leading applications.
* Customer Service: Maximize call-center performance and customer satisfaction through application integration.

IBM Acquires WebDialogs; Launches Family of Unified Communications and Collaboration Software

2010-01-06T00:10:00.003+01:00

IBM has acquired WebDialogs, Inc., a Billerica, Massachusetts-based, privately-held provider of web conferencing and communications services, with more than 500,000 users worldwide. Financial details were not disclosed. WebDialogs offers online meeting and collaboration services that combine web and audio conferencing into one experience. Since it is deployed as a service, WebDialogs technology does not require support from an organization’s IT department.

With the acquisition of WebDialogs, IBM is adding a software as a service delivery model to the Lotus Sametime family of products, providing customers with choice and flexibility in how they buy and operate their web conferencing services. IBM will also integrate the service with its collaboration portfolio, including IBM Lotus Notes and IBM Lotus Sametime software.

IBM has also unveiled the next phase of its unified communication and collaboration strategy by expanding its IBM Lotus Sametime software into a product family that will include new telephony integration software. Lotus Sametime “Unified Telephony” software is a new IBM offering that is being developed to bring telephone communications into the business applications people use most. Based in part on elements of Siemens Openscape technology, this product will make it possible to manage communications across multiple business telephone systems and access them through virtually any application.

With the Lotus Sametime “Unified Telephony” product, users will be able to: manage incoming calls; see who is available to communicate and how; connect with colleagues faster; and connect with a variety of back-end and legacy systems. Lotus Sametime “Unified Telephony” will be designed to work in mixed-vendor environments with multiple business telephone systems, enabling businesses to provide a common look and feel for their users, regardless of back-end systems.

Định nghĩa Unified communication (UC)

2010-01-06T00:10:00.001+01:00

Các nhà cung cấp dịch vụ VoIP (như Cisco, avaya, nortel,… ) định nghĩa Unified communication là : dịch vụ VoIP với lớp cộng tác ở phía trên. Các hãng cộng tác truyền thống như IBM, Microsoft định nghĩa: UC là sự cộng tác với truyền thông thời gian thực thêm vào. Hãng mới như Google định nghĩa UC là nơi lưu các dịch vụ tầng ứng dụng mà lấy về từ web.

Theo tác giả bài báo, định nghĩa chính xác về UC là ở quan niệm của người dùng, và định nghĩa của các nhà cung cấp đều có những giới hạn nhất định. Tác giả cũng cho rằng còn sớm để đưa ra định nghĩa về UC, nhưng tin chắc vài điều sau về UC:

- Thứ nhất thoại phức tạp hơn những gì người ta thấy và nó không thể là một ứng dụng desktop. Nhưng đó không phải là điểm then chốt, vấn đề thiết lập truyền thông thời gian thực hoạt động tin cậy trong phạm vi lớn rõ ràng là một thách thức lớn hơn.

- thứ hai: dù cho định nghĩa như thế nào đi nữa thì UC phải có giao diện với wireless/mobile.

- cuối cùng: trong vài năm tới, lĩnh vực UC sẽ rất thú vị.

Chi tiết: http://www.networkworld.com/columnists/2007/022807johnson.html?zb&rc=voip_uc

6 câu hỏi với VoIP

2010-01-06T00:09:00.000+01:00

Gartner Group dự đoán năm 2008, hệ thống VoIP chiếm khoảng 97% các hệ thống được bán. Tuy nhiên có vài vấn đề đặt ra và bài báo này sẽ trả lời những vấn đề đó.

* Liệu có thể tin tưởng Microsoft về VoIP?
* SIP đã sẵn sàng cho desktop?
* Làm cách nào để điều hành công ty sử dụng Skype?

1. Liệu có thể tin tưởng Microsoft về VoIPMicrosoft đang có ý định trở thành nhà cung cấp điện thoại và tin nhắn IP lớn. Vấn đề là Microsoft cần phải tạo được lòng tin của khách hàng về tính thời gian thực và tin cậy của luồng lưu lượng thoại và ứng dụng trong các công nghệ máy chủ của hãng. Hiện nay sản phẩm Office Communications server 2007 của Mircrosoft vẫn chưa được bán trên thị trường.

Giống như các hệ thống VoIP thương mại khác của Avaya,Cisco, Nortel, Siemens, khách hàng sẽ phải mua những giao thức và công nghệ độc quyền đi kèm với OCS 2007. Khách hàng sẽ phải mua codec, mở rộng của chuẩn SIP, và hệ thống máy chủ Mediation Server.

Về vấn đề tin cậy, trong một năm trở lại đây, các nhà cung cấp dịch vụ VoIP như Avaya, Cisco, Nortel, Siemens, 3Com’s VCX đã chuyển sang các plaform khác (Linux, Sun solary) không dùng Window Server để chạy các ứng dụng IP PBX. Lý do là vì các khách hàng của những nhà cung cấp dịch vụ phàn nàn về tính bền vững của hệ điều hành Window. Họ phải thường xuyên cập nhập các bản vá lỗi.

Để lấy lòng tin từ khách hàng, Mircrosoft đã liên kết với Nortel.

Với tất cả những điều trên, tham vọng của Microsoft thật đáng để nghi ngờ.

2. SIP đã sẵn sàng cho desktop.

SIP được mong đợi là tương lai của VoIP. Nhưng vấn đề là hầu hết nhà cung cấp đều tin tưởng vào giao thức của chính công ty họ. Hầu hết các hãng như avaya, cisco, nortel, siemens đều sử dụng giao thức VoIP riêng và hỗ trợ chuẩn SIP có thể nạp vào phần cứng của họ. Thậm chí phần mềm mã nguồn mở Asterisk IP PBX cũng có giao thức riêng không phải SIP cho việc liên lạc giữa máy chủ và thiết bị đầu cuối.

Với điện thoại desktop, thì yếu tố quan trọng nhất vẫn thuộc về người dùng. Và các giao thức như Cisco SCCP, Siemens CoreNet, và một số giao thức khác vẫn là chuẩn cho IP, phone, PBX.

Tuy nhiên, yêu cầu sử dụng SIP vẫn tăng khi tích hợp đặc điểm trình diễn và đa phương tiện vào VoIP.Điều này khiến các nhà cung cấp phải tạo những mở rộng riêng hỗ trợ SIP cho phù hợp yêu cầu người dùng.

3. Làm cách nào để điều hành công ty sử dụng Skype

Skype thông báo là có 100 triệu người đăng ký và ước tính 30% số cài đặt là doanh nghiệp. Nhiều công ty đã tích hợp Skype vào hệ thống VoIP lớn hơn. Một ví dụ là công ty chế tạo đồ nội thất Eastern Accents ở Chicago và có chi nhánh ở Trung quốc.

Elvin Rakhmankulov, trưởng phòng IT của công ty muốn tạo kết nối an toàn và rẻ từ Trung Quốc với 200 nhân viên ở Chicago và các chi nhánh. Eastern Accents hiện đang sử dụng hệ thống 3Com NBX IP PBX. Khu vực Los Angeles, North Carolina sử dụng 3Com IP phone liên kết với NBX tại Chicago. Với hệ thống này, cuộc gọi sang Trung Quốc bị trễ lớn trong khi Skype hoạt động tốt, chiếm ít băng thông. Do đó Rakhmankulov đã nảy ra ý tưởng kết tích hợp Skype vào hệ thống hiện có, như vậy còn giúp cho nhân viên không phải sử dụng headphone, microphone,…

Rakhmankulov thực hiện bằng cách gắn NBX với VoSky. 3Com NBX kết nối với VoSky Exchange 900 qua 4 trung kế tương tự. Liên kết USB từ VoSky đến máy tính chạy WinXP có 4 tài khoản Skype chạy cùng một lúc. VoSky sử dụng cơ sở dữ liệu để chuyển tên các nhân viên ở Trung Quốc sang số mở rộng. Khi người dùng ở Chicago quay 8 số và 3 số mở rộng từ 3Com phone, thì sẽ tạo kết nối đến nhân viên ở Trung quốc sử dụng Skype với headset.

http://www.networkworld.com/news/2007/070507-voip-questions.html?page=1

What is the future trend of communications ?

2010-01-06T00:08:00.003+01:00

Answer:
IP communications - Everything over IP (EoIP) :D
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Everything over IP, is a collective term for the immerging culture of communicating and transporting data, of all sorts, over IP. VoIP Voice over IP is one of the first to lexically join the club. Although Post over IP (POIP), Company information, marketing and even trade over IP (HTML), News over IP, Games over IP are all embedded in the world of IP for many years.

The distance we have already come into EoIP is evident by the way we treat email and surfing as a part of life. Other medias, have yet to be marketed under the EoIP banner and like the previous technology may not be specifically associated with EoIP; TV and Radio over IP are available if under other names, Video over IP has been used for many years if under other guises used for person to person calls, conferences, eLearning, and many other uses. Books, Films and Music downloading is becoming the norm and IP internet connections are as important, to many people, as other services, electricity, gas, water, both domestically and in business. eBanking, eCommerce, stock prices and trading are all available over IP. International and inter-bank money transfers, ATM machines, credit card processing; online or in shops connect to each other over IP.

Many countries around the world are engaging in eGovernment and other types of digital citizenship. Signing documents legally with some form of digital signature will one day be the norm for everything from buying a house, getting married to voting, accessing education, taking exams and international travel. Medicine is not left out of this, there are many devices that can assist today with diagnosis, monitoring, drug administration and even surgery and this is just the tip of the iceberg.

British Telecom is spending £10,000,000,000 on a project called 21st Century Network (21CN), transforming the whole UK phone system to VoIP over the next 3 years. Leaving everyone with just a high speed DSL IP connection or an analogue voice only line (POTS), converted to VoIP at the exchange. The 21CN project will provide the UK with a pure IP communications network, another step and a mental shift in to the EoIP world.

Other countries such as Japan, Singapore and South Korea have thriving digital communities that are making the most of EoIP.

The term Everything over IP is a good way to name the direction we are going in (2007), but it is not a technology is it a buzzword or aspiration, filling in vocabulary as needed. This acronym has it is use now but will fade from the lime light when EoIP is as much the norm as email.

EoIP is of course not totally accurate until science makes some quite large jumps. It will be a long time before we have Me over IP and probably by then, if ever, IP will only be demonstrated in history museums, which you will of course visit over whatever the new communication protocol of the day is.

Defining the Impact of Microsoft Office Communications Server 2007

2010-01-06T00:08:00.002+01:00

By some estimations, a new era in the IP communications age will start next Tuesday when Microsoft (News - Alert) officially launches Office Communications Server 2007. There’s been buzz surrounding the new software, and its companion Office Communicator 2007 messaging client, for months now. With the launch now imminent, the excitement is palpable.

What exactly is Office Communications Server 2007 (OCS 2007)? That seems like a good question to answer as the software’s launch approaches. Microsoft defines the product as a solution that “delivers streamlined communications for your users so they can find and communicate with the right person, right now, from the applications they use most.”

That is a somewhat vague definition, or maybe it’s just broad without talking about the technology under the hood. One could say, without too much opposition, that OCS 2007 represents Microsoft’s push into the IP communications market—which includes unified communications,VoIP , and messaging.

OCS 2007, as TMCnet blogger Tom Keating (News - Alert) pointed out in a June 26 post, “uses Session Initiation Protocol (News - Alert) (SIP) standards-based protocol to enable presence-based VoIP call management, as well as VoIP communication.” In his post, Keating also noted that Microsoft is marketing OCS 2007 to the enterprise (big business) market.

Readers of this article probably are familiar with the term ‘presence,’ which Keating referenced in his post. Just in case, though, ‘presence’ in the communications market refers to technology that lets users see the status of other users—something along the lines of an instant messaging client “buddy list” that shows if a person is ‘available’ or ‘away,’ but more advanced in its functionality and applications.

Unified communication, another term that doesn’t always have a clear definition, generally is understood to include the concept of presence as a way to make communications more efficient and useful. For the most part, OCS 2007 is being talked about as a unified communications product, as well it should, although some analysts (like Keating) are also looking at its impact on the VoIP market.

Wikipedia describes OCS 2007 as the latest incarnation of Live Communications Server, specifying that the new solution is “an enterprise real-time communications server, providing instant messaging and collaboration functionality. It is originally a spin-off from the then new feature in Exchange Server 2000.”

The Wikipedia entry for OCS 2007 goes on to specify that the main use of the solution is “instant messaging within a single network, including presence information, application sharing, file transfer and voice and video communication.” Further, the entry points out that Office Communicator is now being promoted by Microsoft as the recommended messaging client.

In his June blog post, Keating described Communicator 2007 as a unified communications client that work in conjunction with OCS 2007.

“This solution delivers a presence-based, enterprise VoIP “softphone” for secure, enterprise-grade instant messaging that allows for intercompany federation and connectivity to public instant messaging networks such as MSN,AOL ( News - Alert) and Yahoo!,” Keating said.

So there you have it—OCS 2007 touches on presence messaging, UC, VoIP and various other aspects of IP-based communications. Its full impact is yet to be fully understood, but the response of companies like NEC and Unisys, partnering on or developing forward-thinking UC strategies, provides an indication of how significant Microsoft’s new product is perceived to be.

Stay tuned.

Điện thoại qua giao thức Internet: có hứa hẹn?

2010-01-06T00:07:00.000+01:00

Các dịch vụ thoại qua giao thức Internet được nhận định là rất có tương lai khi công nghệ băng rộng trở nên phổ cập.

Điện thoại qua giao thức Internet có cước siêu rẻ, thậm chí miễn phí qua WiFi, WiMax, đã khiến một số nhà khai thác viễn thông lo ngại.

Tuy nhiên, cũng có ý kiến cho rằng, đây chỉ là những dịch vụ bổ sung cho các nhu cầu sử dụng khác nhau và những dịch vụ thoại qua giao thức Internet vẫn còn rất nhiều hạn chế nên chưa thể đe dọa đến các nhà khai thác viễn thông.

Mối đe dọa tới gần?

Mặc dù đã được cấp phép cung cấp một số dịch vụ viễn thông, song FPT Telecom chưa đủ tiềm lực để xếp mình vào danh sách các công ty viễn thông, mà vẫn chỉ được biết đến như một nhà cung cấp dịch vụ Internet (ISP). Ở cái thế “chả có gì mà mất”, FPT Telecom đã tiến sang một công nghệ khác mà giới phân tích cho rằng sẽ là mối lo ngại cho các nhà cung cấp dịch vụ viễn thông, đó là công nghệ SIP. Trả lời Báo BĐVN, bà Chu Thanh Hà, Phó giám đốc FPT Telecom cho biết hiện công ty này đang thử nghiệm mạng SIP (nhưng chưa có kết quả gì!). SIP (Session Initiation Protocol -Giao thức khởi tạo phiên) là một giao thức tín hiệu điện thoại IP dùng để thiết lập, sửa đổi và kết thúc các cuộc gọi điện thoại VoIP. Chuẩn này có khả năng thiết lập và truyền tín hiệu các cuộc gọi trong mạng Internet, tương tác theo thời gian thực, có thể xử lý thông tin trong cấu trúc mạng phức tạp mà mỗi phiên có thể là một cuộc gọi điện thoại 2 chiều, một thông báo các tin nhắn...

Giới chuyên môn nhận định, SIP sẽ có vị trí vững vàng trong công nghệ không dây và mạng di động thế hệ thứ ba (3G). Cisco và nhiều hãng sản xuất thiết bị tổng đài IP đang đưa SIP vào phần cứng, còn Skype, Yahoo, Microsoft… cũng đã tích hợp SIP trong hệ thống tin nhắn nhanh của họ. Phía FPT Telecom cũng thông tin rằng với mạng SIP kết hợp với dịch vụ Internet WiFi đang được lắp miễn phí ở nhiều nơi thì những người dùng máy tính xách tay, điện thoại di động có thể gọi điện thoại miễn phí được. Những dòng máy ĐTDĐ sử dụng HĐH Symbian9, Windows Mobile 2003/2005 ngoài sử dụng dịch vụ thoại của các nhà cung cấp viễn thông còn có thể kết nối được với SIP để “alô” qua Internet với giá rẻ, thậm chí miễn phí. Trong khi đó, FPT đang mở rộng mạng WiFi tại nhiều trung tâm lớn ở Hà Nội và TP. HCM.

Một số chuyên gia trong lĩnh vực viễn thông của VNPT còn cho rằng, không chỉ có công nghệ SIP đe dọa đến các dịch vụ truyền thống, mà khi Việt Nam cấp phép WiMax sẽ có hàng loạt các dịch vụ gọi điện thoại miễn phí trên mạng băng rộng này như Skype chẳng hạn. Với ưu thế truyền dẫn Internet băng rộng và có tính năng di động, chắc chắn sẽ có hàng loạt các dịch vụ qua giao thức Internet sẽ được cung cấp trên nền mạng này. Lúc đó, các thiết bị hỗ trợ WiMax di động sẽ được người sử dụng có thể sẽ lựa chọn các dịch vụ qua giao thức Internet nhiều hơn vì việc kết nối băng thông rộng sẽ đem lại dịch vụ chất lượng tốt hơn và có khả năng di động nên sẽ khắc phục được nhiều nhược điểm của dịch vụ này hiện nay.

Mới đây, hãng Nokia đã tung ra dòng máy Nokia N800 hỗ trợ WiMax và có thể thực hiện điện thoại Skype. Giới phân tích cho rằng, hiện thị trường VoIP còn nhỏ, thế nhưng đây vẫn sẽ là mối đe dọa trong tương lai. Chẳng hạn, nếu như dùng Skype, khách hàng sẽ có thể sẽ bỏ điện thoại cố định hay di động mà cần cài đặt phần mềm Skype vào máy tính, mua một cái micro và tai nghe, thế là có thể dễ dàng biến PC thành điện thoại và mới đây công nghệ này đã được tích hợp tất cả trong chiếc điện thoại.

Các nhà khai thác viễn thông nói gì?

Tỏ ra khá lạc quan về tính cạnh tranh của mạng di động GSM, ông Phan Hữu Châu, Trưởng phòng Kỹ thuật của Vinaphone cho biết, với các công nghệ thoại mới trên giao thức Internet như SIP sẽ bổ sung cho các dịch vụ mà khách hàng tùy theo nhu cầu để sử dụng chứ không thể đe dọa đối với dịch vụ di động hiện nay. Trong khi đó các dịch vụ này còn nhiều nhược điểm và phạm vi sử dụng còn hạn chế. Ông Phan Hữu Châu còn cho rằng, trong tương lai, các nhà khai thác viễn thông sẽ tiến lên mạng băng rộng. Lúc đó tỷ lệ thoại trên mạng băng rộng nhỏ, thậm chí các nhà khai thác viễn thông có thể cho miễn phí thoại mà chỉ tính tiền dịch vụ phi thoại.
Khác với ý kiến của phía Vinaphone, ông Hoàng Sơn, Phó giám đốc Viettel Telecom cho rằng, những dịch vụ thoại miễn phí qua giao thức Internet có thể đe dọa đến các nhà khai thác viễn thông. Tuy nhiên, mối đe dọa như thế nào còn phụ thuộc vào chính sách quản lý. “Ở những nước phát triển có vùng phủ WiFi miễn phí, người sử dụng có thể sử dụng miễn phí các dịch vụ thoại qua giao thức Internet. Nhưng ở Việt Nam vùng phủ sóng WiFi hẹp và các nhà khai thác viễn thông đang trong quá trình đầu tư mạng lưới nên việc triển khai miễn phí các dịch vụ thoại qua giao thức Internet một cách rộng rãi vẫn là câu chuyện xa”, ông Hoàng Sơn nói.

Voice over IP: 10 Reasons!

2010-01-06T00:06:00.000+01:00

Reason to switch to using the model of phone Internet protocol (Voice over Internet Protocol - VoIP) can be mentioned a lot, depending on whether you want to develop business models and operations of the business ( DN) how you. However, according to documents from Customer's Avaya corporations, is 10 the most important reason to use the phone system (phone) IP (IP Telephony) or VoIP service.

Transition strategic direction of service providers and VoIP

According to many experts, the coming few years will be about 300 billion U.S. dollars of investment funding for telecom industry was transferred to the equipment and network services to support IP Telephony solutions over LAN and the VoIP service package on the WAN. Thanks to appear at the right time, network technology to support IP Telephony and VoIP has influence and impact a strong market to provide telecommunications services.

Diverse functions - high economic efficiency

Most of the features of the phone system Phone POTS-PSTN (Plain Old Telephone Service - Public Switched Telephone Network) has made a tradition of "name" in the field of telecommunications and became very familiar to us. The outstanding features as possible to the voice mail (voicemail), transfer calls between your branches (call tranfer), call forwarding to some other phone (call forwarding) and the call triangle (three - way calling). Cost [the investment or use] of these features can be included in the cost of [the investment or use] of the enterprise phone system, or user fee for using these features as form of separate services are integrated on individual subscriber Tel.

Obviously IP Telephony and VoIP have made phone features traditionally become obsolete as all features, applications communicate new information has presented the world converged telecommunications platform support IP. Number and type of phone features available in the IP Telephony and VoIP solutions today are rated as very much and attractive. All this functionality is available without requiring any additional investment costs because they do work on IP network platforms and be "shipping" on your computer medical computer applications as usual.

These features are effective in IP Telephony and VoIP solutions - such as the ability to check the station phone you are using (or exactly are connected to) and get a "virtual sign" through which inform you that someone in the group calls its "presence" but is busy answer / make a call or away - have proved so superior to any system that features POST Phone System provides.

POTS phone system, separate systems must always be maintained to manage existing user on the system and information about them. Depending on the POTS phone system that businesses are using, the manager must set the user's information and update data on selected list of phone systems.

Meanwhile, the telecommunications system converged IP platform, most (if not to say all) information as may be updates only once and stored (maintained) in an institution homogeneous data. Whether companies are using ERP (Enterprise Resource Planning) or DBMS (Database Management Systems), the software is integrated with all related application systems in enterprises when necessary, including all systems IP Telephony and VoIP today through communication Application Programming Interface (API).

Investment cost savings VoIP ago

Today, most organizations and companies are using (or also more) model of the traditional phone system, or converted in whole or in part to IP Telephony and VoIP systems to support better for business. If the company has equipped digital devices (eg PBX), you can save considerable investment costs by reusing most of the equipment with new VoIP system.

Easy maintenance

Thanks to the ability to remove root systems get more information duplication and redundancy, the task of installing and managing VoIP and IP Telephony systems becomes easier. The management system phone offered many applications efficiently and directly to support the many challenges they encounter. They not only manage the application data on computer systems but also phone or video conferencing (video conference) on the IP platform.

The action moves, additions and changes (user information or system configuration) in the POTS phone system requires resources previously complex and very expensive, but with an IP Telephony network is VoIP automatically self-adjust to be compatible with the new location of the user. System managers can be from any networked computer that conducted the inspection setting purposes (usage), account (account) and other data users. With the IP telecommunications systems, public management and maintenance of the network phone becomes economically and efficiently.

Flexibility and mobility

IP Softphone is a software solution called Internet Phone for computers using Windows operating system. This software allows users easy access to telecommunication services in real time (with or without charge), and features improvements to the method used is simple: just click to dial call. In a telecommunication network IP platform, employees can travel freely anywhere in the office, connect the laptop to the network, start working and receive / make calls. Whether sitting in a position "temporarily" but people still provides full features available in computers at the main desk. Network will automatically identify the user and impose personal information (profile) of the user that the database's control system. Even employees can transfer calls to any phone on any desk in a temporary location (this phone is not necessary to support IP). With this model, the system administrator no costly and time to create compatibility between computer data and phone connections for a colleague is working temporarily at the office by their charge.

Many attractive features

If you are a manager (network computing or telecommunications in business) and still skeptical about the advantages of VoIP, such as reduced communication costs; easy integration of data systems, voice and video; database capable of centralized control; mobile features improved help save time and costs ... probably not something you can convince to switch to using the solution hey.

But just in case you face a barrier and not sure which way should you consider the phenomenon of alcohol emerged as "the SIP (Session Initiation Protocol) - allows multiple applications , new hardware easily deploy VoIP solutions on the LAN, WAN or Internet. Most of the current ADSL modem and router support VoIP and SIP, small and medium enterprises can quickly deploy Internet phone models through Internet connection with other free VoIP services such as IPTel, DrayTel, MediaRing, IPTEL, Voice777 ...

Ability to manage comprehensive

The telecommunication network provides the IP is always a platform for effective comprehensive management system. Therefore, you can control information to each bit of data flow on the network is IP Telephony (LAN) or VoIP network (WAN).

Similarly, you can use specialized tools, such as diagnostic tool system of Avaya EXPERT Systems Diagnostics, to quickly detect and correct these problems occur on the network. These tools also support monitoring site or remotely. In this dedicated network (the IP telecom network), voice quality can reach 99.99%. This number does not mean "problems never happen" but the environment the IP network convergence, the ability to detect early symptoms and change the settings before you have any problems do occur that has been improved significantly.

Collaboration time

If you are still skeptical about the benefits that IP Telephony and VoIP delivers, please consider the following fact: VoIP platform operating on IP networks and web applications previously only works on the Internet is now can operate on the IP network. Users can access the website you need right from the desktop phone or the IP link to the homepage of the particular phone being used. Currently, there are many Web-based applications-HTML suitable for operation on the IP phone.

Users can add images Solutions Phone Video Telephony Solution with software-based IP video applications, thereby allowing a desktop or notebook computer to simulate a IP phone in the office. Quality images and sound using the internal network usually better to use the internet connection is rarely encountered condition late hours or frame.

Bandwidth use reasonable

Many users completely wrong that the additional IP Telephony and VoIP solutions to the enterprise network computer network is not providing enough bandwidth to support this change. However, the fact that data flow computer (not a data device) on the regular phone network, accounts for 30% of the overall system bandwidth.

If the review terms of LAN, the ability to isolate failures is provided through switches, to help maintain a stable regime for the operation of the network. If any "dead spots" (chokepoint) are discovered, they can be overcome almost immediately by changing the point of connection or the network managers called "load balancing "(load balancing). However, the IP management system will tell you this before it becomes a problem.

If the review terms of WAN, the ability to download data (bandwidth) should be more focused. WAN bandwidth is often evaluated based on how many standard Service Channel Number (Digital Service Channels - DSOs) are supported.

For example, the transmission line T-1 (1.544Mbit / s) support 24 DSOs is one of the most common connection in the world today. If the phone system deployed in the form of POTS-PSTN switch via T-1 link, you can make 24 calls simultaneously. However, the advantages of VoIP is closing the data packets and data packets are transmitted via T-1 connection on a small portion of the bandwidth of a channel DSO. Therefore, you have the bandwidth of multiple VoIP services if compared with POTS-PSTN phone system on the WAN.

Reduce long-distance phone charges

If a company has many offices away from each other (in the cities, provinces or other countries), the VoIP will save considerable cost long distance phone calls between this office. All that companies need to do is adding hardware solutions and software appropriate to network your computers available. For a complete VoIP system, even businesses can deploy solutions intelligent call forwarding, which allows "limited" long-distance calls directly from a region (province, city or country ) to a different area (the area must have two offices of this company) by transferring data through the phone internal VoIP network and then forwarded to the PSTN system.

VNPT cung cấp dịch vụ giải pháp mạng điện thoại nội bộ - IPCentrex

2010-01-06T00:05:00.000+01:00

VNPT chính thức cung cấp dịch vụ giải pháp mạng điện thoại nội bộ IP Centrex dành cho các tổ chức, doanh nghiệp. Đây là giải pháp mới của mạng NGN cho các ứng dụng tương đưng dịch vụ tổng đài nội bộ doanh nghiệp. Theo đó, các doanh nghiệp có nhiều văn phòng, chi nhánh ở các địa điểm khác nhau trên toàn quốc, khi sử dụng IP Centrex sẽ thiết lập được một mạng riêng cho tất cả văn phòng, chi nhánh của mình cũng như sử dụng các dịch vụ, ứng dụng mới linh hoạt trên nền IP.

Với dịch vụ IP Centrex, các nhân viên có thể làm việc ngay tại nhà riêng hoặc ở bất kỳ nơi nào mà vẫn tương tự như đang ở cơ quan. Từ một địa điểm, mỗi nhân viên chỉ cần bấm máy lẻ để liên hệ trực tiếp với các chi nhánh, văn phòng khác trên khắp lãnh thổ Việt Nam. Thông qua mạng Internet/VPN/leased line, các nhân sự này chỉ cần dùng username đăng nhập cùng mật khẩu là có thể thực hiện được các cuộc gọi.

Tùy từng cơ cấu tổ chức của mỗi doanh nghiệp, VNPT sẽ thiết lập cấu hình cho từng doanh nghiệp: số lượng thuê bao, tính năng sử dụng... giữa các chi nhánh khác nhau cũng như việc đặt tên hiển thị, chuyển cuộc gọi... và tạo một kế hoạch đánh số chung cho tất cả các chi nhánh của doanh nghiệp. Các doanh nghiệp lựa chọn loại hình tổng đài này sẽ không cần đầu tư tổng đài PBX, không cần người quản trị hệ thống bởi công việc này sẽ do Công ty Viễn thông liên tỉnh VTN trực tiếp quản lý và đảm nhiệm. Đối với khách hàng sử dụng dịch vụ trực tiếp, cần phải có mạng máy tính nội bộ được kết nối đến hệ thống của VNPT bằng VPN hoặc có kết nối Internet.

Dịch vụ IP Centrex được VNPT cung cấp trên phạm vi toàn quốc. Khách hàng có thể xem chi tiết tại website:http://www.vtn.com.vn/IPCentrex hoặc gọi tới số điện thoại hỗ trợ miễn phí 18001719 để biết thêm thông tin.

SIP and Web-Based Applications

2010-01-06T00:04:00.000+01:00

Doug Tucker, CTO Americas, Ubiquity Software
By Tim Scannel, Internetnews.com

Since its founding in 1993, Ubiquity Software has pretty much lived up to its moniker of developing applications and services that are available anywhere and everywhere.

In 1999, the company revamped its strategy to produce software based on the Session Initiation Protocol, or SIP. The technology was originally designed to be a framework for the development of converged voice, video and data services over IP networks, and has since become a key element in telecom-class communications services today.

The market for such services, based on SIP, is expected to reach $4.7 billion by 2009, according to market researcher Yankee Group, as a range of new applications and products is deployed that make use of enhanced IP services. These also include VoIP, Web conferencing, instant messaging and gaming applications.

In January, the UK-based Ubiquity agreed to be acquired by Avaya, a move that gave this company an edge over Nortel, Cisco and other competitors. And just this month, Avaya agreed to an $8.2 billion acquisition deal put on the table by private-equity firms Silver Lake and TPG Capital, which effectively makes it a private corporation.

Ubiquity has emerged as a key element in Avaya's plans to expand its IP-based telephony footprint and spearhead the development of converged voice and data applications and services -- especially those offered by the wireless carriers.

Recently, internetnews.com talked with Doug Tucker, CTO of Ubiquity's Americas operations, about its evolving SIP platform and how the technology will play a key role in the IP-based applications landscape.

Q: Why is SIP important to current and future Web-based applications?

You can think of SIP as a generic interconnect mechanism for communications sessions. It's not limited to voice or IP telephony. In fact, IP telephony is an application kind of superimposed on top of the SIP interconnects. It is a way to establish sessions between communications participants and then negotiating different ways communications sessions run on top of it.

Q: Are sessions of this type possible without a SIP protocol or architecture?

There are other protocols you could use, like H.323, which has been the predominate protocol that enterprises have used in the past for voice and video conferencing and collaboration. Actually H.323 and T.120 have been the predominant ones.

These are now migrating to a SIP infrastructure. There are still H.323 products in use today, but clearly the future is moving towards SIP.

Q: What is the problem with H.323 as compared to SIP?

H.323 is a very constrained infrastructure. It was designed for very specific deployment architecture and specific uses, while SIP is a much more open protocol that allows the applications to define the usage and the topology that SIP is applied to.

Q: So would the problem with H.323 and others be that they were essentially developed before the development and growth of mobile devices and advanced Internet applications?

I was actually on the committee that created H.323 10 or 12 years ago. It was really targeted as an IP-modeling of the PSDM (perceptual speech distortion metric) infrastructure, and was strictly designed for point-to-point multi-point video and voice communications.

It wasn't viewed as a general applications infrastructure, where SIP was really built around the Internet protocols and a very distributed deployment model and also a very open usage model. It's much better suited to today's view of where applications are going, which are rapidly breaking down the notion of vertical applications and approaching more of a horizontal services capability that are more dynamically grouped into point applications for the enterprise.

Q: Are there various types and iterations of SIP, much like there are different types of Unix, Java and other platforms?

There is one SIP set of standards that people adhere to, but it's a very flexible base. On top of that you layer in what I call your "service topology," and that's how you make use of SIP to make the connection session and to define what the sessions are. And those service definitions are what changes from deployment to deployment.

Q: Ubiquity's SIP is a Java-based platform. Does this make a big difference as compared with something that may be based on Brew, or does Brew have a comparable platform?

First of all, there is the SIP infrastructure, and on top of that you need to layer an environment that allows you to write programs and make use of it. The Ubiquity SIP platform uses Java, and more specifically a SIP server container that is equivalent to a Web server container. So, the programming model on top of that matches Web developers who are creating Web portals or Internet-style applications.

Something like Brew is a more closed environment that is really built around the notion of a monolithic vertical application development, and it has its own programming environment. As a result, you would have to retrain your redevelopment community around that infrastructure.

Q: One trend in mobile applications development today is to put more code and operational rules on the server side and less programming on the mobile device itself. This not only conserves real estate on the device, but provides more security should the device be lost. What is your opinion on this approach?

There is a development model that I support of not putting a lot of intelligence on the endpoint. It is definitely a growing trend in the industry.

When you look at how applications are built, both carriers and enterprises are looking toward an SOA-based hosting and development model. And in that model, the endpoint participation is evolving to the point where the endpoint itself can host its services into the SOA environment.

But those services aren't necessarily a part of the application, but they may be elements that the application can attach to and make use of in the context of the subscriber session with that application.

Q: Are most carriers up to speed on using SIP, or is there a ways to go before it is firmly implanted and used as a matter of course?

I think SIP has clearly won the mindshare in terms of when people are implementing IP as part of their telephony infrastructure. It's just assumed that it's SIP.

The latest evolution in thinking is now modeling the call model and operational model of a network more closely to how SIP operates as opposed to how they were operating as a PDM infrastructure.

Q: A couple of years ago CERT questioned the security of SIP as an infrastructure because it made use of the IP architecture. Has that situation changed today?

I think we are well on our way to solving the security issues. The issues are understood, and it's not just related to the protocol. The protocol itself doesn't take care of security. It's really edge devices that have to be in the frontline of security in the service network.

Q: Does this mean that applications developers using SIP must be very aware of its characteristics and what it can and cannot do in terms of security as they build applications for service networks?

Not anymore. The growing trend is that applications developers really work at the applications layer, which is well buffered from the service network. They may not even see SIP at all. And security is a layer underneath the SIP layer.

As everything moves closer to a Web services and a service-oriented architecture approach, applications developers will be able to be highly abstracted from the service definitions underneath the application and the endpoints as well as any of the issues around security.

Q: What do you view as he biggest roadblock to the continued drive toward SIP and the development of SIP-enabled applications?

I think the next evolution has to be at the applications layer. Service-oriented architectures are going to be the new arena over the next couple of years, and SIP and HTTP are underlying access technologies and transports for applications sessions that are being defined at a very high level through SOA techniques.

It's not necessarily SIP itself that is going to evolve significantly, but the number of rich applications is going to grow and drive more utilization of SIP.

SIP to become mainstream by 2010

2010-01-06T00:03:00.002+01:00

Session Initiation Protocol services coming to a phone near you.

Session Initiation Protocol (SIP) services will become the norm after 2010 and will rapidly begin to dominate the world's telecoms markets, industry experts predict.

ABI Research expects that almost half of all telecoms customers will be using at least one SIP-based service by 2012, and are likely to have many services from multiple devices able to communicate across the web.

This will generate over $150 billion in service revenues annually by 2012, with cumulative infrastructure capital expenditure of over $10 billion by that date, according to the analyst firm.

ABI's Worldwide SIP Services Market paper noted that the move to all-IP networks is the chosen path for introducing new services, with next-generation networks as the ultimate goal.

"The path to this all-IP goal is complicated. Migrating existing services onto IP networks while retaining resources until they can be taken out of service is not a straightforward process," said ABI analyst Ian Cox.

"Mobile operators' voice services are already optimised to reduce network traffic, and the move to VoIP is not an easy choice until the introduction of LTE or HSPA. One method that has gained some momentum is to use an IP overlay based on SIP."

Cox explained that this approach allows new services to be designed and launched using a well-supported standard.

"Using SIP, telephony becomes another web application which can be integrated into other internet services," he said. "It allows service providers to build converged voice and multimedia services."

ABI expects almost 1.2 billion VoIP users to be active by 2012, and that most users will also subscribe to several forms of messaging and video sharing driven by the interest in user-generated content.

Additional services supported by SIP will include "core" items such as presence, click-to-dial, buddy lists, e-mail and web access which will be included as standard in any service offering, and bundled with broadband access.

A portion of the VoIP users will also be connected to a fixed-mobile convergence service.

Các "ông kẹ" viễn thông dòm ngó tương lai của VoIP

2010-01-06T00:03:00.001+01:00

Hầu hết các mạng di động đều coi VoIP như kẻ thù, nhưng mới đây, Deutsche Telekom - mạng di động lớn nhất nước Đức lại công khai tuyên bố sẽ mua lại cổ phần của Jajah, một hãng công nghệ điện thoại VoIP mới thành lập và đang được Intel chống lưng.

Công nghệ của Jajah cho phép người dùng sử dụng điện thoại cố định hoặc di động để thực hiện các cuộc gọi VoIP giá rẻ, thậm chí là miễn phí. Trong giai đoạn hợp tác đầu tiên, T-Online Venture Fund, công ty đầu tư tài chính trực thuộc Deutsche Telekom sẽ rót 20 triệu USD cho Jajah.

Đây là dấu hiệu rõ ràng cho thấy các ông lớn trong làng điện thoại đang dần thay đổi quan điểm: biến VoIP từ địch thành bạn. Trước đây, các hãng điện thoại có vai có vế, nhất là tại Mỹ đều hướng ánh mắt thù địch đến Skype và Vonage (một hãng biến kết nối băng thông rộng thành đường điện thoại), e sợ họ sẽ đe dọa công việc kinh doanh truyền thống của mình.

Thậm chí Verizon Communications còn đi xa tới mức khởi kiện Vonage, đòi hãng này phải "bồi thường tổn thất lên tới 58 triệu USD".

Tuy nhiên, tình hình chiến sự căng thẳng có vẻ như sắp được xoa dịu. Jajah sẽ trở thành hãng VoIP đầu tiên nhận được sự ủng hộ công khai, chính thức từ một nhà cung cấp dịch vụ điện thoại nổi tiếng.

Giới phân tích đều nói rằng họ không ngạc nhiên khi "bày tỏ thiện chí kết bạn đầu tiên" là một doanh nghiệp đến từ châu Âu, nơi thị trường viễn thông cạnh tranh hết sức gay gắt. Chỉ có điều không ai dám chắc các mạng điện thoại tại Mỹ có "học tập theo" hay không.

"Sự thay đổi quan điểm này rất đáng được ghi nhận. Các đại gia viễn thông nhận ra rằng họ phải học hỏi nhiều điều từ những hãng công nghệ mới nổi như Jajah. Nhưng các hãng điện thoại của Mỹ thì không cởi mở được như vậy, họ vẫn nghĩ VoIP là thủ phạm khiến doanh thu đàm thoại sụt giảm", Giám đốc nghiên cứu Will Stofega của IDC cho biết.

Bạn và thù

Thực ra, thị trường VoIP hiện nay còn tương đối nhỏ, nhưng các mạng viễn thông vẫn coi đây là mối nguy canh cánh bên sườn.

Hệ quả là họ buộc phải mở rộng lĩnh vực kinh doanh của mình, chi hàng tỷ USD để nâng cấp mạng mới, hỗ trợ những dịch vụ hiện đại như Truy cập Internet tốc độ cao hay truyền hình. Tuy nhiên, doanh thu từ những dịch vụ mới này chỉ là hạt cát so với cước phí thuê bao hàng tháng.

Ngay cả dịch vụ điện thoại cơ bản nhất (không bao gồm tính năng chờ cuộc gọi, gọi 3 bên song song hay tin nhắn thoại) cũng có mức cước khoảng 20 USD/tháng. Các mạng điện thoại ung dung bỏ túi số tiền này mà chẳng cần phải làm gì ngoài việc cung cấp một đầu số lúc đầu.

Nhưng nếu như dùng Skype, khách hàng sẽ có thể xếp xó điện thoại cố định. Họ chỉ việc cài đặt phần mềm Skype vào máy tính, mua một cái micro và tai nghe, thế là có thể dễ dàng biến PC thành điện thoại rồi. Tương tự, khách hàng của Vonage có thể cắt bỏ hoàn toàn đường điện thoại gia đình của mình, biến kết nối Internet băng thông rộng thành đường dây liên lạc chính của cả gia đình.

Tuy nhiên, ưu điểm nổi bật của Jajah là vì công nghệ của hãng này rất "thân thiện với các hãng viễn thông". Nó cho phép người dùng sử dụng điện thoại hiện có (cả cố định lẫn di động) để thực hiện các cuộc gọi Internet giá rẻ chứ không bó hẹp trong máy tính như Skype.

Bạn chỉ việc truy cập vào website của hãng, nhập vào số điện thoại cần liên lạc và số điện thoại của mình là Jajah sẽ kết nối cho cả hai. Dịch vụ của Jajah sẽ tìm kiếm đường link IP rẻ nhất và đáng tin cậy nhất để truyền tải tín hiệu cuộc gọi. Hiện tại, nó đang phủ sóng ở hơn 55 quốc gia với mức cước vẻn vẹn 3 cent/phút.

Một cuộc gọi đường dài tiến hành qua Jajah có thể rẻ hơn bình thường tới 80%, và đó là một tỷ lệ mà không ai có thể thờ ơ.

Tất cả những khía cạnh đó đủ để thuyết phục Deutsche Telekom ký vào bản hợp đồng đầu tư. "Jajah không phải là một nguy cơ, vì người dùng vẫn cần có điện thoại cố định hoặc di động mới dùng được dịch vụ của hãng", Giám đốc điều hành Georg Schwegler của T-Venture Holding cho biết.

Microsoft: "You can make money from VoIP rates"

2010-01-06T00:02:00.000+01:00

Software giant Microsoft said that the businesses and organizations switching to VoIP can help airlines earn more "billions of U.S. dollars" sales each year.

Trends switching to Internet telephony will accelerate in the next 3 years, and new VoIP server software from Microsoft will "contribute to changing the face of telecommunications industry, like the way the Windows operating system has changing computer industry, "said Jeff Raikes, president of Microsoft Business Division said.

"We will see historical repetition," said Raikes spoke before the audience attending VoiceCon Spring 2007.

In fact, business strategy is completely different from Microsoft VoIP with Cisco Systems. While Cisco offer the complete integrated system (the full network device + software separately), then Microsoft rely on the "distribution" of the computer industry, where software central role .

Last year, Microsoft and Nortel have signed important cooperation agreements on VoIP. According to Raikes, VoIP software market will grow to a global turnover of 35-40 billion dollars in the future, and Microsoft to constitute a part of that pie.

Office of the heart

Currently, the number of business VoIP technology applications are increasing, mainly to save costs. Yankee Group estimates that operates up to 82% of U.S. companies currently use VoIP system. Cisco, Microsoft and Avaya are the three major competitors in the enterprise VoIP market, Skype is still holding throne unique position of civil VoIP market.

Raikes predicted that the cost of installing a VoIP phone system will reduce to 50% within three years, and this stimulates the business sector as to acquire this technology faster.

Microsoft's VoIP strategy, the Office business software role as a heart. It will open up a communication net nhuat environment where employees can be "moved" between email, IM chat, phone and Web phone wireless seamlessly, quickly, simply.

Microsoft predicted that over 100 million business users (large scale than the current market) will make VoIP calls through software Office for 3 years. The airline plans to provide a demo of Office Communications (ie the best platform office software, servers and information) at the end of this month.

Google mua lại dịch vụ điện thoại Internet

2010-01-06T00:01:00.002+01:00

Gã khổng lồ tìm kiếm vừa thâu tóm GrandCentral Communications, một doanh nghiệp vừa mới thành lập với công nghệ cho phép người dùng quản lý tất cả các số điện thoại cũng như hòm thư thoại qua mạng Web, cứ như thể chúng là một tài khoản duy nhất.

Đặt trụ sở tại California, Grand Central chỉ là một trong hàng chục hãng công nghệ mới, nhăm nhe khai phá mảnh đất dịch vụ Web và điện thoại VoIP màu mỡ.

Tuy nhiên, hướng đi của Grand Central khá giống như Jajah, khi cho phép người dùng cá nhân và doanh nghiệp gọi điện qua mạng Internet, song vẫn tận dụng được điện thoại truyền thống.

"Bạn sẽ có một số điện thoại duy nhất, forwards các cuộc gọi tới cho tất cả các máy điện thoại mà bạn đang sở hữu trong tay", Craig Walker và Vincent Paquet - hai đồng sáng lập ra GrandCentral cho biết.

Dẫn đầu thị trường VoIP hiện nay là Skype, với hơn 200 triệu người dùng trên toàn thế giới. Một số cái tên khác mới hơn nhưng cũng đã thu hút được hàng triệu người dùng trong suốt năm qua là Jajah, Jangl, Jaxtr và Rebtel.

Trên thực tế, ý tưởng về GrandCentral nảy ra trong đầu Walker sau khi anh này hạ cánh ở một sân bay địa phương. "Tôi cần phải kiểm tra tới 3 hộp thư thoại - một của điện thoại di động, một ở chỗ làm và một của chiếc máy BlackBerry", Walker kể lại.

"Bất cứ ai sở hữu nhiều số điện thoại khác nhau (nhà, chỗ làm, di động) đều phải trải qua những phiền toái này. Nếu như họ có một số điện thoại mà cả ba cái máy cùng rung chuông thì sẽ tiện biết bao", Walker viết trên blog cá nhân của mình. "Bằng cách này, số điện thoại của bạn sẽ luôn dính chặt lấy bạn như hình với bóng, bất kể là bạn đang ở đâu".

Hướng kiếm tiền mới

Thay vì cạnh tranh trực tiếp với dịch vụ của những mạng viễn thông như Vodafone hay China Telecom, nhiều dịch vụ VoIP kiểu mới đã chọn con đường khác "hòa bình" và khôn khéo hơn: tích hợp những tính năng đặc trưng của điện thoại vào các dịch vụ Internet hoặc blog, hoặc mạng xã hội ảo kiểu như MySpace và Facebook.

"Mạng xã hội ảo và blog đều phục vụ mục đích giao tiếp giữa người với người. Vậy mà điện thoại lại vắng bóng ở đó", ông Konstantin Guericke, Giám đốc điều hành dịch vụ Jangl cho biết.

Hiện tại, dịch vụ của GrandCentral mới đang ở giai đoạn thử nghiệm, tất nhiên là miễn phí. Nhưng nếu muốn dùng thử, bạn phải đăng ký thành viên tại địa chỉ www.grandcentral.com và nhận được lời mời (invitation) từ một thành viên khác, trước khi có số điện thoại riêng của mình.

"Tính năng gọi điện thoại sẽ sớm được tích hợp vào nhiều Website lớn trong vòng 3 tháng đến 1 năm tới", Guericke dự đoán. "Tôi nghĩ là ngành công nghiệp này có thể đẻ ra doanh thu tới 1 tỷ USD trong tương lai".

Google mua phần mềm hội nghị truyền hình

2010-01-06T00:01:00.001+01:00

Google đã mua phần mềm hội nghị Video từ Marratech AB, hãng phần mềm có trụ sở tại Stockholm, Thuỵ Điển và thuê lại các kỹ sư có liên quan đến phần mềm này.

Lúc này Google đã lên kế hoạch sử dụng phần mềm trên như một công cụ dành cho các nhân viên. Họ từ chối đề cập đến việc liệu Google có cố gắng tiếp thị công nghệ hay tích hợp nó vào một trong các sản phẩm thương mại của họ hay không.

Google nên quyết định tiếp thị hay tích hợp công nghệ vào các sản phẩm của họ. Động thái này có thể được thấy như một bước cho thấy Google chuẩn bị bước vào lĩnh vực công cụ cộng tác. Marratech sẽ tiếp tục hoạt động độc lập, hỗ trợ các khách hàng của họ và những người bán lẻ mua phần mềm. Tuy nhiên, phần mềm hiện thuộc Google do đó Marratech sẽ không thể tiếp thị nó nữa.

Các kỹ sư của Marratech hiện trở thành các nhân viên của Google sẽ vẫn ở văn phòng Thuỵ Điển và tiếp tục phát triển phần mềm hội nghị này. Các điều khoản của vụ mua bán phần mềm không được tiết lộ.

Với phần mềm hội nghị kiểu Cross-Platform, Marratech cho phép mọi người thực hiện hội nghị thấy hình, chia sẻ ứng dụng và nhắn tin nhanh. Phần mềm Client chạy trên Windows 2000 hay XP, Mac OS X 10.4 hoặc các phiên bản của Linux bao gồm gồm Mandriva 10.2, Suse 9.1 hay Fedora Core 2 và đòi hỏi máy tính có bộ xử lý Pentium III hay G4 chạy ở tốc độ trên 1 GHz. Nó còn bao gồm một thành phần chạy trên máy chủ là Marratech Manager.

Marratech cho biết nó rẻ hơn đối với các công ty mua hệ thống dựa trên máy chủ của họ và sử dụng các dịch vụ hội nghị dựa trên Web tính tiền cước phí theo phút. Các công ty về dịch vụ như vậy là WebEx Communications đã được Cisco Systems mua vào tháng trước với giá 3,2 tỉ đôla Mỹ.

Cisco ra mắt hệ thống truyền thông hợp nhất

2010-01-06T00:00:00.003+01:00

Ngày 13/7, nhằm đánh dấu 7 năm hiện diện trên thị trường truyền thông theo giao thức Internet (IP), Cisco đã tổ chức Hội thảo giải pháp truyền thông (VoiceComm 2006).

Hội thảo VoiceComm 2006 kéo dài trong vòng một ngày và là hội thảo dành cho các chuyên gia CNTT nhằm khám phá những công nghệ mới nhất trong công nghệ truyền thông IP và những giá trị mà Giải pháp truyền thông hợp nhất của Cisco mang lại cho các tổ chức kinh doanh.

Hội thảo này được bảo trợ bởi Bộ Bưu chính Viễn thông Việt Nam cùng với sự tham gia của nhiều đối tác trong và ngoài nước.

Nhân hội thảo VoiceComm 2006, Cisco cũng đã chính thức đưa ra giới thiệu Hệ thống truyền thông hợp nhất của Cisco - hệ thống vừa được công bố trên toàn thế giới vào tháng 3 vừa qua. Đây là hệ thông kết hợp các sản phẩm và ứng dụng truyền thông thoại và video nhằm giúp các tổ chức kinh doanh hoạt động liên kết hiệu quả hơn, giúp các tổ chức kinh doanh hợp lý hoá các chu trình kinh doanh, tiếp cận đúng nguồn tài nguyên ngay lần đầu tiên, và nâng cao hiệu suất và hiệu quả kinh doanh của doanh nghiệp.

Gồm nhiều phiên thảo luận chuyên đề đi kèm với các triển lãm của các đối tác về những giải pháp truyền thông IP, Hội thảo Cisco VoiceComm 2006 là một bước tiếp cận nền tảng "tất cả trong một" cho người sử dụng để tự cập nhật về những thông tin mới nhất trong chương trình đào tạo, sự phát triển và những đột phá trong công nghệ, giải pháp và sản phẩm truyền thông IP.

“Một hệ thống mạng hội tụ kết hợp các dịch vụ dữ liệu, thoại và video sẽ giúp các doanh nghiệp nhận thức rõ về tính hiệu quả của chi phí đầu tư và điều hành doanh nghiệp. Hệ thống mạng này cũng cho phép các tổ chức kinh doanh xây dựng một không gian làm việc ảo trên nền IP nơi mọi người có thể tương tác với nhau với hiệu quả thời gian thực và các thông tin luôn sẵn sàng phù hợp với từng người sử dụng vào đúng lúc họ cần” ông James Chia, Tổng Giám đốc Cisco Systems Việt Nam phát biểu.

Hệ thống truyền thông hợp nhất của Cisco

Hệ thống truyền thông hợp nhất của Cisco bao gồm 5 nhóm thành phần chính: Thoại IP (Phần mềm xử lý cuộc gọi, các máy điện thoại và các điểm giao dịch cuối); các ứng dụng truyền thông hợp nhất của Cisco (ứng dụng truyền thông hợp nhất dành cho máy khách, thông điệp, hội nghị với nội dung đa phương tiện phong phú); các ứng dụng trung tâm giao tiếp khách hàng hợp nhất của Cisco; cơ sở hạ tầng truyền thông của Cisco; bộ sản phẩm quản lý truyền thông hợp nhất của Cisco.

Theo Cisco, những tổ chức doanh nghiệp ngày nay đang phải cạnh tranh khốc liệt trong những môi trường truyền thông ngày càng phức tạp với hàng loạt những phương thức liên lạc khác nhau. Người lao động, đối tác kinh doanh và khách hàng liên lạc với nhau thông qua vô số những phương tiện như điện thoại, tin nhắn thoại, thư điện tử, fax, điện thoại di động và các công cụ phục vụ hội nghị có nội dung đa phương tiện phong phú. Tuy nhiên, những công cụ liên lạc này rất nhiều khi không phát huy được hết hiệu quả như khả năng vốn có của chúng. Kết quả là thông tin bị quá tải và việc liên lạc bị chệch hướng sẽ trì hoãn việc ra quyết định, làm chậm các quá trình kinh doanh và làm giảm hiệu suất hoạt động.

Những giải pháp truyền thông sử dụng công nghệ IP đã chứng minh khả năng của chúng trong việc giúp đỡ các tổ chức – doanh nghiệp giải quyết những vấn đề nói trên, cho phép họ tối ưu hóa các chu trình kinh doanh và giảm chi phí vận hành. Nhiều năm qua, các tổ chức – doanh nghiệp thuộc mọi quy mô đã và đang nhận ra những lợi ích thiết thực mà công nghệ truyền thông thoại, dữ liệu, video vận hành trên một cơ sở hạ tầng IP chung nhất có thể mang lại.

Trong môi trường toàn cầu hiện nay, những nhân viên có trình độ cao thường xuyên phải đi công tác ra ngoài văn phòng hay ra khỏi nơi làm việc nhóm của mình, đến những vị trí xa đồng nghiệp, xa cấp dưới và xa các giám sát viên của mình. Các tổ chức kinh doanh từ quy mô lớn đến các quy mô vừa và nhỏ (SMB) đã và đang lãng phí một khoản ngân sách không nhỏ để trang bị cho nhân viên có trình độ cao của mình các phương tiện truyền thông cũng như các ứng dụng được thiết kế cho mục đích duy trì liên lạc với các nhân viên này. Nhưng hiệu suất hoạt động vẫn thấp cộng với việc các dự án vẫn có thể bị chậm trể khi khách hàng không thể liên lạc kịp thời được với những người nắm vai trò quan trọng trong dự án.

Truyền thông hợp nhất là một sự kết hợp của các ứng dụng và dịch vụ đang phát triển rất nhanh hiện nay được thiết kế nhằm tăng cường khả năng liên lạc trong các tổ chức hiện đại - nhằm duy trì liên lạc giữa các nhóm làm việc, cho phép họ làm việc cộng tác với nhau một cách hiệu quả đồng thời tối ưu hóa các chu trình kinh doanh.

Hệ thống Truyền thông Hợp nhất của Cisco mang lại khả năng hoạt động thông minh và nền tảng mang tính cấu trúc giúp các tổ chức tích hợp những giải pháp liên lạc của mình một cách chặt chẽ hơn với các chu trình kinh doanh, và đảm bảo thông tin đến được với người nhận một cách nhanh chóng, thông qua phương thức giao tiếp hiệu quả nhất.

Các doanh nghiệp có thể phối hợp thời gian thực sử dụng những ứng dụng tiên tiến như là hội nghị truyền hình; hội nghị tích hợp thoại và web; điện thoại IP di động; hộp thư thoại; và nhiều hơn thế từ một giao diện tích hợp dễ sử dụng. Giải pháp này tiết kiệm thời gian và giúp kiểm soát chi phí vận hành đồng thời nâng cao hiệu suất kinh doanh và tính cạnh tranh của doanh nghiệp.

Cũng theo Cisco, hiện Cisco có hơn 130 khách hàng triển khai hơn 5.000 máy điện thoại IP, hơn 40 khách hàng triển khai hơn 10.000 máy điện thoại IP và hơn 15 khách hàng triển khai hơn 20.000 máy điện thoại IP. Tại Việt Nam Cisco có những khách hàng sử dụng Cisco Unified Communication như: Cisco Systems toàn cầu, Cisco Systems Việt Nam, Vietsovpetro, FPT Corporation, FPT Telecom, VDC Khu vực 2, Bảo Việt,...

Tại sự kiện Voice 2006 cũng đã thu hút được nhiều các diễn giả đến từ trong và ngoài nước: Bà Manjula Talreja, Giám đốc Cấp cao, Nhóm Công nghệ Thoại Cisco, Cisco Systems Toàn cầu, với chủ đề: Nâng cao hiệu quả kinh doanh với nền tảng truyền thông IP; Ông Nguyễn Xuân Việt, Giám đốc Trung tâm Công nghệ, FPT Information System, với chủ đề: Giới thiệu Hệ thống truyền thông hội tụ của Cisco – Cisco Systems,... Ông Manoj Menon, Giám đốc, Technology Practice, Frost & Sullivan châu Á – Thái Bình Dương với chủ đề: Hội tụ IP trong doanh nghiệp,...

Cisco chi 3,2 tỷ USD mua lại WebEx

2010-01-06T00:00:00.002+01:00

Ngày 15/3, Cisco Systems cho biết đã đồng ý mua lại hãng WebEx Communications - chuyên phát triển các ứng dụng cho phép thực hiện hội họp qua mạng với giá 3,2 tỷ USD tiền mặt. Điều này đã khiến Cisco trở thành đối thủ của Microsoft trong lĩnh vực truyền thông đa phương tiện trên nền Web, Microsoft đã đưa ra giải pháp Live, Office Communications Server 2007 cũng với những chức năng tương tự như giải pháp kết hợp giữa Cisco và ứng dụng web từ WebEx.

Chủ tịch kiêm CEO Cisco, John Chambers

Cisco, hãng sản xuất router và switch hàng đầu thế giới, cho biết thương vụ trên đã được hội đồng quản trị hãng này thông qua, và dự kiến sẽ hoàn tất vào quý IV năm ngân sách 2007.

Cisco hy vọng kết quả từ vụ sáp nhập trên sẽ "hiện hữu" trong doanh thu năm 2008. Giá mà Cisco trả cho mỗi cố phiếu của WebEx là 57USD, và hiện đã chuyển trước 300 triệu USD.

Trong những năm gần đây, Cisco liên tục thực hiện các vụ mua bán những công ty cỡ nhỏ chuyên về hệ thống mạng, giao tiếp, mạng xã hội và những mảng khác chuyên về các ứng dụng cải thiện lưu lượng mạng.

Về phần WebEx, hãng này chuyên sản xuất ứng dụng cho phép thực hiện các cuộc họp trực tuyến và tin nhắn tức thời bảo mật. WebEx cũng cho rằng hãng đang chiếm tới 64% thị phần hội họp trực tuyến với 3,5 triệu khách hàng trên thế giới đang sử dụng các dịch vụ của hãng này.

"Cisco đã đột ngột trở thành một gã khổng lồ tích hợp các giải pháp truyền thông đa phương tiện trên nền tảng Web", Mike Gotta nói, với một nghiên cứu từ Burton Group. "Cisco đã linh hoạt trong việc kết hợp các ứng dụng của mình với giải pháp từ WebEx và điều đó sẽ là mối lo ngại lớn với Microsoft"

Cisco có sự hiện diện của mình với một sức mạnh to lớn từ những lĩnh vực nhỏ và việc tạo ra những tiêu chuẩn chung cho quá trình truyền thông, và bây giờ đang lấp những khoảng chống vốn có của mình để nâng cao vị thế.

Trong khi hội thảo qua mạng trên nền Web được Cisco đã có ý định, trong khi sản phẩm chính thức đã được WebEx đưa ra với việc mua lại WebEx sẽ giúp Cisco lấy lại vị thế của mình trong lĩnh vực truyền thông và cũng là một mối lo cho Microsoft.

"Tất cả đều là sự bất ngờ khi Cisco chơi nước cờ mua lại WebEx", Gotta nói "Tôi không hiểu vì sao họ lại làm như vậy, nhưng với tình hình này họ sẽ trực tiếp đối mặt với Microsoft".

Khai sinh ra hội thảo qua Web, được đưa ra bởi WebEx Media Tone Networ (MTN), một giải pháp mạng toàn diện với nền tảng được thiết kế cho vấn đề bảo mật của ứng dụng. Microsoft đang cố gắng xây dựng một giải pháp tương tự với việc tổng hợp các dịch vụ của mình dưới một kênh là Live.

Thêm vào đó, WebEx có WebOffice, nó chạy trên MTN và cung cấp giải pháp chia sẻ, đặt lịch, dữ liệu cũng như hội thảo qua Web. WebOffice được thiết kế cho môi trường doanh nghiệp nhỏ, đội dự án của Microsoft cũng đưa ra giải pháp Office Live.

WebEx AOL/AIM Pro Busines Edition cung cấp tính năng VoIP, tập trung quản trị, bảo mật truyền file kết hợp với OutLook.

Microsoft cũng đưa ra một nhóm với tính năng tương tự với tên gọi Office Communications Server 2007, Office Communicator client và Outlook.

WebEx AOL/AIM Pro Bussiness Edition cũng kết hợp với các ứng dụng thương mại của WebEx và nó là những phần trong WebEx Connect. Sự kết hợp tạo ra những dạng khác nhau và những ứng dụng kết hợp trong Microsoft cũng làm việc giữa Dynamics ERP và CRM, nó là một dịch vụ online và là một nền tảng truyền thông thống nhất.

WebEx Connect sẽ kết hợp với dữ liệu của người dùng từ nhiều ứng dụng và cho phép họ tuỳ biến để phù hợp với môi trường làm việc của họ đáp ứng nhu cầu làm việc độc lập cũng như tương tác trong thương mại. Một thiết lập của WebEx Media Tone APIs được phát triển kết hợp với máy tính cá nhân và ứng dụng doanh nghiệp để tạo ra những ứng dụng cao cấp.

WebEx có đối tác như SugarCRM, là một tổ chức mã nguồn mở chuyên đưa ra các ứng dụng cho quản lý, và họ đã có những ứng dụngcho WebEx Connect.

Một sự tương phản, Microsot không kết hợp với bất kỳ một hãng nguồn mở nào cho đến cuối năm ngoái. CEO Steve Ballmer giới thiệu dịch vụ Microsoft's Dynamics Live CRM vào tháng 3 vừa qua, đó là một ứng dụng doanh nghiệp cho phép hội thảo trực tuyến, "Nó à một ví dụ điẻn hình bằng việc tích hợp giải pháp phần mềm và các dịch vụ tạo ra sự thuận lợi cho quá trình làm việc của khách hàng và các đối tác.

Tuần trước Microsoft đã đưa ra nền tảng giao tiếp với tên gọi VoiceCon đáp ứng hội thảo trực tuyến và vào tháng 3 này sẽ đưa ra sản phẩm Tellme để thêm vào nền tảng giao tiếp này.

Sự tích hợp giữa Cisco và một nền tảng truyền thông đa phương tiện như - Call Manager, Unified Presênc Server, MeetingPlace và MeetingPlace Express - Với các sản phẩm của mình và sự kết hợp giữa các công cụ truyền thông đa phương tiện giúp Cisco bành trướng sự ảnh hưởng của mình trong lĩnh vực mạng và các dịch vụ truyền thông.

Việc Cisco tham gia vào việc đưa ra các sản phẩm hỗ trợ hội thảo qua mạng trên nền Web sẽ tạo ra sự khó khăn lớn hơn cho Microsoft, khi mà họ đang muốn lấy lại các khách hàng của WebEx với LiveMeeting và giải pháp hội thảo qua nền Web, tiếp đến sẽ có sản phẩm phần mềm kết hợp như Outlook và các gói phần mềm của Microsoft khác tạo thành nền tảng truyền thông đa phương tiện của hãng.

Tuy nhiên hiện nay Cisco mới đang lên kế hoạch để chạy WebEx không thể đưa ra sơm hơn được "Họ đang tìm kiếm giải pháp kết hợp tối ưu giữa WebEx và giải pháp truyền thông đa phương tiện trên nền tảng mạng của Cisco.

Cisco hiện nay là đối tác chiến lược của IBM vào tuần trước khi mua lại WebEx đó là một lĩnh vực phức tạp và trong tương lai có thể Cisco sẽ làm việc với IBM để kết hợp giải pháp trong WebEx với Lotus Sametime, và giải pháp hội thảo trực tuyến.

FPT Telecom mobile network operators

2010-01-05T23:59:00.000+01:00

FPT Telecom mobile network operators

SIP network FPT Telecom is about to put into operation within three months, the network Wi-Fi Internet services will enhance productivity to exploit not only for laptop users (notebooks), but also for all mobile users (mobile phone)

SIP network with the Mobile Service Center of FPT Telecom deployed, the type of life that runs mobile phone operating systems such as advanced Symbian9, Windows Mobile 2003/2005 as Nokia mobile phones

(N80, N95, E61, E61i) or O2 XDA ... be nothing more phone "SIM 2. Because besides the ability to connect to the traditional GSM network, they connect to the SIP.

But why should more SIP networks while has radio CDMA, GSM? The answer is: Do not dismiss the possibility that the Internet can bring, especially in voice services. VN users should not have been very interested with the phone call service rates (which is essentially phone calls over Internet technology applications VoIP) as 177, 178. Or about students love technology still utilize Yahoo, Skype to make calls across continents. Now senior business has long been familiar with the system Assembly talks and conferences and television companies like Sony, Tandberg, Polycom ... VoIP application standards such as H.232 or SIP. Also, the unified communications system that IBM, CISCO, Microsoft, Avaya, Nortel, Siemens .... often referred to the technology through the IP network is the most powerful solution for enterprises to track , communication and control of human resources work as well as mobile handling multiple types of data and information.

A SIP phone as creating conditions for individuals to work and mobile entertainment can "hello" over the Internet with low cost without using landline phones. Foreseeing this need, FPT Telecom begins testing and is expected to exploit the system to three months in the SIP. Mr. Nguyen Van Phuc, deputy director of Research and Development Center of FPT Telecom, said: "This time, the number of users using Modern equipment is relatively more adequate to our decision to test and exploit SIP system. "Just where can connect wireless (Wi-Fi) to the Internet is a SIP call can current.

SIP, short for Session Initiation Protocol (Protocol Start-Version) is a signaling protocol IP phones used to set up, modify and end of VoIP phone calls. Standard to set up and signaling calls in the Internet, interactive real-time, can handle the complex network structure in which each version can be a two-way phone calls, a notification messages ... Besides Internet telephony application, SIP is now widely deployed in e-commerce. SIP support integrated voice with other web services to implement a multimedia communications like video conferencing, online exchanges ... With SIP, enterprises can replace equipped with a media center based on Internet protocols to answer the phone support of customers with low cost.

FPT has deployed Wi-Fi in 5000 at Ho Chi Minh City and Hanoi. Most of these nodes are exploiting them on a private network for wireless connectivity - Wi-Fi Mega of FPT Telecom. Mrs. Truong Nguyen Thu Ha - Deputy Director of Mobile Services Center of FPT Telecom, said: "All Wi-Fi is located in the house (indoors), including the system of restaurants, café, bank ( public areas for guests waiting transaction), the floor of the Stock Exchange for more than 10 bank ... large systems such as Trung Nguyen coffee, Ciao, Pho 24, Lotteria, KFC and India have Wi FPT-Fi. Main points of Wi-Fi will create conditions for the SIP phone users connect to the call via the Internet. However, Mr Phuc said that to date, this service runs on the phone most stable operating system Symbian9 have built software called SIP phone support.

Deploying Wi-Fi free, FPT will take anything to "make up more? "Users will SIP network access a forum dedicated to the handheld devices is the FPT www.pda.vn building," she Ha replied, "we do not have plans to deploy the system advertising, messages or services of value on this. To the FPT put into operation the SIP system, if we can negotiate better results with manufacturers of hardware, the user will be able to have a service SIP phone services free or very low cost is 50,000 VND per month for example.

SIP technology start dominate market

2010-01-05T23:49:00.000+01:00

SIP technology start dominate market

Starting from the countries with information technology development, more than half decades after the birth, SIP application technology has spread worldwide, contributing to the socialization of the telecommunications industry in the world, including Vietnam .

Session initialization protocol SIP (Session Initiation Protocol) is used to establish sessions in an IP network. Each version can be a two-way telephone call, a notification messages ... SIP protocols like H.323 was born earlier in that both have the ability to set and signaling calls in the Internet. However, other than H.323, SIP is a protocol peer interaction in real time, so can handle the complex network structure.

Since 1999 the international organization responsible for Internet technology professionals (Internet Engineering Task Force) SIP announced, hundreds of manufacturers have started business server and SIP phone features. SIP also played a key role in many multimedia applications such as instant messaging, video, game online ...

Particular, a few years, the community assessment SIP VoIP protocol is leading to transmit signals over the Internet. Technology became an important component of the integrated IP network data (text) and voice (voice). For example, companies can save money when only a single line to the desktop using IP (no line Monday for the traditional telephone). Meanwhile, the computer acts as a phone softphone, allowing users to make calls by selecting numbers in the phonebook on PC.

Besides Internet telephony application, SIP is now widely deployed in e-commerce. Protocol is based on HTTP, SIP supports easy integration of voice with other Web services to implement a multimedia communications like video conferencing, online exchanges ... Division answer customer calls can now be replaced by a media center based on Internet protocol more efficient, able to manage and expand multi-channel customer relationship.

Analysts will evaluate SIP is located steadfastness in wireless technology and mobile third generation (3G). Cisco and many manufacturers of IP equipment are put into SIP hardware, also Skype, Yahoo, Microsoft and America Online also integrates SIP instant messaging system for them. Telecommunications corporation known as WorldCom permanently opened a business technologies and SIP coordination with Microsoft to support the protocol in the test Windows XP Server.

In Vietnam, the standard SIP only been put into application in the Internet phone service from the last international in 2005. Pioneered the application of the Voice777 service telecommunications center new generation VN (VNGT). Company computing and data transmission are VDC and Saigon Postel technology investment and transmission lines for the birth of new services using SIP.

Providers in the country will have to invest more to new applications are widely accepted in the market, but can not deny that in 2006 SIP applications will also be asked to many in the industry and information technology Telecom Vietnam.

Internet phone service cheap international VNPT

2010-01-05T23:48:00.000+01:00

At first, iFone-VNN provided in the form of PC-to-PC domestic, international (free calls PC-to-PC) and PC-to-Phone (from the computer to type your phone different) abroad. Customers can sign up to use pay services provider services or prepaid iFone-VNN longer duration of the call and still balance in the account sufficient to make a call.

Morning 20/6 Group Vietnam Posts and Telecommunications (VNPT) has officially provides Internet phone service cheap international iFone-VNN as "iFone-VNN lyrics of the Internet".

iFone-VNN Internet phone service is being developed SIP-based technology (sesión Initation Protocol) using the Internet to set up global calls between regular phones, personal computers (PC) and the other terminals.

At first, iFone-VNN provided in the form of PC-to-PC domestic, international (free calls PC-to-PC) and PC-to-Phone (from the computer to type your phone different) abroad. Customers can sign up to use pay services provider services or prepaid iFone-VNN longer duration of the call and still balance in the account sufficient to make a call.

By using the card iFone-VNN, the international communications of customers will become very simple and convenient charges as super-cheap, good call quality, you can use a variety of other endpoints together to make calls and are able to monitor usage charge through the web ... Moreover, the iFone-VNN, customers can choose to use a separate phone number (with 9 digits 88xxxxxxx) to contact when traveling anywhere in the country or abroad. Phone iFone-VNN will be VNPT retains during customer service.

In addition to conversation, iFone-VNN also provides other features for users similar to a mobile or fixed wireless as a missed call notification (missed call), recall (return call), shows the number of calls, phone book (phone book), the call (call forward), chat, instant messaging (SMS) email ...

For more information about iFone-VNN service, customers can visit the website: http://ifone.vnn.vn.

start sip9

2009-12-12T22:16:00.001+01:00

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start sip9