Skype and its Future

In this world of technology, the one constant you can count on is that technology will constantly change, keeping us on our toes and delivering new ideas almost on a daily basis. While some of this technology is at times overwhelming, at others times it is exciting, easy to understand and extremely cost effective. Take for example Skype. Although an odd sounding name, Skype is anything but odd and is making quite a dent in the marketplace.

As an alternative form of telephone communication via the Internet, Skype is not only efficient and high quality, but it also allows users to communicate with people around the world. Not a big deal at first glance, right? But when you look more closely, it is a big deal, because Skype allows users to communicate with other Skype users around the world completely free. In fact, Founders, Zennstrom and Friis set out to create something that would be user-friendly and cost effective and they’ve done exactly what they set out to do.

What Exactly is Skype?

No large phone or controlling computer companies are involved with Skype as Skype is a VoIP (Voice Over Internet Protocol) transmitted via the Internet. The setup allows users to speak to other Skype users anywhere in the world, but rather than using traditional phone numbers, users communicate via usernames.

Because Skype is free, Skype data centers are causing quite a stir among other broadband phone service providers. The reason? All Skype takes is your computer or pocket PC and users can start communicating with people anywhere in the world and not pay a cent.

In addition to Skype’s standard features allowing users to talk free to other Skype users, it also allows conference calls with up to five users at a time all with the highest quality phone connection. Chats and file transfers are also included. Skype has optional services as well, which allow users to call traditional telephone numbers for a small fee in addition to receiving calls from traditional phones, but standard Skype communication is totally free.

How Skype Works!

The operation is tied to a computer and uses standard telephone sounds. The big difference is that standard phone frequency is 300Hz to 3.4KHz, while Skype leverages transmission in digital voice and reliable broadband networks, transmitting the full frequency range of human hearing (20Hz to 20KHz). This is the reason Skype audio works so well.

And as one of the easiest applications to implement, it is therefore rapidly growing in popularity. In fact, it’s grown so rapidly there has yet to be any marketing expenses associated with Skype and already more than 50 million people are registered users, with registered users growing by about 150,000 each day. Many of those millions that have already joined Skype discovered it simply by word of mouth.

To implement Skype one simply has to follow a few easy steps and voila, anyone can talk free to someone anywhere in the world.

Easily download the software.

Install the software which automatically configures itself.

Registers as a Skype user.

Get a headset for your computer.

Locate another Skype user.

Click on a Skype Contact Name.

Start talking.

Another important feature of Skype is that Skype calls are encrypted, which means that even if someone with bad intentions tried to tune in, they wouldn’t be able to hear the conversation because every Skype connection uses 256-bit encryption. In brief, when making a call to a regular phone, a call is encrypted until it reaches the “old” type of phone network. With Skype, the call is encrypted until it is routed on the Internet, so no one can intrude. Skype founders also claim their product is safe as it contains no spyware or any other nasty code that could potentially collect personal information about the user.

Are There Drawbacks?

While the overall consensus is that Skype is hugely successful, especially because of its ease of use and no charge, there are a few drawbacks and some concerns.

First, to be a Skype user, a computer must be turned on, and anyone calling must have Skype running on his or her system to receive a call.

Another issue is that some users report bandwidth and CPU usage increases considerably when using Skype.

Others still have some security worries. Although the calls are encrypted, there is concern that it’s bound to gain the attention of hackers. Although Skype offers its own protection, it’s probably best for each person that uses Skype to include a personal firewall, anti-virus protection and anti-spyware just to be on the safe side.

In Conclusion

While Skype will probably not ever be able to take over as the telecommunication of choice, Skype is definitely changing the way people are exchanging information. As a result, it looks like Skype is here to stay. And based on Skype’s overall past and present performance, it’s getting an excellent rating. As a useful, free and ingenious software for both individuals as well as companies, users will in all likelihood find it to be a very beneficial, cost effective and valuable tool that will probably continue growing and expanding around the world.

–About–
Robin Meyer is Web Manager of Your Wireless Source, a company that specializes in portable wireless products and accessories including headsets for all types of telecommunication needs.
Source: ArticleTrader.com

Introduction:

Performance testing has become an integral part of Client/Server and Web environments today. With performance engineering comprising performance testing tools, the testing cycle time can be reduced significantly. The capacity of existing performance testing tools has limitations in supporting various protocols. This white paper indicates that Performance testing tools offering wider coverage without much investment are the need of the hour.

The SituationA performance testing tool can simulate internet user behavior only on http protocol and can be enabled for multi protocols as well by plugging-in an ActiveX Component or a Java Class programming language code establishing a TCP/IP Connection within the script of the tool.

Common Protocol Support:

http

https

Telnet

FTP

While with licenses for the http protocol, the compact version of existing performance testing tools can only support the http/https; additional licenses are required to be purchased for supporting other protocol(s).
Enabling tool to support other protocols:

A particular performance testing tool can be made to support a group of desired protocols by setting up a server / application specific development process. This performance engineering is a complex process involving the following steps

Steps involved:

1. Deciding the transactions.

2. Setting up a packet sniffer for capturing the packets traveling between the user machine and the Server.

3. Executing the transactions manually.

4. Saving the details of the captured packets.

5. Analyzing the flow of packets starting from packet

6. Determining the user input in the packets.

7. Parameterizing the user inputs in the packets.

8. Creating functions to simulate the packet flow.

9. Creating an instance for that component inside the script of the tool for plugging-in the component within the tool.

10. Calling functions from the script.

Analyzing Packets:

Analyzing packets is the major part in the process requiring tedious coding and decoding.

Phases in Analyzing:

Six phases in the process of analyzing packets -

Separating Headers and Data portions in the packet.

Analyzing the Header packet.

Analyzing the Data packet.

Determining the constant and variable bytes in the data specific packet, both are usually in HEX String.

Repeating the process with various user inputs getting accustomed with the bytes transferred.
Checking if any data for the previous response is transmitted with the next request.

Ethereal - Network Protocol Analyzer is very helpful sniffing tool to see new variable bytes by simulating traffic from code detecting several protocols, such as:

H323 / Q931 VOIP Protocol.

ICQ Protocol.

Java RMI.

Remote Procedure Call.

IEEE 802.11 wireless LAN.

Ethereal has a rich display filter language and can view reconstructed stream of a TCP session.

Creating the PSEUDO Client:

While designing the client, the transaction is divided into stages with separate functions for every user interaction and server reply.

Separate functions are required for:

Sending the first request and receiving the first response.

Extracting data from the response to append in the second request.

Sending second request and receiving second response.

Similar process must be followed for the remaining traffic. Maximum number of variables must be created for keeping track, with the possible constant fields stored as variables. Following is the screen-shot of an ethereal record of a simple http transaction.

Integration:

Finally, the client component is integrated in the tool, which requires creating an instance of the component and calling the function with different parameters i.e., user inputs, which allow the desired protocol to be supported by the tool.

Process Pros:

1. Perfect simulation with perfect packet analysis.

2. Frequent update for new plug-ins for the new protocols.

3. Cost effective.

Process Cons:

1. Difficulty in decoding.

2. For encrypted applications the process cannot be implemented.

3. For Oracle NCA protocol the process can be implemented only if plug-in is readily available.

Conclusion:

Performance testing can demonstrate the system meeting performance criteria. The later a performance defect is detected, the higher the cost of remediation. This is true in the case of functional testing, but even more so with performance testing, due to the end-to-end nature of its scope. It is always helpful to have a statement of the maximum number of users likely to use the system at peak times. Performance testing almost invariably concludes that it is the software rather than hardware that contributes most to bottlenecks in processing data.

ReadyTestGo is a professional Software Testing Company ( http://www.readytestgo.com/readytestgo.htm ) and Outsourcing QA
( http://www.readytestgo.com/services.htm ) For more details, please contact marketing@readytestgo.com

–About–
ReadyTestGo is a professional href=http://www.readytestgo.com/readytestgo.htm> Software Testing Company and Outsourcing QA . For more details, please contact marketing@readytestgo.com
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Session Initiation Protocol (SIP) is a signaling protocol for establishing sessions in an IP network. But if you are like most of us, that means nothing to you. In layman’s terms, SIP is a method by which various computers can talk to one another so that they can complete voice calls. The protocol is increasingly being adopted as the standard means by which computers communicate to facilitate VoIP or Voice Over Internet Protocol. So you can imagine SIP as a common language for new generation operators to speak to connect calls. However, there are no operators there is only your computer (or other hardware) and that of the person you are speaking with. That brings this introduction to some of the many benefits of SIP communication.

The goal of SIP was to provide users with many of the functions and features they typically expect with making phone calls, such as familiar rings, hearing the ring back tone when a call is placed, and the process of dialing a number. SIP goes beyond this however, also implementing a number of advanced features. Despite its’ convenient interface that mirrors that of a typical telephone call, SIP is based on an internet protocol rather than that of the telephone industry. Because of this, SIP is able to work seamlessly alongside other internet based protocols. This has allowed the technology to uniquely establish a user location, meaning that you can tell the IP address or “location” from which a person is making a call, something vital for offering emergency services. It also allows for coordination amongst the various participants in deciding upon what call features will be supported, as well as providing the protocol for call management which allows for adding, dropping, or transferring call participants.

One of the most exceptional benefits of Session Initiation Protocol (SIP) is its application with Private Branch Exchange (PBX). A private branch exchange is a private telephone network used within an enterprise in which users share a certain number of outside lines for external telephone calls. This provides a significant cost savings to the company because it allows companies to quickly and easily make calls within their institution, as well as save by limiting the number of external phone lines that must be maintained. SIP can extend these cost savings dramatically by offering users free long distance calls worldwide. Once again, because SIP is internet based rather than running over traditional telephone lines, the cost of call transmission are as cheap as say sending an e-mail, that is to say, Free! While PBX is already an efficient use of office resources incorporating SIP into a PBX means taking such savings and capabilities to a new level. Incorporating SIP gives users’ access to free interoffice communications, long distance calls, as well as huge savings in setup and transaction costs. These transaction cost savings are due to the fact that SIP is based on internet protocol allowing for the ability to physically move phones without any need for rewiring or new setup costs. Because that the system is peer-to-peer rather than cog and wheel like hardwired telephony means that there is no complicated setup necessary, but rather users can simply plug the phone into any available broadband connection and without the need for any complicated hardware or software, calls are ready to be made and received.

This brings us to what is potentially the greatest savings that SIP provides. We all hate to have to pay $2.00 for making a simple local call from a hotel room. But for those of us that have had to pay exorbitant long-distance charges from hotels, the level of angst felt reaches new heights. Add to this the fact that often companies require conference calling for their employees, a service that hotels are all too happy to charge a high premium to provide. For companies that have a lot of their employees traveling or practicing in various locales, the overhead costs of making such calls can really put a damper on year end profits. SIP offers a solution to hotel bills, and for that matter all telephone bills whatsoever. Just as sending an e-mail is free whether you are sending that e-mail to a person across the street or around the world, so making a call using SIP from any locale to any other locale in the world is free. All that is needed is a broadband connection. And just as one can send an email from wherever the internet can be accessed, SIP users can make their free calls from the office, the home, or even (gasp) the hotel room. Not to mention the huge savings in regular long-distance charges, the simple fact that long distance and local calls can be made for free from hotel rooms (most of which offer complimentary broadband service) is an enormous long-term cost savings. But add to this the fact that conference calls are available at no additional cost, which can be explained by following the principle that sending an e-mail to many individuals at the same time is just as cheap as one to one communication. Altogether these savings mean that companies who have in the past been laden with high local and long distance phone bills will be able to have their employees keep in close communication no matter their location for free.

SIP goes beyond this, however. Because SIP uses peer-to-peer connections there are no extra costs for having hundreds or even thousands of employees making SIP calls whereas the management costs for such a system under traditional telephony would be astronomical. This is never the case with PBXs that utilize SIP in a peer to peer connection format, however, where structural costs do not increase as your business grows and your usage of the PBX increases.

Session Initiation Protocol (SIP) offers customers of traditional telephony fantastic cost savings, the flexibility to make free calls from anywhere in the world at any time, and the ability to expand a PBX system with no setup costs. Because there is no complicated hardware or software necessary to setup SIP in a PBX, implementing the technology is as simple as having access via broadband to the internet. Due to these benefits as well as innumerable others, expanded use of SIP in PBX and in a variety of other settings is simply a matter of expanded customer exposure to the possibilities of SIP.

–About–

Jim Sherman writes about interesting topics such as Mobalex technology. See http://www.mobalex.comfor more information.

Source: ArticleTrader.com

Today’s telecommunication arena is rapidly moving towards next-generation networks (NGNs) that offer ubiquitous, converged services over converged voice, video, data and mobile networks. Until very recently, the prevailing telecommunications paradigm was based on multiple types of networks, each serving different types of applications.

While IP-related technology holds the most promise for meeting the requirements of NGNs, the use of the global Internet is quite limiting. The IP Multimedia Subsystem (IMS) is an architecture that allows delivery of identical services to fixed and mobile customers – regardless of whether they are connected through the packet-switched (PS) or circuit-switched (CS) network. IMS-based services enable communication in a variety of modes – including voice, text, location, presence, messaging, pictures and video, or any combination of these.

In addition to service creation and delivery, IMS handles call control issues, and can easily be adapted to serve roaming subscribers. The IMS architecture is inherently capable of bridging between separate networks, and will ultimately be used for all types of networks, such as wireline Voice-over-IP (VoIP) networks, WiMax wireless networks and packet cable networks.

The expansion and acceptance of IMS is an indication of just how important this technology is to the future of unified communications. Many organizations which are responsible for network standardization are currently adopting IMS technology. The ramifications for the industry are significant, and the entire telecommunications industry is gearing up for the imminent widespread implementation of IMS.

Based on a Strong Signaling Foundation: Session Initiation Protocol (SIP)
IMS technology was originally developed for the cellular arena to define how to set up advanced services for 3G cellular networks and grew out of a group of standards created by the 3rd Generation Partnership Project (3GPP).

IMS is a Media-over-IP network and uses the Session Initiation Protocol (SIP), originally standardized by the IETF, as its base signaling protocol. The 3GPP chose SIP as its base protocol because previous telecom signaling protocols failed to comply with all IMS requirements. Because SIP is an Internet protocol, it can accommodate convergence, and has the potential to meet all the needs of the IMS architecture. For instance, SIP can signal between different network entities, including endpoints and servers. In IMS, each network server has its own role, in contrast to traditional networks where a central office switch does it all, including call control and service control. In addition, SIP uses Internet extensibility mechanisms. A service provider with IMS networks initially may only have a small number of subscribers. As the subscriber base grows, IMS networks must be easily scalable to add more subscribers. SIP is also very flexible, and uses standard extensions. SIP’s flexibility enables IMS networks to adapt and change signaling protocols to meet dynamic market needs. Finally, SIP provides adequate security, with both internal and external security mechanisms.

IMS SIP: A Complex Challenge
While offering the right foundation, SIP in its IMS form has proven to be quite complex and presented many technological challenges. There were many gaps between the SIP initially defined by the IETF, and the features required for full IMS support. To solve this problem the 3GPP defined dozens of SIP extensions – additions that are specific to IMS networks. Collectively, these extensions comprise the IMS SIP protocol, which is defined in the 3GPP TS.24.229 standard. These extensions, such as extended call control, presence and instant messaging, extend the functionality of SIP on IMS networks. This new IMS SIP usage profile is perhaps the most important in the telecommunications industry, and is uniquely the most appropriate for NGN networks.

To illustrate the inherent complexity of IMS SIP and all its extensions, we will review the major extensions below:

SigComp (RFC 3320)
The SigComp extension defines how to compress SIP textual signaling data, which can be very large and problematic to transmit, causing delay. SigComp solves the challenges of roundtrip delays, as well as mobile user equipment battery life
P-headers (RFCs 3455 and 3325)
(P- Private) In addition to standard headers, the 3GPP defined additional headers targeted at solving specific IMS network problems, such as obtaining information about the access network (cell ID) and the visited network (roamed network), and determining caller identity.
Security Agreement (RFC 3329)
This IMS SIP extension specifies how to negotiate security capabilities for multiple types of endpoints.
AKA-MD5 (RFC 3310)
This IMS SIP extension determines how terminals and networks are authenticated using already defined mechanism (e.g. ISIM), as well as specific key exchange.
IPSec
IPSec is used on various IMS interfaces and between different IMS networks. IMS uses IPSec in the transport mode, as opposed to the standard used in VPN services.
Media Authorization (RFC 3313)
Ensures that only authorized media resources are used.
Mobile Registration (RFCs 3327 and 3608)
On IMS networks, the terminal registration process is more complicated, as it includes various security extensions and must deal with registration from a visited network. RFC 3608 and RFC 3327 define the syntax and SIP entity usage of the Service-route and Path headers.
Reg-event Package (RFC 3680)
Used by the terminal and the P-CSCF to know the terminal registration status on the network.
IPv6
IMS prefers IPv6 networks, which offers distinct advantages. It permits a larger set of addresses and contains embedded IPSec functionality that may eliminate the need for entities like NATs and firewalls.
Preconditions (RFC 4032)
Specifies method for negotiating QoS, security and other required call behavior between two terminals.
IMS Resource Reservation (RFC 3312)
Defines how to make resource reservations for phone calls or sessions.
Session Description Protocol (SDP)
SDP defines the basic negotiation process for the media streams, and includes the bit rate and codec to be used, as well as other media attributes. IMS extends SDP with even more extensions, such as grouping of media lines, QoS and preconditions attributes, supplemental codec support, and bandwidth modifiers.
XML Usage
IMS SIP signaling uses XML protocols extensively, including XCAP, to implement various kinds of SIP message contents, and to allow full function interfaces between IMS entities.
IMS Simple Extensions
The SIMPLE group is an IETF working group that defines presence and instant messaging signaling requirements. Basic SIMPLE definitions were inadequate for IMS applications because they were not efficient enough for use on the air link. IMS SIP extended this standard with the following:
. Partial Notifications / Publications
. Notifications filtering
. Resource list / SIP exploders
. Message Session Relay Protocol (MSRP)

IMS SIP Expertise: A Prerequisite for Success
The use of SIP in IMS networks requires a great deal of adaptation and extension of the original signaling protocol. Given the breadth, variety and complexity of IMS SIP, it is indeed an arduous task to develop new services and applications from scratch. A more reasonable approach is to use prepared toolkits and infrastructure products that encompass all the nuances of IMS SIP, and where much of the development effort and interoperability testing (IOT) have already been completed.

In order to execute IMS roadmaps and ensure on-time deployment, developers need solutions that are finely tuned to the unique characteristics of IMS SIP and that provide the extended SIP signaling infrastructure needed for IMS applications.

–About–

A longer 25 page version of this white paper is available at http://www.radvision.com/Resources/WhitePapers/ims_sip.htm
Adi Paz is a Senior Product Marketing Director at RADVISION, a leading provider of video network infrastructure and developer tools for unified visual communications over IP, 3G, and emerging next-generation networks.
Source: ArticleTrader.com

Security is an obvious concern when it comes to any sort of technology, but even more so with any technology that is run through the Internet. Because VoIP runs through the Internet any information can be intercepted by anyone at any time. Because many things go through phone line, private information can wind up in the hands of the wrong person. Obviously, nothing is a one hundred percent guarantee because as fast as technology is made to keep information from getting in the wrong hands, the wrong hands are working to figure out how to break through those systems. Luckily, VoIP security is becoming more and more well rounded all the time and soon it’ll be so well done that even the best of the best won’t be back to get their hands on personal information.

One of the ways that most VoIP providers secure their customers personal information is through the tunneling and encryption process. These techniques keep hackers and those will ill intent from capturing information packets as they pass through the internet. Most VoIP providers use Layer 2 tunneling and an encryption method called Secure Sockets Layer or SSL to keep anyone from getting into the information they shouldn’t have. The security of VoIP will undoubtedly change and become more sophisticated as technology allows and consumers demand more security and more privacy. For some time to come VoIP security will remain a huge concern, just because it’s widely known that all information that passes over the internet could potentially fall into the hands of someone with ill intent.

Don’t let VoIP security issues keep you from getting VoIP services. The benefits of VoIP far outweigh the security risks. The bottom line is that you are more at risk every time you get online sending emails and paying bills than you will be every time you use your VoIP services. So, the features and convenience are well worth the small security risk associated with the internet access associated with it!

–About–
Laura Rupert is a freelance writer who contributes to The Tech FAQ
Source: ArticleTrader.com

Paid VoIP plans start at around $20 a month which comes to about $250 a year. So why pay for it if you can get it for free? But before I delve into that let me notice that even with the free VoIP to landline or cell phone offers – there are often small long distance charges with common exceptions often dealing with free promotional offers.

There are certain features that the paid VoIP provider might provide which a free one might not. One such feature is the 911 emergency call service functionality. This particular issue has to do with the difficulty of pinpointing a geographical location of a call that is being routed through the Internet. FCC has regulated in 2005 that all VoIP providers hooking up to PSTN network are required to comply with providing full 911 call compatibility or otherwise not to obtain new customers in areas where it is mandatory.

Sometimes there is also raised a question of a voice transmission quality difference between the two types of VoIP although in my personal experience that has not been an issue.

Next there is the local phone number availability. Some free VoIP providers offer it as an add-on paid feature. The same may apply to extras like caller ID, call waiting, call forwarding and voice mail. Others allow you even to have more then one local number and in remote geographical locations.

If you are a business VoIP user the ability to send faxes over VoIP might be important to you as well. Or if you are using VoIP for conference calling with large number of simultaneous participants then it is definitely something to find out – how far will your VoIP provider be able to accommodate you with that.

The final answer to which type of VoIP is better will depend on whether you need these extra features. Also many paid VoIP providers have sign up specials offering either free VoIP phone sets and/or interface boxes for regular corded or cordless phones . So if you are planning on using regular phone sets for dialing up to VoIP network that might be then also something to consider.

And if you are not depending on free long distance VoIP to PSTN offers from the free VoIP providers then the amount of time you spend on the phone can also be a factor in deciding which type of VoIP is best suite to your needs.

–About–

Ramzes Berezecki is recognized for his informative writings relating to
VoIP phone and
VoIP service provider. For more
information regarding VoIP please visit his blog at

http://voipchoice.blogspot.com

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It’s actually quite easy to compare VoIP (http://www.tech-faq.com/voip.shtml) providers because the technology is so new that the features that each provider offers differ quite a bit. Just like any service, you want to be sure that you get the best deal, and a program that will fit your needs and desires. Luckily, comparing VoIP providers will be quite simple and you’ll quickly be able to select the provider that is best suited to you.

The first thing you’ll want to do when you want to compare VoIP providers is find out which ones offer services in your area. You may be able to eliminate a couple providers just by looking at their areas of service. Once you eliminate the providers that just won’t work out, you’ll have to look at features and price.

One of the biggest differences between VoIP providers is the features that they can offer. Many have very easy to use routers that allow you to use a regular analog signal and digital signal when it’s suitable. Some VoIP providers only allow calls to other VoIP users, and others allow you to call anyone, anywhere, anytime.

If you make a lot of long distance calls you’ll really want to pay attention to the plans and the provider that you go with. Some VoIP providers don’t distinguish between local and long distance calls, so you basically get all your calls one month for just a service fee. Other VoIP service providers don’t allow you to make long distance calls, or they charge exorbitant fees for them. So, the amount of long distance phone calls you make may be the deciding factor in your choice for a VoIP provider.

Do a lot of research, just like you would when you are looking into any service. If you know whom you are dealing with it’s a lot easier to make an educated decision. When you determine exactly what you need then it’s just a matter of matching your needs with what is offered! Good luck, VoIP is definitely the wave of the future and it will only get easier and more advanced!

–About–
Laura Rupert is a freelance writer who contributes to The Tech FAQ
Source: ArticleTrader.com