voip-nayeem

International VoIP implementation

2009-12-04T04:16:00.000-08:00

In Japan, IP telephony is regarded as a service applied by VoIP technology to the whole or a part of the telephone line. As of 2003, IP telephony services have been assigned telephone numbers. IP telephony services also often include videophone/video conferencing services. According to the Telecommunication Business Law, the service category for IP telephony also implies the service provided via Internet, which is not assigned any telephone number.

IP telephony is naturally regulated by Ministry of Internal Affairs and Communications (MIC) as a telecommunication service. The operators have to disclose necessary information on its quality, etc., prior to making contracts with customers, and have an obligation to respond to their complaints cordially.

Many Japanese Internet service providers (ISP) are including IP telephony services. An ISP who also provides IP telephony service is known as a "ITSP (Internet Telephony Service Provider)". Recently, the competition among ITSPs has been activated, by option or set sales, in connection with ADSL or FTTH services.

The tariff system normally applied to Japanese IP telephony is described below;

* A call between IP telephony subscribers, limited to the same group, is usually free of charge.
* A call from IP telephony subscribers to a fixed line or PHS is usually a uniformly fixed rate all over the country.

Between ITSPs, the interconnection is mostly maintained at VoIP level.

* Where the IP telephony is assigned normal telephone number (0AB-J), the condition for its interconnection is considered same as normal telephony.
* Where the IP telephony is assigned specific telephone number (050), the condition for its interconnection is described below;
o Interconnection is sometimes charged. (Sometimes, it is free of charge.) In case of free-of-charge, mostly, communication traffic is exchanged via a P2P connection with the same VoIP standard. Otherwise, certain conversions are needed at the point of the VoIP gateway which incurs operating costs.

Since September 2002, the MIC has assigned IP telephony telephone numbers on the condition that the service falls into certain required categories of quality.

High-quality IP telephony is assigned a telephone number, normally starting with the digits 050. When VoIP quality is so high that a customer has difficulty telling the difference between it and a normal telephone, and when the provider relates its number with a location and provides the connection with emergency call capabilities, the provider is allowed to assign a normal telephone number, which is a so-called "0AB-J" number.

Comparison of VoIP software

2009-12-02T05:51:00.000-08:00

VoIP software is used to conduct telephone-like voice conversations across Internet Protocol (IP) based networks. VoIP stands for "Voice over IP". For residential markets, VoIP phone service is often cheaper than traditional public switched telephone network (PSTN) service and can remove geographic restrictions to telephone numbers (i.e. have a "New York" PSTN phone number in Tokyo).

For enterprise or business markets, VoIP enables the enterprise to manage a single network (the IP network) instead of separate voice and data networks, while enabling advanced and flexible capabilities to the end user.

Softphones are end-user–based clients for initiating and receiving voice and video communications over the IP network with the standard functionality of most "original" telephones and usually allow integration with IP phones and USB phones instead of utilizing a computer's microphone and speakers (or headset). Most softphone clients run on the open Session Initiation Protocol (SIP) supporting various codecs. Skype runs on a closed proprietary network. Online "Chat" programs now also incorporate voice and video communications.

Other VoIP software applications include conferencing servers, intercom systems, virtual FXOs and adapted telephony software which concurrently support VoIP and PSTN like IVR systems, dial in dictation, on hold and call recording servers.

Compatibility with traditional analog telephone sets

2009-12-01T22:51:00.000-08:00

Compatibility with traditional analog telephone sets

Some analog telephone adapters do not decode pulse dialing from older phones. The VoIP user may use a pulse-to-tone converter, if needed.[citation needed]
[edit]
Fax handling

Support for sending faxes over VoIP implementations is still limited. The existing voice codecs are not designed for fax transmission; they are designed to digitize an analog representation of a human voice efficiently. However, the inefficiency of digitizing an analog representation (modem signal) of a digital representation (a document image) of analog data (an original document) more than negates any bandwidth advantage of VoIP. In other words, the fax "sounds" simply don't fit in the VoIP channel. An alternative IP-based solution for delivering fax-over-IP called T.38 is available.

The T.38 protocol is designed to work like a traditional fax machine and can work using several configurations. The fax machine could be a traditional fax machine connected to the PSTN, or an ATA box (or similar). It could be a fax machine with an RJ-45 connector plugged straight into an IP network, or it could be a computer pretending to be a fax machine. [53] Originally, T.38 was designed to use UDP and TCP transmission methods across an IP network. The main difference between using UDP and TCP methods for a FAX is the real time streaming attributes. TCP is better suited for use between two IP devices. However, older fax machines, connected to an analog system, benefit from UDP near real-time characteristics[citation needed].

There have been updated versions of T.30 to resolve the fax over IP issues, which is the core fax protocol. Some new fax machines have T.38 built-in capabilities which allow the user to plug right into the network with minimal configuration changes[citation needed]. A unique feature of T.38 is that each packet contains a copy of the main data in the previous packet. This is an option and most implementations seem to support it. This forward error correction scheme makes T.38 far more tolerant of dropped packets than VoIP[citation needed]. With T.38, two successive lost packets are needed to actually lose any data. The data you lose will only be a small piece, but with the right settings and error correction mode, there is a high probability that you will receive the whole transmission.

Tweaking the settings on the T.30 and T.38 protocols could also turn your unreliable fax into a robust machine[citation needed]. Some fax machines pause at the end of a line to allow the paper feed to catch up. This is good news for packets that were lost or delayed because it gives them a chance to catch up. However, were this to happen on every line, your fax transmittal would take a long time. Another possible solution is to treat the fax system as a message switching system, which does not need a real-time data transmission (such as sending a fax as an email attachment (see Fax) or remote printout (see Internet Printing Protocol)). The end system can completely buffer the incoming fax data before displaying or printing the fax image.

Planning for the Future: Testing for Scalability

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. Planning for the Future: Testing for Scalability
Tolly Researchs testing and evaluation of VoIP and converged networks often focuses around
small-scale implementations, which affords us the opportunity to isolate the technologys impact in
the absence of other variables and establish credible benchmarks.
Our engineers have investigated the impact of VoIP components on network performance, weve
benchmarked voice quality, evaluated the impact of latency and packet loss on voice quality and
the impact of firewalls on VoIP performance.
While our initial benchmark research paints a picture of the performance capabilities of VoIP
under limited-load conditions, Tolly Research and user organizations need to expand that picture
by conducting the same type of tests under actual load conditions, and scale up network loads to
stress test the technology to determine if it can deliver suitable performance at projected loads in
their specific network environments. Using traffic optimization techniques and QoS, we can reevaluate
performance under stress and compare to various benchmarks.
In essence, what were talking about here is scalability. While testing a particular technology from a
functional standpoint may yield acceptable latency and voice quality results, organizations need to
assess how that performance varies under heavy and peak loads.
As you do so, youll begin to learn about the critical performance points in your network. While a
given technology may perform well under certain server loads, that same technology may suffer
as loads scale up and you learn that the networks queue management or a given devices I/O
processing cannot handle the increased demand.
The need for this type of testing is much more significant in the presence of real-time VoIP traffic
running alongside data-oriented IP traffic. The real-time orientation of VoIP places great strain on
available bandwidth, and adding to the equation, voice traffic creates unpredictable loads making
scalability testing much more important to ensure network capacity is architected well above
anticipated thresholds.

Call Handling and Testing
When we consider scalability testing, we have to consider the scalability of the network to
handle VoIP calls, in addition to data traffic. In order to determine that level, proper network
testing is required to determine the maximum call capacity sustainable by the network under
normal and peak data loads. Adequate call handling testing should be conducted with QoS
enabled, and disabled, with network devices such as firewalls and VPNs enabled and
disabled, because organizations need to understand the various set points regarding call
handling under various operating conditions and the extent to which voice quality is
impacted.
Moreover, testing should include the evaluation of converged network services, such as
integrated fax, voicemail/E-mail, and unified directorie

How can I get free VoIP?

2009-11-20T22:17:00.000-08:00

The first thing to know about free VoIP calls is that none of them are actually completely free. Even in the best hypothetical case in which the VoIP provider doesn't charge you at all, remember that you still have to pay for your broadband Internet connection. You must understand that the goal is not to achieve completely free calls to all destinations, but to use the VoIP operator that suits your needs best. Keeping that in mind, you will learn that most VoIP companies will let you talk for free in their own network but also they will charge you for making calls outside their proprietary network.

The main way for free VoIP calls companies is to offer free calls inside their own network and also towards other specially selected destinations. Using this tactic users are drawn to make calls to free destinations and afterwards purchase credits to make calls towards paid destinations.

There are several ways used by VoIP companies to lure customers and make a profit in the same time. The great thing about VoIP calls is that they're very cheap, but not completely free, here are some systems used today in the VoIP calls market:

If you take Skype for example, one of the most popular VoIP services on the market, you will see that you can initiate conversations with other PC users of Skype free of charge. Of course, this is an advantage for long distance calls, as there is no fee for calls inside the Skype network. But if you want to make calls to regular landlines, you'll have to pay. The subscription fee for calls in North America is $30 per year. It's not a great deal of money but it still isn't free. You can make free phone calls on a PC to PC basis using the Skype software, and the number of users on the Skype network is continuously growing. On the other hand if you want to reach someone that doesn't have a PC or an Internet connection, you'll have to pay the required fees.

Another approach to this, could be the way Raketu is seeing things. Raketu is offering free phone calls to landlines in 42 countries and besides that, it also offers live video television. The downside to Raketu's service is that they ask you to pay $9.95 up front in order to use their free services. They say it's used as credit if you happen to call destinations that are not on the free call list, but either way you look at it; it's money that you have to pay to use the service.

Building an Open Voice-Over-IP Solution

2009-11-20T22:15:00.000-08:00

Building an Open Voice-Over-IP Solution
Some VoIP solutions that exist today try to solve the interoperability problem using Session Initiation Protocol (SIP). SIP, a protocol created to standardize the setup of IP calls, multimedia conferencing, and other Internet communications was designed to standardize VoIP technology and create vendor interoperability. While it has been successful in creating industry standards, in practice the market has been stymied by the way the protocol is being used. The standard has been “polluted” as vendors add their own proprietary features to the protocol. The resulting vendor differentiation and customer lock-in is useful to vendors; interoperability problems stem from vendors taking different approaches to SIP features, delaying the wide adoption of commercial VoIP solutions. (Marsan, 2004).
Establishing a truly standards-based dominant solution in the market requires a broadly accepted technology and a business model that leads to pervasive adoption. In the telephony market, this requires combining two parallel worlds -- voice-over-IP and open source. VoIP solutions were created to benefit Enterprises, allowing businesses to take advantage of the cost savings and scalability of voice integration over data networks, reducing long distance toll charges and eliminating the need to run separate networks for voice and data. At the same time, open source is at the heart of the current business revolution. Major enterprises are running mission-critical functions on open source software and big vendors such as IBM and Oracle have lined up to support it. Linux, Apache, Tomcat, and Java are just a few of the open source software solutions corporations are using today. A Forrester Research study conducted in 2003 found that 72 percent of large corporations surveyed said that Linux usage would increase over the next two years. IBM derived over $1 billion in revenue from Linux in 2002, and Oracle has roughly 6,000 employees in its global Linux support team. Chief information officers (CIOs) who have implemented open source solutions reported large total cost of ownership reductions. (Koch, 2003)

Open Source IP Communications

2009-11-20T22:13:00.000-08:00

Open Source IP Communications
In a small or medium-sized business, the cost of telephone equipment, phone lines, and long distance calls can easily reach tens to hundreds of thousands of dollars. A chief executive officer or operations manager, looking for ways to lower costs, finds that there is little choice when searching for solutions. Equipment prices are high; management fees are steep; proprietary software locks a company into a particular vendor’s solution, and that’s not all. Phone equipment vendors, which virtually become monopolies after insinuating themselves into a business, expect companies to pay for costly upgrades on their time schedule, not their customer’s.
Although attempts to solve business problems with innovative voice-over-IP (VoIP) solutions exist, the final product is simply a replication of the vendor lock-in model. The freedom to choose between vendors at any stage, which has become the increasing norm since the telecommunications industry deregulated, has yet to enter the IP telephony market, until now.
Pingtel’s new Enterprise Communications Model, based on Enterprise-grade open source software, empowers small-to-medium-sized businesses (SMB) with the ability to decide when and how to use voice-over-IP (VoIP) technologies. Businesses can choose to use the Enterprise-grade VoIP solution at any stage in a deployment, whether it’s in conjunction with an existing traditional-phone system, an upgrade to another vendor’s VoIP system, or a new build; technologies and vendor solutions can be mixed and matched. It’s a radically different solution. It invites competition. The simplicity of an open source model creates obvious cost benefits and value for businesses.
SIPXchange, the industry’s first open source Enterprise communications suite, combines the cost savings of voice-over-IP technology with the quality and adaptability of open source software created by a community of developers; it’s a solution that can improve a business exponentially. Using an open source solution, companies are no longer forced to pay licensing fees for software or premiums for proprietary hardware and software. While companies will pay less for licensing, they will continue to receive the Enterprise-grade support they get from traditional vendors, including software additions, service level agreements, and additional services such as training. If there is a failure, companies will have the ability to call and resolve the issue. Guaranteed service contracts will make open source the low risk solution for wide adoption in an Enterprise environment.
Using a voice-over-IP solution, a company can take advantage of the merged benefits of IP telephony. Rather than paying to run

QoS Policy Effectiveness on the WAN

2009-11-20T22:11:00.001-08:00

QoS Policy Effectiveness on the WAN
The same traffic queue management issues relevant on the LAN apply to the WAN, though
with far greater complexity. The issue over WAN links, unless they are privately managed, is
how much control your organization maintains once traffic leaves your private network for
public connections. Moreover, once prioritized traffic leaves your network, what happens to
those set priority levels as they are remapped across the Internet or across service provider
networks?
Even if you maintain some degree of bandwidth control over the WAN, can you still
guarantee end-to-end bandwidth priorities?
Proponents of circuit-switched networks maintain that users can, in fact, guarantee
bandwidth priorities on an end-to-end basis. Proponents of VoIP, meanwhile, maintain that
circuit-switched connections are wasteful.

In effect, organizations need to conduct rigorous testing of WAN connections to study the
effects of bandwidth reservation, traffic prioritization and possible remapping of packet
priorities as they traverse different portions of an end-to-end connection.
Testing QoS over the WAN connections may be easier said than done. In most instances,
service providers may not make WAN links accessible to test. Consequently, that means
QoS evaluations should focus on testing between pairs of LAN/WAN interfaces. Meaning,
you can inspect the QoS attributes that are applied to a data stream on one end of a
connection, and examine the same stream and its QoS parameters on the receiving LAN. In

VoIP Components

2009-11-20T22:09:00.001-08:00

VoIP Components
The components of VoIP include: end-user equipment, network components, call
processors, gateways and protocols.
End-user equipment is used to access the VoIP system to communicate with another
end point. Connection to the network may be physically cabled or may be wireless.
The end-user equipment may be a phone that sits on a desk or a softphone that is
installed on a PC.6 Functions include voice and possibly video communication, and may
contain instant messaging, monitoring and surveillance capabilities. 7
Though end-user equipment is often deployed on an internal, protected network, it is
usually is not individually protected by other devices (firewalls) and may be threatened if
the equipment has vulnerabilities. The threat, of course, is also dependent on the level
of security that exists on the internal network. If the device is allowed to reach or can be
reached from a public or unprotected network, there may be threats that are not
normally found on the internal network. Softphone software may have vulnerabilities,
there may be vulnerabilities in the operating system (OS) it is running on, and there may
be vulnerabilities of other applications running on the OS. Patching OSs, softphone
software and those other applications can help mitigate the risk of any threats that are
present. Additionally, some end-user equipment may have firmware upgrades that can
be applied or may be able to obtain updated software during registration.
For OS based VoIP solutions, consideration should be given to virus detection and hostbased
firewalls as well as host-based intrusion detection. Centralization of
management of these security components is best, allowing the users of the solution to
focus on their duties instead of security details, increasing productivity.
Network components include cabling, routers, switches and firewalls. Usually the
existing IP network is where a new VoIP system is installed. The impact on the IP
network is greater than merely adding more traffic. The added traffic has more of an
urgency to reach its destination than most of the data traffic that is already supported.
Switches, routers and firewalls will need to recognize and act on VoIP data in order to
keep latency down. Additional security measures, addressed later, will complicate this
process.

Emergency calls

2009-11-20T22:02:00.000-08:00

Emergency calls

The nature of IP makes it difficult to locate network users geographically. Emergency calls, therefore, cannot easily be routed to a nearby call center. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department. In the United States, at least one major police department has strongly objected to this practice as potentially endangering the public

A fixed line phone has a direct relationship between a telephone number and a physical location. A telephone number represents one pair of wires that links a location to the telephone company's exchange. Once a line is connected, the telephone company stores the home address that relates to the wires, and this relationship will rarely change. If an emergency call comes from that number, then the physical location is known.

In the IP world it is not so simple. A broadband provider may know the location where the wires terminate, but this does not necessarily allow the mapping of an IP address to that location. IP addresses are often dynamically assigned, so the ISP may allocate an address for online access, or at the time a broadband router is engaged. The ISP recognizes individual IP addresses, but does not necessarily know what physical location to which it corresponds. The broadband service provider knows the physical location, but is not necessarily tracking the IP addresses in use.

There are more complications, since IP allows a great deal of mobility. For example, a broadband connection can be used to dial a virtual private network that is employer-owned. When this is done, the IP address being used will belong to the range of the employer, rather than the address of the ISP, so this could be many kilometer away or even in another country. To provide another example: if mobile data is used, e.g., a 3G mobile handset or USB wireless broadband adapter, then the ip address has no relationship with any physical location, since a mobile user could be anywhere that there is network coverage, even roaming via another cellular company.

In short, there is no relationship between IP address and physical location, so the address itself reveals no useful information for the emergency services.

At the VoIP level, a phone or gateway may identify itself with a SIP registrar by using a username and password. So in this case, the Internet Telephony Service Provider (itsp) knows that a particular user is online, and can relate a specific telephone number to the user. However, it does not recognize how that IP traffic was engaged. Since the IP address itself does not necessarily provide location information presently, today a "best efforts" approach is to use an available database to find that user and the physical address the user chose to associate with that telephone number—clearly an imperfect solution.

VoIP Enhanced 911 (E911) is another method by which VoIP providers in the United States are able to support emergency services. The VoIP E911 emergency-calling system associates a physical address with the calling party's telephone number as required by the Wireless Communications and Public Safety Act of 1999. All "interconnected" VoIP providers (those that provide access to the PSTN system) are required to have E911 available to their customers.[31] VoIP E911 service generally adds an additional monthly fee to the subscriber's service per line, similar to analog phone service. Participation in E911 is not required and customers can opt-out or disable E911 service on their VoIP lines, if desired. VoIP E911 has been successfully used by many VoIP providers to provide physical address information to emergency service operators.

One shortcoming of VoIP E911 is that the emergency system is based on a static table lookup. Unlike in cellular phones, where the location of an E911 call can be traced using Assisted GPS or other methods, the VoIP E911 information is only accurate so long as subscribers are diligent in keeping their emergency address information up-to-date. In the United States, the Wireless Communications and Public Safety Act of 1999 leaves the burden of responsibility upon the subscribers and not the service providers to keep their emergency information up to date.
[edit] Lack of redundancy

With the current separation of the Internet and the PSTN, a certain amount of redundancy is provided. An Internet outage does not necessarily mean that a voice communication outage will occur simultaneously, allowing individuals to call for emergency services and many businesses to continue to operate normally. In situations where telephone services become completely reliant on the Internet infrastructure, a single-point failure can isolate communities from all communication, including Enhanced 911 and equivalent services in other locales.

Quality of Service

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Quality of Service

Because the underlying IP network is inherently less reliable, in contrast to the circuit-switched public telephone network, and does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (QoS) guarantees, VoIP implementations may face problems mitigating latency and jitter.

Voice, and all other data, travel in packets over IP networks with fixed maximum capacity. This system is more prone to congestion[citation needed] and DoS attacks[27] than traditional circuit switched systems; a circuit switched system of insufficient capacity will refuse new connections while carrying the remainder without impairment, while the quality of real-time data such as telephone conversations on packet-switched networks degrades dramatically.

Fixed delays cannot be controlled (as they are caused by the physical distance the packets travel), however some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, DiffServ). Fixed delays are especially problematic when satellite circuits are involved, because of long round-trip propagation delay (400–600 milliseconds for links through geostationary satellites).

A cause of packet loss and delay is congestion, which can be avoided by means of teletraffic engineering.

The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a jitter buffer upon arrival and before producing analog audio, although this further increases delay. This avoids a condition known as buffer underrun, in which the voice engine is missing audio since the next voice packet has not yet arrived. When IP packets are lost or delayed at any point in the network between VoIP users there will be a momentary dropout of voice if all packet delay and loss mechanisms cannot compensate.

It has been suggested to rely on the packetized nature of media in VoIP communications and transmit the stream of packets from the source phone to the destination phone simultaneously across different routes (multi-path routing).[28] In such a way, temporary failures have less impact on the communication quality. In capillary routing it has been suggested to use at the packet level Fountain codes or particularly raptor codes for transmitting extra redundant packets making the communication more reliable.[citation needed]

A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP Extended Report (RFC 3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC 3611 VoIP Metrics block is generated by an IP phone or gateway during a live call and contains information on packet loss rate, packet discard rate (because of jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, Mean Opinion Scores (MOS) and R factors and configuration information related to the jitter buffer.

RFC 3611 VoIP metrics reports are exchanged between IP endpoints on an occasional basis during a call, and an end of call message sent via SIP RTCP Summary Report or one of the other signaling protocol extensions. RFC 3611 VoIP metrics reports are intended to support real time feedback related to QoS problems, the exchange of information between the endpoints for improved call quality calculation and a variety of other applications.
[edit] Layer-2 Quality of Service

A number of protocols that deal with the Data link layer and Physical Layer include Quality of Service mechanisms that can be used to ensure that applications like VoIP work well even in congested scenarios. Some examples include:

* IEEE 802.11e is an approved amendment to the IEEE 802.11 standard that defines a set of Quality of Service enhancements for wireless LAN applications through modifications to the Media Access Control (MAC) layer. The standard is considered of critical importance for delay-sensitive applications, such as Voice over Wireless IP.
* IEEE 802.1p defines 8 different classes of service (including one dedicated to voice) for traffic on layer-2 wired Ethernet.
* The ITU-T G.hn standard, which provides a way to create a high-speed (up to 1 Gigabit/s) Local area network using existing home wiring (power lines, phone lines and coaxial cables). G.hn provides QoS by means of "Contention-Free Transmission Opportunities" (CFTXOPs) which are allocated to flows (such as a VoIP call) which require QoS and which have negotiated a "contract" with the network controller.

What You Need to Know Before You Deploy VoIP

2009-11-20T21:54:00.000-08:00

What You Need to Know Before You Deploy VoIP

1. The Need to Test

Are you ready?

That.s the strategic question facing data network architects and IT managers intent on migrating

delay-sensitive voice traffic onto existing IP-based network infrastructures this year.

Before IT organizations can answer yes definitively to the question, they.ll have to perform the

required due diligence to test systemic issues, as well as network components, to make sure the

network is .voice ready..

For data network operators, the prospect of merging voice onto a network optimized and tuned to

support data applications may end up a daunting assignment. Voice traffic, after all, has some

inflexible demands. Good quality conversations require low latency and jitter, low packet loss and

sufficient bandwidth.

Allow one or more of those variables to deviate from the requirements and the quality of voice

service may not be acceptable to end-users.

Making sure your network is .VoIP-ready. is a sizable undertaking. Recently, a field engineer for a

major telecommunications equipment giant involved in a major voice/data convergence project

revealed the magnitude of his convergence test plan . all 250 pages of it.

Any organization undertaking the enormous complexity of integrating voice over an IP (VoIP)

network infrastructure must understand the implications of converged network services and the

issues likely to surface from a technology assessment angle. This white paper will explore five

critical sectors IT teams must weigh as they undertake a voice data network conversion and

conduct thorough technology testing:

Functional constituents . Users must understand the issues behind the major components of

the converged network and what to test;

Testing to ensure voice quality . With data networks, the use of numerous compression

schemes for voice encoding and decoding will yield different voice.quality levels, which may

be applicable to a range of enterprise scenarios. Net planners need to understand where and

when to deploy compression to achieve different quality voice services.

Striving for successful network integration . Designers of converged network services

must comprehend the impact that existing data-oriented equipment has on the transmission of

real-time voice. And, planners also must test and evaluate WAN services to understand their

overall capacity to handle VoIP calls alongside data;

How Does it Work?

2009-11-20T21:42:00.000-08:00

How Does it Work?

The most common way VoIP works is that the end user establishes a hi speed broadband connection, a router and a VoIP gateway. Instead of a standard telephone line, the router sends the telephone calls over an Internet connection. The VoIP gateway, placed somewhere in direct proximity of the connected Internet converts the analog signals into digital format, which are further broken down into smaller chunks called 'packets', before sending it over the Internet, much like the way data is transmitted to and from the computer. These packets are sent to their final destination and instructions for bringing back into an understandable form are embedded in them. It then goes through a VoIP gateway where the packets are reconverted into the original analog format utilizing a PSTN (Public Switched Telephone Network), thereby routing the call to the number the caller has dialed blending old school technology and hi tech delivery in a seamless and instantaneous way.

What Is VoIP Security?

2009-11-20T21:33:00.000-08:00

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!