VOIP Is In Your Budget

by Jason Brashear

In today's fast paced world, IT costs have sky rocketed making it extremely difficult to stay current with todays network technologies.Many agencys are implementing VOIP (Voice Over IP) to save mula on communication expenditures. By harnessing the power of the Internet to make phone calls, long-distance contact with clients and customers doesn't have to drain company coffers. VOIP is an inexpensive way to set up a phone system for your business, but it becomes even more cost-effective if you can set it up on a budget.

One way to save mula on a VOIP network is to make use of open source software.Asterisk@home is open source software that allows you to set up VOIP without the added expense of purchasing a commercial application. An open source telephony engine and tool kit, Asterisk@home is the best VOIP program you can find that doesn't cost a cent. You can download this program for free from the Asterisk@home website and use it to seamlessly integrate VOIP into your business.The most popular open source telephony program in the world, Asterisk@home is flexible to the needs of your agencys, as it can be configured as the core of an IP or hybrid PBX, to switch calls, manage routes, enable features, and connect callers through analog and digital connections.TIP: Voice over IP (VoIP) can provide a great ROI for your business.In addition, Asterisk@home can run on Linux, Mac OS X, Open BSD, FreeBSD, and Sun Solaris with everything you expect from a higher-cost proprietary PBX. The flexibility of Asterisk@home is its greatest asset, allowing your company to handle VOIP in multiple protocols with almost all standards-based telephony equipment in conjunction with affordable hardware.Combine Asterisk@home with Cisco 7940g IP Phone hardware and you can establish a VOIP network at the office on a shoestring. Cisco 7940g IP Phone hardware lets you enjoy the benefits of VOIP without sacrificing the features and convenience of a standard business phone. Cisco's VOIP phone hardware has the potential to increase workforce productivity, increasing profits. The IP phones from Cisco are available with color LCD displays, and dynamic soft keys for call features and functions. The phones provide support for information services with XML capabilities to expand IP phone systems. You can customize Cisco IP phones so that users can access information such as stock quotes, directories, and online content through the phones.
If you purchase re certified servers to set up VOIP, you can save even more mula. Re certified servers are sold at a fraction of the original cost, but operate just as well as new servers. Re certified simply refers to the fact that the equipment was returned by another customer, but never used. The manufacturer tests the product again to ensure proper functioning, repackages it, and sells it for a reduced price. Re certified servers can't be sold as new, even though they are good as new, so they are re-sold for much less than the original asking price. Purchasing re certified servers is an effective cost-saving strategy for any company wanting to establish a VOIP network on a budget.Things to know when buying from a reseller:Contact the dealer's local Better Business Bureau:While a clean record by itself might not be definitive proof of an honest dealer , it is one more tool to use in deciding where to place your business.

Ask what professional organizations their company belongs to:There are a number of dealer/reseller organizations that companies belong to which enforce high ethical standards. If they hesitate in answering , consider it a yellow caution flag. If they answer , ask for the organization's phone number/email. Contact the organization directly and confirm if the dealer is a legitimate member. Teksavers Networking is a member in good standing with UNEDA.com. What is the refurbishment/testing process?Ask the dealer before placing any orders what steps they perform on their refurbished Cisco and refurbished Juniper prior to shipping product. You want to be sure that it involves more than blowing the dust off. Click for Teksavers Networking's testing/refurbishment process.

What is the company's return policy?Returning product is always a touchy subject with all dealers. If the product is a special order , some manufacturers may refuse to accept it back from the reseller. If the product has been opened , the value drops considerably. This is true even if it has never been used. Just breaking the seal on a box renders it as used equipment as far as a bank or flooring company is concerned. It helps to investigate this and ask any questions/concerns you have before you place any orders and avoid any unpleasantness after the fact. Dealers post usually their return policies on their Web sites as well as their invoices. Read these policies carefully or you may get stuck with the product or a hefty restocking if you order incorrectly. Dealers vary widely in their policies; from no returns unless the product is DOA to generous open ended returns with no questions asked and no re-stocking fees. Special orders may be non-cancelable and non-returnable; again , check with the individual dealer before placing that purchase order.

About the Author
Jason Brashear is a professional writer specializing in computer networking technology. He often recommends to his clients select used or refurbished cisco routers and used or refurbished cisco switches instead of paying full price for new computer networking equipment. Visit more of his writings at www.teksavers.com/content/.

Evolution of the Business Telephone System

by Peter Spackman

In the late 19th century Alexander Graham Bell first transmitted speech electrically with an order to his assistant Thomas Watson. Since then, the telephone (or the telephone system) has revolutionised the way we live, interact socially and do business, but its ultimate potential was less than apparent back in the 19th century society.
This technology developed over time into the Telephone system. In the early days of the old-fashioned switchboard operator, a phone system was a ground-breaking business tool, with the physical patching of phone extensions in order to handle calls, providing very basic telephone system functionality. These early office phone systems were referred to as PBX (Private Branch Exchange) systems and were in the form of one operator with analogue Single Line Telephone’s (SLT’s).
Electronic Key Telephone Systems were subsequently developed which provided more features for users to handle incoming call traffic by the ability to select lines via LED keys on the telephone. Hybrid Telephone systems were developed shortly after, which offer the facility to connect both analogue and electronic phones.
As technology improved and user demands increased, phone system extensions became more functional. With the advent to digital lines (ISDN technology) in the early 1990’s DDI (direct dialing in) changed the way systems operate because incoming calls could be efficiently handled by other departments / individuals as opposed to purely via reception, which had considerable benefits in terms of operational efficiency. This led to the advent of the Digital Business Telephone System, which gave more functionality to the extension users and combined with voicemail to allow staff to become their own phone managers.

Now in today’s converged communications environment, a telephone system per se will involve new terminology revolving around Internet Protocol (IP), Voice over IP (VoIP) and IP Telephony Systems. The first variant which became available is the IP-enabled office phone system, which is effectively a digital telephone system, with an IP connection attached.
Voice Over Internet Protocol is a means for handling your phone calls over internet network connections instead of a traditional phone line. Business operations of all sizes are moving to the new technology because it saves money, is easier to operate, support and upgrade.
IP Telephony is part of a complete Unified Communications network which uses the same infrastructure for voice, video and data. Effectively, one cable can facilitate versatile connections for your phones and your network. Additions, moves and changes can be done with simplicity and usually in a matter of minutes.

Whereas traditional phone calls work by allocating an entire phone line to each call, with VoIP voice data is compressed and transmitted over a computer network. This means VoIP uses substantially less bandwidth than a traditional telephone call and is consequently more cost effective. There are several other benefits to using VoIP technology:
Simplified infrastructure. Because VoIP systems work on your Network there is no need for separate cabling.

Scalability. Traditional analogue phone systems may need to be scrapped periodically as a company increases in size or restructures internally. This is not the case with VoIP systems.
Reduce operating costs. Because a VoIP exchange is based on software rather than hardware, it is easier to alter and maintain. Improve productivity. VoIP treats voice as if it were any other kind of data, so users can attach documents to voice messages or participate in virtual meetings using shared data and videoconferencing.Flexibility. If your company has its own VPN and combines it with VoIP, you can set up a fully functioning office anywhere there is a broadband connection.

About the Author
Peter Spackman has worked in the Business Telecommunications industry for over 20 years. His expertise has enabled him to become an independent Telecoms Consultant, advising companies of all sizes how to boost productivity and save money using the latest Business Communications Systems.

VoIP solutions: Enhances productivity with minimum input

The motive of any business or corporate house is to enhance the productivity and gain more profits with minimum input. The businesses on outsourcing or with a global presence are availing the benefits of internet telephony solutions for the same. With the emergence of high-end communication, cost of calling or other services over the Voice over IP has significantly lowered down compared to the traditional telephony costs.

As a matter of fact, statistics reveals that popularity of VoIP solutions has spread in the different sectors of the industry. With its benefits and high-end services, IP solutions have widened scope to residential users also. The benefits seem more clear when we take into account the fact that the internet telephony solutions provide long distance or over sea calls at lowers rates.

The increase in ratio of people using VoIP solutions for their daily or business communication needs has raised the importance of solution providers. As a matter of fact, the tough competition among providers has motivated them to offer bundled services that includes call waiting, call forwarding, 3-way calling, call conferencing and so on. Therefore, while opting for the best VoIP solution provider, users must check the additional features and compare offers from other providers.

The VoIP business solutions work smoothly on the packet switching technique. The internet is the front runner in the entire process, as it digitalises the analog voice signals to compressed IP packets. The best part of the compressed IP packets is that it allows users to transmit data, images and video through a single network. This high-end communication solution via internet is cheaper, as it detours other related surcharges, unlike PSTN services.

The impeccable connectivity options and availability of round the clock technical support enable business houses to enjoy calling with excellent voice communication. Lastly, the VoIP industry is fast increasing as it assures users and business houses with features like reliability, security, cost efficiency and scalability. Thereby, these features offer a definite hike in the revenue of any business user actively involved in this field.

by Kristen Kiya

About the Author

To know more about these solutions, visit: VoIP Solutions offered by one of the best VoIP Providers.

Introduction Voice over Internet Protocol : VoIP (5)

From Wikipedia

IP telephony in Japan

In Japan, IP telephony (IP電話, IP Denwa ?) is regarded as a service applied by VoIP technology to 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 basically 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's 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.
Telephone number for IP telephony in Japan

Since September 2002, the MIC has assigned IP telephony telephone numbers on the condition that the service falls into certain required categories of quality. Highly qualified IP telephony is assigned a telephone number. Normally the number starts with 050. But, when its quality is so high that customer almost could not tell 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.

Technical details

The two major competing standards for VoIP are the ITU standard H.323 and the IETF standard SIP. Initially H.323 was the most popular protocol, though in the "local loop" it has since been surpassed by SIP. This was primarily due to the latter's better traversal of NAT and firewalls, although recent changes introduced for H.323 have removed this advantage.[citation needed]

However, in backbone voice networks where everything is under the control of the network operator or telco, H.323 is the protocol of choice. Many of the largest carriers use H.323 in their core backbones[citation needed], and the vast majority of callers have little or no idea that their POTS calls are being carried over VoIP.

Where VoIP travels through multiple providers' softswitches the concepts of Full Media Proxy and Signalling Proxy are important. In H.323, the data is made up of 3 streams of data: 1) H.225.0 Call Signaling; 2) H.245; 3) Media. So if you are in London, your provider is in Australia, and you wish to call America, then in full proxy mode all three streams will go half way around the world and the delay (up to 500-600 ms) and packet loss will be high. However in signaling proxy mode where only the signaling flows through the provider the delay will be reduced to a more user friendly 120-150 ms.

One of the key issues with all traditional VoIP protocols is the wasted bandwidth used for packet headers. Typically, to send a G.723.1 5.6 kbit/s compressed audio path requires 18 kbit/s of bandwidth based on standard sampling rates. The difference between the 5.6 kbit/s and 18 kbit/s is packet headers. There are a number of bandwidth optimization techniques used, such as silence suppression and header compression. This can typically save 35% on bandwidth usage.

VoIP trunking techniques such as TDMoIP can reduce bandwidth overhead even further by multiplexing multiple conversations that are heading to the same destination and wrapping them up inside the same packets. Because the packet header overhead is shared between many simultaneous streams, TDMoIP can offer near toll quality audio with a per-stream packet header overhead of only about 1 kbit/s.

Introduction Voice over Internet Protocol : VoIP (4)

Corporate and telco use
Although few office environments and even fewer homes use a pure VoIP infrastructure, telecommunications providers routinely use IP telephony, often over a dedicated IP network, to connect switching stations, converting voice signals to IP packets and back. The result is a data-abstracted digital network which the provider can easily upgrade and use for multiple purposes.
Corporate customer telephone support often use IP telephony exclusively to take advantage of the data abstraction. The benefit of using this technology is the need for only one class of circuit connection and better bandwidth use. Companies can acquire their own gateways to eliminate third-party costs, which is worthwhile in some situations.
VoIP is widely employed by carriers, especially for international telephone calls. It is commonly used to route traffic starting and ending at conventional PSTN telephones.
Many telecommunications companies are looking at the IP Multimedia Subsystem (IMS) which will merge Internet technologies with the mobile world, using a pure VoIP infrastructure. It will enable them to upgrade their existing systems while embracing Internet technologies such as the Web, email, instant messaging, presence, and video conferencing. It will also allow existing VoIP systems to interface with the conventional PSTN and mobile phones.
Electronic Numbering (ENUM) uses standard phone numbers (E.164), but allows connections entirely over the Internet. If the other party uses ENUM, the only expense is the Internet connection. Virtual PBX (or IP PBX) allow companies to control their internal phone network over an existing LAN and server without needing to wire a separate telephone network. Users within this environment can then use standard telephones coupled with an FXS, IP Phones connected to a data port or a Softphone on their PC. Internal VoIP phone networks allow outbound and inbound calling on standard PSTN lines through the use of FXO adapters.

Use in Amateur Radio
Sometimes called Radio Over Internet Protocol or RoIP, Amateur radio has adopted VoIP by linking repeaters and users with Echolink, IRLP, D-STAR, Dingotel and EQSO. In fact, Echolink allows users to connect to repeaters via their computer (over the Internet) rather than by using a radio. By using VoIP Amateur Radio operators are able to create large repeater networks with repeaters all over the world where operators can access the system with actual ham radios.
Ham Radio operators using radios are able to tune to repeaters with VoIP capabilities and use DTMF signals to command the repeater to connect to various other repeaters, thus allowing them to talk to people all around the world, even with "line of sight" VHF radios.

click to call
Main article: Click-to-callClick-to-call is a service which lets users click a button and immediately speak with a customer service representative. The call can either be carried over VoIP, or the customer may request an immediate call back by entering their phone number. One significant benefit to click-to-call providers is that it allows companies to monitor when online visitors change from the website to a phone sales channel.

Legal issues in different countries
As the popularity of VoIP grows, and PSTN users switch to VoIP in increasing numbers, governments are becoming more interested in regulating VoIP in a manner similar to PSTN services,[10] especially with the encouragement of the state-mandated telephone monopolies/oligopolies in a given country, who see this as a way to stifle the new competition.
In the U.S., the Federal Communications Commission now requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers. VoIP operators in the U.S. are required to support local number portability; make service accessible to people with disabilities; pay regulatory fees, universal service contributions, and other mandated payments; and enable law enforcement authorities to conduct surveillance pursuant to the Communications Assistance for Law Enforcement Act(CALEA). VoIP operators also must provide Enhanced 911 service, disclose any limitations on their E-911 functionality to their consumers, and obtain affirmative acknowledgements of these disclosures from all consumers. VoIP operators also receive the benefit of certain U.S. telecommunications regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of "nomadic" VoIP service -- those who are unable to determine the location of their users -- are exempt from state telecommunications regulation.[11]
Some Latin American and Caribbean countries, fearful for their state owned telephone services, have imposed restrictions on the use of VoIP, including in Panama where VoIP is taxed. In Ethiopia, where the government is monopolizing telecommunication service, it is a criminal offense to offer services using VoIP. The country has installed firewalls to prevent international calls being made using VoIP. These measures were taken after a popularity in VoIP reduced the income generated by the state owned telecommunication company.
In the European Union, the treatment of VoIP service providers is a decision for each Member State's national telecoms regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" (and so should be subject to certain obligations). A general distinction is usually made between VoIP services that function over managed networks (via broadband connections) and VoIP services that function over unmanaged networks (essentially, the Internet).
VoIP services that function over managed networks are often considered to be a viable substitute for PSTN telephone services (despite the problems of power outages and lack of geographical information); as a result, major operators that provide these services (in practice, incumbent operators) may find themselves bound by obligations of price control or accounting separation.
VoIP services that function over unmanaged networks are often considered to be too poor in quality to be a viable substitute for PSTN services; as a result, they may be provided without any specific obligations, even if a service provider has "significant market power".
The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power (e.g., the obligation to offer access to emergency calls), and it is impossible to say definitively whether VoIP service providers of either type are bound by them. A review of the EU Directive is under way and should be complete by 2007.
In India, it is legal to use VoIP, but it is illegal to have VoIP gateways inside India. This effectively means that people who have PCs can use them to make a VoIP call to any number, but if the remote side is a normal phone, the gateway that converts the VoIP call to a POTS call should not be inside India.
In the UAE, it is illegal to use any form of VoIP, to the extent that websites of Skype and Gizmo Project don't work.
In the Republic of Korea, only providers registered with the government are authorized to offer VoIP services. Unlike many VoIP providers, most of whom offer flat rates, Korean VoIP services are generally metered and charged at rates similar to terrestrial calling. Foreign VoIP providers such as Vonage encounter high barriers to government registration. This issue came to a head in 2006 when internet service providers providing personal internet services by contract to United States Forces Korea members residing on USFK bases threatened to block off access to VoIP services used by USFK members of as an economical way to keep in contact with their families in the United States, on the grounds that the service members' VoIP providers were not registered. A compromise was reached between USFK and Korean telecommunications officials in January 2007, wherein USFK service members arriving in Korea before June 1, 2007 and subscribing to the ISP services provided on base may continue to use their U.S.-based VoIP subscription, but later arrivals must use a Korean-based VoIP provider, which by contract will offer pricing similar to the flat rates offered by U.S. VoIP providers.[12]

Introduction Voice over Internet Protocol : VoIP (3)

Security
Many consumer VoIP solutions do not support encryption yet, although having a secure phone is much easier to implement with VoIP than traditional phone lines. As a result, it is relatively easy to eavesdrop on VoIP calls and even change their content.[9] There are several open source solutions that facilitate sniffing of VoIP conversations. A modicum of security is afforded due to patented audio codecs that are not easily available for open source applications, however such security through obscurity has not proven effective in the long run in other fields. Some vendors also use compression to make eavesdropping more difficult. However, real security requires encryption and cryptographic authentication which are not widely available at a consumer level. The existing secure standard SRTP and the new ZRTP protocol is available on Analog Telephone Adapters(ATAs) as well as various softphones. It is possible to use IPsec to secure P2P VoIP by using opportunistic encryption. Skype does not use SRTP, but uses encryption which is transparent to the Skype provider.
The Voice VPN solution provides secure voice for enterprise VoIP networks by applying IPSec encryption to the digitized voice stream.

Pre-Paid Phone Cards
VoIP has become an important technology for phone services to travelers, migrant workers and expatriates, who either, due to not having a fixed or mobile phone or high overseas roaming charges, choose instead to use VoIP services to make their phone calls. Pre-paid phone cards can be used either from a normal phone or from Internet cafes that have phone services. Developing countries and areas with high tourist or immigrant communities generally have a higher uptake.

Caller ID
Caller ID support among VoIP providers varies, although the majority of VoIP providers now offer full Caller ID with name on outgoing calls. When calling a traditional PSTN number from some VoIP providers, Caller ID is not supported.
In a few cases, VoIP providers may allow a caller to spoof the Caller ID information, making it appear as though they are calling from a different number. Business grade VoIP equipment and software often makes it easy to modify caller ID information. Although this can provide many businesses great flexibility, it is also open to abuse.

VoIM
Voice over Instant Messaging (VoIM) presents VoIP as one communication mode among several, with an IM user interface (contact list and presence) as the primary user experience. Many instant messenger services added client-to-client or client-to-PSTN VoIP in the mid-2000s.


Adoption

Mass-market telephony
A major development starting in 2004 has been the introduction of mass-market VoIP services over broadband Internet access services, in which subscribers make and receive calls as they would over the PSTN. Full phone service VoIP phone companies provide inbound and outbound calling with Direct Inbound Dialing. Many offer unlimited calling to the U.S., and some to Canada or selected countries in Europe or Asia as well, for a flat monthly fee.
These services take a wide variety of forms which can be more or less similar to traditional POTS. At one extreme, an analog telephone adapter (ATA) may be connected to the broadband Internet connection and an existing telephone jack in order to provide service nearly indistinguishable from POTS on all the other jacks in the residence. This type of service, which is fixed to one location, is generally offered by broadband Internet providers such as cable companies and telephone companies as a cheaper flat-rate traditional phone service. Often the phrase "VoIP" is not used in selling these services, but instead the industry has marketed the phrases "Internet Phone", "Digital Phone" or "Softphone" which is aimed at typical phone users who are not necessarily tech-savvy. Typically, the provider touts the advantage of being able to keep one's existing phone number.
At the other extreme are services like Gizmo Project and Skype which rely on a software client on the computer in order to place a call over the network, where one user ID can be used on many different computers or in different locations on a laptop. In the middle lie services which also provide a telephone adapter for connecting to the broadband connection similar to the services offered by broadband providers (and in some cases also allow direct connections of SIP phones) but which are aimed at a more tech-savvy user and allow portability from location to location. One advantage of these two types of services is the ability to make and receive calls as one would at home, anywhere in the world, at no extra cost. No additional charges are incurred, as call diversion via the PSTN would, and the called party does not have to pay for the call. For example, if a subscriber with a home phone number in the U.S. or Canada calls someone else within his local calling area, it will be treated as a local call regardless of where that person is in the world. Often the user may elect to use someone else's area code as his own to minimize phone costs to a frequently called long-distance number.
For some users, the broadband phone complements, rather than replaces, a PSTN line, due to a number of inconveniences compared to traditional services. VoIP requires a broadband Internet connection and, if a telephone adapter is used, a power adapter is usually needed. In the case of a power failure, VoIP services will generally not function. Additionally, a call to the U.S. emergency services number 9-1-1 may not automatically be routed to the nearest local emergency dispatch center, and would be of no use for subscribers outside the U.S. This is potentially true for users who select a number with an area code outside their area. Some VoIP providers offer users the ability to register their address so that 9-1-1 services work as expected.
Another challenge for these services is the proper handling of outgoing calls from fax machines, TiVo/ReplayTV boxes, satellite television receivers, alarm systems, conventional modems or FAXmodems, and other similar devices that depend on access to a voice-grade telephone line for some or all of their functionality. At present, these types of calls sometimes go through without any problems, but in other cases they will not go through at all. And in some cases, this equipment can be made to work over a VoIP connection if the sending speed can be changed to a lower bits per second rate. If VoIP and cellular substitution becomes very popular, some ancillary equipment makers may be forced to redesign equipment, because it would no longer be possible to assume a conventional voice-grade telephone line would be available in almost all homes in North America and Western-Europe. The TestYourVoIP website offers a free service to test the quality of or diagnose an Internet connection by placing simulated VoIP calls from any Java-enabled Web browser, or from any phone or VoIP device capable of calling the PSTN network.

Introduction Voice over Internet Protocol : VoIP (2)

Reliability
Conventional phones are connected directly to telephone company phone lines, which in the event of a power failure are kept functioning by back-up generators or batteries located at the telephone exchange. However, household VoIP hardware uses broadband modems and other equipment powered by household electricity, which may be subject to outages in the absence of a uninterruptible power supply or generator. Early adopters of VoIP may also be users of other phone equipment, such as PBX and cordless phone bases, that rely on power not provided by the telephone company. Even with local power still available, the broadband carrier itself may experience outages as well. While the PSTN has been matured over decades and is typically reliable, most broadband networks are less than 10 years old, and even the best are still subject to intermittent outages. Furthermore, consumer network technologies such as cable and DSL often are not subject to the same restoration service levels as the PSTN or business technologies such as T-1 connection.

Quality of service
Some broadband connections may have less than desirable quality. Where IP packets are lost or delayed at any point in the network between VoIP users, there will be a momentary drop-out of voice. This is more noticeable in highly congested networks and/or where there are long distances and/or interworking between end points. Technology has improved the reliability and voice quality over time and will continue to improve VoIP performance as time goes on.
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). 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.
A number of protocols have been defined to support the reporting of QoS/QoE for VoIP calls. These include RTCP XR (RFC3611), SIP RTCP Summary Reports, H.460.9 Annex B (for H.323), H.248.30 and MGCP extensions. The RFC3611 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 (due to jitter), packet loss/discard burst metrics (burst length/density, gap length/density), network delay, end system delay, signal / noise / echo level, MOS scores and R factors and configuration information related to the jitter buffer.
RFC3611 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. RFC3611 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.

Difficulty with sending faxes
The support of sending faxes over VoIP is still limited. The existing voice codecs are not designed for fax transmission. An effort is underway to remedy this by defining an alternate IP-based solution for delivering Fax-over-IP, namely the T.38 protocol. Another possible solution to overcome the drawback is to treat the fax system as a message switching system, which does not need 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.

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, and are impossible on some VoIP systems. Sometimes, VoIP systems may route emergency calls to a non-emergency phone line at the intended department. In the US, at least one major police department has strongly objected to this practice as potentially endangering the public.
Moreover, in the event that the caller is unable to give an address, emergency services may be unable to locate them in any other way. Following the lead of mobile phone operators, several VoIP carriers are already implementing a technical work-around.[citation needed] For instance, one large VoIP carrier requires the registration of the physical address where the VoIP line will be used. When you dial the emergency number for your country, they will route it to the appropriate local system. They also maintain their own emergency call center that will take non-routable emergency calls (made, for example, from a software based service that is not tied to any particular physical location) and then will manually route your call once learning your physical location.[citation needed]
e911 is another method by which VOIP providers in the US are able to support emergency services. The e911 emergency-calling system automatically associates a physical address with the calling party's telephone number as required by the Wireless Communications and Public Safety Act of 1999 and is being successfully used by many VOIP providers to provide physical address information to emergency service operators.

Integration into global telephone number system
While the traditional Plain Old Telephone Service (POTS) and mobile phone networks share a common global standard (E.164) which allocates and identifies any specific telephone line, there is no widely adopted similar standard for VoIP networks. Some allocate an E.164 number which can be used for VoIP as well as incoming/external calls. However, there are often different, incompatible schemes when calling between VoIP providers which use provider specific short codes.

Single point of calling
With hardware VoIP solutions it is possible to connect the VoIP router into the existing central phone box in the house and have VoIP at every phone already connected. Software based VoIP services require the use of a computer, so they are limited to single point of calling, though telephone sets are now available, allowing them to be used without a PC. Some services provide the ability to connect WiFi SIP phones so that service can be extended throughout the premises, and off-site to any location with an open hotspot.[5] However, note that many hotspots require browser-based authentication, which most SIP phones do not support.

Mobile phones & Hand held Devices
Telcos and consumers have invested billions of dollars in mobile phone equipment. In developed countries, mobile phones have achieved nearly complete market penetration, and many people are giving up landlines and using mobiles exclusively. Given this situation, it is not entirely clear whether there would be a significant higher demand for VoIP among consumers until either public or community wireless networks have similar geographical coverage to cellular networks (thereby enabling mobile VoIP phones, so called WiFi phones or VoWLAN) or VoIP is implemented over 3G networks. However, "dual mode" telephone sets, which allow for the seamless handover between a cellular network and a WiFi network, are expected to help VoIP become more popular.
Phones like the NEC N900iL, and later many of the Nokia Eseries and several WiFi enabled mobile phones have SIP clients hardcoded into the firmware. Such clients operate independently of the mobile phone network unless a network operator decides to remove the client in the firmware of a heavily branded handset. Some operators such as Vodafone actively try to block VoIP traffic from their network and therefore most VoIP calls from such devices are done over WiFi.Several WiFi only IP hardphones exist, most of them supporting either Skype or the SIP protocol. These phones are intended as a replacement for PSTN based cordless phones but can be used anywhere where WiFi internet access is available.
Another addition to hand held devices are ruggedized bar code type devices that are used in warehouses and retail environments. These type of devices rely on "inside the 4 walls" type of VoIP services that do not connect to the outside world and are solely to be used from employee to employee communications.