Voice over Internet Protocol (also called VoIP [pronounced "voyp"], IP Telephony, Internet telephony, and Broadband Phone) is the routing of voice conversations over the Internet or through any other IP-based network.
Protocols used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols. They may be viewed as commercial realizations of the experimental Network Voice Protocol (1973) invented for the ARPANET.
Voice over IP traffic can be deployed on any IP network, including ones lacking a connection to the rest of the Internet, for instance on a local area network.
In general, phone service via VoIP is free or costs less than equivalent service from traditional sources but similar to alternative PSTN (Public Switched Telephone Network) service providers. Some cost savings are due to using a single network to carry voice and data, especially where users have existing under-utilized network capacity they can use for VoIP at no additional cost. VoIP to VoIP phone calls on any provider are typically free, whilst VoIP to PSTN calls generally costs the VoIP user. Internet Bandwidth.
There are two types of PSTN to VoIP services: DID and access numbers. DID will connect the caller directly to the VoIP user while access numbers requires the caller to input the extension number of the VoIP user. Access numbers are usually charged as a local call to the caller and free to the VoIP user while DID usually has a monthly fee. There are also DID that are free to the VoIP user but is chargeable to the caller.
VoIP can facilitate tasks that may be more difficult to achieve using traditional phone networks:
Incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network. Take your VoIP phone with you on a trip, and wherever you connect to the Internet, you can receive incoming calls. Free phone numbers for use with VoIP are available in the USA, UK and other countries from organizations such as VoIP User. More >> Voice over IP (VoIP) AltiGen Communications released the world's first VoIP phone system in 1996. Call center agents using VoIP phones can work from anywhere with a sufficiently fast Internet connection. Many VoIP packages include PSTN features that most telcos normally charge extra for, or may be unavailable from your local telco, such as 3-way calling, call forwarding, automatic redial, etc.VoIP allows users to travel anywhere in the world and still make and receive phone calls:
Subscribers of phone-line replacement services can make and receive local phone calls regardless of their location. For example, if a user has a New York City phone number and is traveling in Europe and someone calls the phone number, it will ring in Europe. Conversely, if a call is made from Europe to New York City, it will be treated as a local call. Of course, there must be a connection to the Internet e. g. Up to 30000 users. WiFi to make all of this possible. Users of Instant Messenger based VoIP services can also travel anywhere in the world and make and receive phone calls. VoIP phones can integrate with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books and passing information about whether others (e. g. friends or colleagues) are available online to interested parties.VoIP technology still has a few shortcomings that have led some to believe that it is not ready for widespread deployment. However, many industry analysts predicted that 2005 was the "Year of Inflection," where more IP PBX ports shipped than legacy digital PBX ports. This diagram displays a Stand Alone VoIP phone system with conventional voice services, T1, PRI, or POTS lines.
Because IP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved. 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. This functionality is usually accomplished by means of a jitter buffer.
Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from a protected enterprise network. Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. Stand Alone Phone System with PRI,T1, or POTS voice service. Other methods to traverse firewalls involve using protocols such as STUN or ICE.
VOIP challenges:
DelayPacket lossJitterEchoFixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive (see, for example, Diffserv).
The principal cause of packet loss is congestion, which can be controlled by congestion management and avoidance. Carrier VoIP networks avoid congestion by means of traffic engineering.
Variation in delay is called Jitter. The effects of jitter can be mitigated by storing voice packets in a buffer (called a play-out buffer) upon arrival, before playing them out. This avoids a condition known as buffer underrun, in which the playout process runs out of voice data to play because the next voice packet has not yet arrived. VoIP forum provide information on best voip Service such as voip call termination, voip equipment, voip minutes, voip routes, voip hardware, voip phone service at cheap rates with the best voip.
Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of the transmit and receive signal at the receiving end.
VoIP technology does not necessarily require broadband Internet access, but this usually supports better quality of service. A sizable percentage of homes today are connected to the Internet through DSL, which requires a traditional phone line. Having to pay for VoIP in addition to both a basic phone line and broadband Internet access reduces the potential benefits of VoIP. However, some regional telephone companies now offer DSL service without the phone (often called "naked DSL" or "dry loop DSL"), thus saving you money when you switch to VoIP. VoIP can also be used with Cable Internet instead of DSL, potentially eliminating the need for a traditional phone line entirely.
Conventional telephones 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. Tripod on Lycos, established in 1994, is one of the leading personal publishing communities on the Web. However, household VoIP hardware uses broadband modems and other equipment powered by household electricity, which may be subject to outages dictating the use of an uninterruptible power supply or generator to ensure availability during power outages. 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 PTSN has been matured over decades and is typically extremely 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 PTSN or business technologies such as T-1 connection.
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 allows you to initiate and terminate calls between your VoIP phone and the public phone system. This is more noticeable in highly congested networks and/or where there is 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.
The nature of IP makes it difficult to geographically locate network users. Emergency calls, therefore, cannot easily be routed to a nearby call center, and are impossible on some VoIP systems. 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. For instance, one large VOIP carrier requires the registration of the physical address the VOIP line will be used at. Voice over IP (VoIP) phone systems buying advice and free Voice over IP quotes from leading vendors. When you dial 911 (or the applicable 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 911 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.
The United States government had set a deadline, requiring VoIP carriers to implement E911, however, the deadline is being appealed by several of the leading VoIP companies.
This is a different situation with IPBX systems, where these corporate systems often have full E911 capabilities built into the system.
While the traditional Plain Old Telephone System (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. AltiGen is the pioneer of VoIP phone systems that help businesses reduce costs,
tie office locations together, improve customer retention and more. 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.
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