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VoIP in Detail

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Voice over Internet Protocol (VoIP) is a protocol optimized for the transmission of voice through the Internet or other. Packet Switched Networks VoIP is often used abstractly to refer to the actual transmission of voice (rather than the protocol implementing it). VoIP is also known as IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband. "VoIP" is pronounced voyp, IPA: /voip/.Companies providing VoIP service are commonly referred to as providers, and protocols which are 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 providers. Some cost savings are due to utilizing a single network to carry voice and data, especially where users have existing underutilized network capacity that can carry VoIP at no additional cost. VoIP to VoIP phone calls are sometimes free, while VoIP to public switched telephone networks, PSTN, may have a cost that is borne by the VoIP user.

Voice over IP protocols carry telephony signals as digital audio, typically reduced in data rate using speech data compression techniques, encapsulated in a data packet stream over IP.

There are two types of PSTN to VoIP services: Direct Inward Dialing (DID) and access numbers. DID will connect the caller directly to the VoIP user while access numbers require the caller to input the extension number of the VoIP user.History

Functionality

VoIP can facilitate tasks and provide services that may be more difficult to implement or expensive using the more traditional PSTN. Examples include:The ability to transmit more than one telephone call down the same broadband-connected telephone line. This can make VoIP a simple way to add an extra telephone line to a home or office. 3-way calling, call forwarding, automatic redial, and caller ID; features that traditional telecommunication companies (telcos) normally charge extra for. Secure calls using standardized protocols (such as Secure Real-time Transport Protocol.) Most of the difficulties of creating a secure phone over traditional phone lines, like digitizing and digital transmission are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream. Location independence. Only an internet connection is needed to get a connection to a VoIP provider. For instance, call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection. Integration 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. Implementation

Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service (known as QoS) guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round trip propagation delay (400 milliseconds to 600 milliseconds for geostationary satellite). 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 in the voice engine.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. Other methods to traverse firewalls involve using protocols such as STUN or ICE.VoIP challenges:Available bandwidth Delay/Network Latency Packet loss Jitter Echo Security Reliability Pulse dialing to DTMF translation Many VoIP providers do not translate pulse dialing from older phones to DTMF. The VoIP user may use a VoIP Pulse to Tone Converter, if needed.[citation needed]Fixed 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 teletraffic engineering.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 audio, although 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.Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of the transmit and receive signal at the receiving end.R

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 (QoS)

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.

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.

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.

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.

Technical details

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.

The above information has been extracted from http://en.wikipedia.org/wiki/VoIP and we give thanks to the author for providing such in depth information.