Voice over IP (VoIP):- Voice over internet Protocol is a group and methodology of technologies for the delivery of voice communications and transmission sessions over internet Protocol (IP) networks, like the web. Different terms unremarkable related to VoIP area unit broadband telephone service, broadband telephone, internet telecommunication, and internet Protocol (IP) telecommunication.
The term internet telecom specifically refers of communications services (voice, fax, SMS, voice-messaging) to the provisioning over the general public web, instead of via the general public switched telephone network (PSTN). The steps and principles concerned in originating VoIP telephone calls are similar to traditional digital telephone and channel setup, conversion of the analog voice signals, involve signaling, and encryption. Instead of being transmitted over a circuit-switched network, however, the digital info is packetized, and transmission happens as internet Protocol (IP) packets over a packet-switched network. Such transmission entails careful concerns about resource management totally different from time-division multiplexing (TDM) networks.
Early providers of voice-over- internet-Protocol (VoIP) services offered business technical and models solutions that reflected the design of the legacy telephone network. Second-generation providers, like Skype, providing the advantage of free calls, have built closed networks for private user bases and convenience whereas probably charging for access to different communication networks, like the PSTN. This has restricted the freedom of users to mix-and-match third-party hardware and software system. Third-generation providers, have adopted the idea, like Google talk of federate voice over IP ( VoIP)—which is a departure from the design of the legacy networks. These solutions typically enable dynamic interconnection on the web when a user needs to place a call between users on any two domains.
Voice over IP (VoIP) systems use session communication and management protocols to regulate set-up, the communication, and tear-down of phone calls. By streaming media, they transport audio streams over information science networks victimization special media delivery protocols that write voice, video, audio with audio codec’s, and video codec’s as Digital audio. Numerous codec’s exist that optimize the media stream supported application needs and network bandwidth; some implementations suppose narrowband and compressed speech, whereas others support sound reproduction stereo codec’s. Some standard codec’s embrace μ-law and a-law versions of G.711, G.722, a preferred open supply voice codec called iLBC, a codec that solely uses eight kbit/s every means known as G.729, and lots of others.
VoIP is offered on several personal computers, SmartPhones, and on web access devices. Calls and SMS text messages are also sent over 3G/4G or Wi-Fi.
Voice over IP has been implemented in various ways using protocols.
The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. However, since the development of newer, less complex protocols such as SIP and MGCP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) has gained widespread Voice over IP (VoIP) market penetration.
Examples of the VoIP protocols are:
- H.248 (Megaco)
- Media Gateway Control Protocol (MGCP)
- Skype protocol
- Secure Real-time Transport Protocol (SRTP)
- Session Description Protocol (SDP)
- Session Initiation Protocol (SIP)
- Inter-Asterisk eXchange (IAX)
- Jingle XMPP VoIP extensions
- Real-time Transport Protocol (RTP)
- RTP Control Protocol (RTCP)
The Media VoIP Gateway interfaces the advanced media stream, in order to finish making the way for voice and also information media. It incorporates the interface for associating the standard PSTN systems with the ATM and Inter Protocol systems. The Ethernet interfaces are additionally incorporated into the current frameworks, which are exceptionally intended to connection calls that are passed through the VoIP.
E.164 is a worldwide numbering standard for both the PSTN and PLMN. Most VoIP executions bolster E.164 to permit calls to be directed to and from VoIP endorsers and the PSTN/PLMN. VoIP executions can likewise permit other distinguishing proof strategies to be utilized. For instance, Skype permits endorsers of pick “Skype names” (usernames) though SIP executions can utilize URIs like email locations. Regularly VoIP usage utilize strategies for making an interpretation of non-E.164 identifiers to E.164 numbers and the other way around, for example, the Skype-In administration gave by Skype and the ENUM administration in IMS and SIP.
Reverberation can likewise be an issue for PSTN coordination. Normal reasons for reverberation incorporate impedance bungles in simple hardware and acoustic coupling of the transmit and get signal at the less than desirable end.
A phone associated with an area line has an immediate relationship between a phone number and a physical area, which is kept up by the phone organization and accessible to crisis responders by means of the national crisis reaction administration focuses in type of crisis supporter records. At the point when a crisis call is gotten by a middle the area is consequently decided from its databases and showed on the administrator console.
In IP telephony, no such direct connection in the middle of area and interchanges end point exists. Indeed, even a supplier having equipment base, for example, a DSL supplier, may just know the estimated area of the gadget, in view of the IP address apportioned to the system switch and the known administration address. Be that as it may, some ISPs don’t track the programmed task of IP locations to client hardware.
IP correspondence accommodates gadget portability. For instance, a private broadband association might be utilized as a connection to a virtual private system of a corporate substance, in which case the IP location being utilized for client interchanges may fit in with the endeavor, not being the system location of the private ISP. Such off-premises augmentations may show up as a major aspect of an upstream IP PBX. On cell phones, e.g., a 3G handset or USB remote broadband connector, the IP address has no association with any physical area known not telephony administration supplier, since a versatile client could be anyplace in a district with system scope, notwithstanding meandering by means of another cell organization.
At the VoIP level, a telephone or portal may recognize itself with a Session Initiation Protocol (SIP) recorder by its record qualifications. In such cases, the Internet telephony administration supplier (ITSP) just realizes that a specific client’s hardware is dynamic. Administration suppliers regularly give crisis reaction administrations by concurrence with the client who enrolls a physical area and concurs that crisis administrations are just given to that address if a crisis number is called from the IP gadget.
Such crisis administrations are given by VoIP sellers in the United States by a framework called Enhanced 911 (E911), in light of the Wireless Communications and Public Safety Act of 1999. The VoIP E911 crisis calling framework relates a physical location with the calling gathering’s phone number. All VoIP suppliers that give access to general society exchanged phone system are required to execute E911, an administration for which the supporter might be charged. Notwithstanding, end-client interest in E911 is not required and clients may quit the administration.
The VoIP E911 framework depends on a static table lookup. Dissimilar to in PDAs, where the area of an E911 call can be followed utilizing helped GPS or different strategies, the VoIP E911 data is just exact inasmuch as supporters, who have the legitimate obligation, are persevering in keeping their crisis address data current.
Nearby number versatility (LNP) and Mobile number compactness (MNP) likewise affect VoIP business. In November 2007, the Federal Communications Commission in the United States discharged a request stretching out number versatility commitments to interconnected VoIP suppliers and transporters that bolster VoIP suppliers. Number versatility is an administration that permits an endorser of select another phone transporter without requiring another number to be issued. Ordinarily, it is the obligation of the previous bearer to “guide” the old number to the undisclosed number doled out by the new transporter. This is accomplished by keeping up a database of numbers. A dialed number is at first gotten by the first transporter and immediately rerouted to the new bearer. Various porting references must be kept up regardless of the fact that the endorser comes back to the first transporter. The FCC commands bearer consistence with these buyer insurance stipulations.
A voice call starting in the VoIP environment likewise confronts difficulties to achieve its destination if the number is directed to a cellular telephone number on a customary versatile bearer. VoIP has been distinguished in the past as a Least Cost Routing (LCR) framework, which depends on checking the destination of every phone call as it is made, and afterward sending the call by means of the system that will cost the client the minimum. This rating is liable to some level headed discussion given the multifaceted nature of call directing made by number convenientce. With GSM number versatility now set up, LCR suppliers can no more depend on utilizing the system root prefix to decide how to highway a call. Rather, they should now decide the genuine system of each number before directing the call.
In this manner, VoIP arrangements likewise need to handle MNP while steering a voice call. In nations without a focal database, similar to the UK, it may be important to question the GSM system about which home system a cell telephone number fits in with. As the fame of VoIP increments in the endeavor markets as a result of slightest expense directing choices, it needs to give a specific level of unwavering quality when taking care of calls.
MNP checks are imperative to guarantee that this nature of administration is met. Taking care of MNP lookups before directing a call gives some affirmation that the voice call will really work.
Support for fax has been dangerous in numerous Voice over IP (VoIP) usage, as most voice digitization and pressure codecs are streamlined for the representation of the human voice and the best possible timing of the modem signals can’t be ensured in a bundle based, association less system. An option IP-based answer for conveying fax-over-IP called T.38 is accessible. Sending faxes utilizing Voice over IP (VoIP) is in some cases alluded to as FoIP, or fax over IP.
The T.38 convention is intended to make up for the contrasts between customary parcel less interchanges over simple lines and bundle based transmissions which are the premise for IP correspondences. The fax machine could be a conventional fax machine associated with the PSTN, or an ATA box (or comparable). It could be a fax machine with a RJ-45 connector connected straight to an IP system, or it could be a PC putting on a show to be a fax machine. Initially, T.38 was intended to utilize UDP and TCP transmission strategies over an IP system. TCP is more qualified for use between two IP gadgets. Be that as it may, more established fax machines, associated with a simple framework, advantage from UDP close constant attributes because of the “no recuperation principle” when a UDP parcel is lost or a blunder happens amid transmission. UDP transmissions are favored as they don’t require testing for dropped parcels and in that capacity since each T.38 bundle transmission incorporates a greater part of the information sent in the earlier bundle, a T.38 end point has a higher level of achievement in re-gathering the fax transmission once more into its unique structure for elucidation by the end gadget. This trying to conquer the obstructions of reenacting continuous transmissions utilizing bundle based convention.
There have been redesigned forms of T.30 to determine the fax over IP issues, which is the center fax convention. Some more current top of the line fax machines have T.38 worked in abilities which permit the client to connect right to the system and transmit/get faxes in local T.38 like the Ricoh 4410NF Fax Machine. An interesting element of T.38 is that every parcel contains a bit of the principle information sent in the past bundle. With T.38, two progressive lost bundles are expected to really lose any information. The information one will lose may be a little piece, however with the right settings and blunder revision mode, there is an improved probability that they will get enough of the transmission to fulfill the prerequisites of the fax machine for yield of the sent record.
While some late-show simple phone connectors (ATAs) bolster T.38, uptake has been restricted the same number of voice-over-IP suppliers perform minimum expense directing which chooses the slightest costly PSTN passage in the called city for an outbound message. There is ordinarily no way to guarantee that that portal is T.38 fit. Suppliers frequently put their own particular gear, (for example, an Asterisk PBX establishment) in the sign way, which makes extra issues as each connection in the chain must be T.38 mindful for the convention to work. Comparative issues emerge if a supplier is buying neighborhood coordinate internal dial numbers from the least bidder in every city, the same number of these may not be T.38 empowered.
There are likewise executions of fax over IP that detours telephony totally, by sending the filtered picture specifically over another convention, similar to email.
The security worries of Voice over IP (VoIP) phone frameworks are like those of any Internet-associated gadget. This implies programmers who think about these vulnerabilities can found disavowal of-administration assaults, harvest client information, record discussions and trade off phone message messages. The nature of web association decides the nature of the calls. Voice over IP (VoIP) telephone benefit additionally won’t work if there is force blackout and when the web association is down. The 9-1-1 or 112 administration gave by Voice over IP (VoIP) telephone administration is additionally not quite the same as simple telephone which is connected with an altered location. The crisis focus will most likely be unable to decide your area taking into account your virtual telephone number. Traded off Voice over IP (VoIP) client record or session certifications may empower an assailant to acquire generous charges from outsider administrations, for example, long-separation or worldwide phone calling.
The specialized subtle elements of numerous Voice over IP (VoIP) conventions make challenges in steering Voice over IP (VoIP) activity through firewalls and system address interpreters, used to interconnect to travel systems or the Internet. Private session fringe controllers are regularly utilized to empower Voice over IP (VoIP) calls to and from secured systems. Different strategies to navigate NAT gadgets include assistive conventions, for example, STUN and Interactive Connectivity Establishment (ICE).
Numerous shopper Voice over IP (VoIP) arrangements don’t bolster encryption of the flagging way or the media, however securing a Voice over IP (VoIP) telephone is adroitly less demanding to actualize than on conventional phone circuits. An aftereffect of the absence of encryption is that it is generally simple to listen in on Voice over IP (VoIP) calls when access to the information system is conceivable. Free open-source arrangements, for example, Wireshark, encourage catching Voice over IP (VoIP) discussions.
Guidelines for securing Voice over IP (VoIP) are accessible in the Secure Real-time Transport Protocol (SRTP) and the ZRTP convention for simple telephony connectors and also for some softphones. IPsec is accessible to secure point-to-point Voice over IP (VoIP) at the vehicle level by utilizing astute encryption.
Government and military associations use different efforts to establish safety to ensure Voice over IP (VoIP) activity, for example, voice over secure IP (VoSIP), secure voice over IP (SVoice over IP (VoIP)), and secure voice over secure IP (SVoSIP). The refinement lies in whether encryption is connected in the phone or in the network or both. Secure voice over secure IP is expert by scrambling Voice over IP (VoIP) with conventions, for example, SRTP or ZRTP. Secure voice over IP is proficient by utilizing Type 1 encryption on a grouped system, as SIPRNet Public Secure Voice over IP (VoIP) is likewise accessible with free GNU programs and in numerous prevalent business Voice over IP (VoIP) programs by means of libraries, for example, ZRTP.
Caller IDsupport among Voice over IP (VoIP) suppliers differs yet is given by the greater part of Voice over IP (VoIP) suppliers. Numerous Voice over IP (VoIP) administration suppliers permit guests to design subjective Caller IDdata, along these lines allowing ridiculing assaults. Business-grade Voice over IP (VoIP) gear and programming frequently makes it simple to alter Caller IDdata, giving numerous organizations incredible adaptability.
The United States instituted the Truth in Caller ID Act of 2009 on December 22, 2010. This law makes it a wrongdoing to “purposely transmit deluding or off base guest distinguishing proof data with the goal to cheat, cause hurt, or wrongfully acquire anything of worth …”. Rules executing the law were embraced by the Federal Communications Commission on June 20, 2011.
Support for other telephony devices
Another test for Voice over IP (VoIP) usage is the correct treatment of active calls from other telephony gadgets, for example, advanced video recorders, satellite TV collectors, alert frameworks, customary modems and other comparable gadgets that rely on upon access to a PSTN phone line for a few or the majority of their usefulness.
These sorts of calls here and there complete with no issues, however in different cases they come up short. On the off chance that Voice over IP (VoIP) and cell substitution turns out to be extremely well known, some subordinate hardware producers might be compelled to overhaul gear, since it would never again be conceivable to expect a routine PSTN phone line would be accessible in buyers’ homes.
Compatibility with traditional analog telephone sets
Most simple phone connectors don’t unravel dial beats produced by more established phones, supporting just touch-tone. Heartbeat to-tone converters are economically accessible; a client reports that a couple of particular ATA models, (for example, the Grandstream 502) perceive beat dial specifically, yet are ineffectively recorded and give no affirmation that more current models in the same arrangement will hold this similarity. This in any case, will work for one Voice over IP (VoIP) discussion at once.
Voice over IP (VoIP) can be an advantage for decreasing correspondence and framework costs. Cases include:
Directing telephone brings over existing information systems to evade the requirement for discrete voice and information systems.
The capacities to transmit more than one phone bring over a solitary broadband association.
Secure calls utilizing institutionalized conventions, (for example, Secure Real-time Transport Protocol). The majority of the challenges of making a safe phone association over conventional telephone lines, for example, digitizing and computerized transmission, are as of now set up with Voice over IP (VoIP). It is just important to encode and confirm the current information stream.
Used existing system foundation to minimize the working expense.
Wiping out the need of contracting work force to welcome and appropriate approaching calls with the utilization of a Virtual PBX
Voice over IP (VoIP) can be scaled effortlessly, permitting it to develop with the business without prohibitive contracts.
Regulatory and legal issues
As the prevalence of Voice over IP (VoIP) develops, governments are turning out to be more keen on directing Voice over IP (VoIP) in a way like PSTN administrations.
All through the creating scene, nations where regulation is frail or caught by the overwhelming administrator, limitations on the utilization of Voice over IP (VoIP) are forced, incorporating into Panama where Voice over IP (VoIP) is saddled, Guyana where Voice over IP (VoIP) is restricted and India where its retail business deals is permitted yet just for long separation administration. In Ethiopia, where the administration is nationalizing telecom administration, it is a criminal offense to offer administrations utilizing Voice over IP (VoIP). The nation has introduced firewalls to counteract worldwide calls being made utilizing Voice over IP (VoIP). These measures were taken after the fame of Voice over IP (VoIP) decreased the pay created by the state possessed telecom organization.
In the United States, the Federal Communications Commission requires all interconnected Voice over IP (VoIP) administration suppliers to agree to prerequisites similar to those for customary information transfers administration suppliers. Voice over IP (VoIP) operators in the US 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).
Operators of “Interconnected” Voice over IP (VoIP) (fully connected to the PSTN) are mandated to provide Enhanced 911 service without special request, provide for customer location updates, clearly disclose any limitations on their E-911 functionality to their consumers, obtain affirmative acknowledgements of these disclosures from all consumers, and ‘may not allow their customers to “opt-out” of 911 service. Voice over IP (VoIP) operators also receive the benefit of certain US telecommunications regulations, including an entitlement to interconnection and exchange of traffic with incumbent local exchange carriers via wholesale carriers. Providers of “nomadic” Voice over IP (VoIP) service—those who are unable to determine the location of their users—are exempt from state telecommunications regulation.
Another legal issue that the US Congress is debating concerns changes to the Foreign Intelligence Surveillance Act. The issue in question is calls between Americans and foreigners. The National Security Agency (NSA) is not authorized to tap Americans’ conversations without a warrant—but the Internet, and specifically Voice over IP (VoIP) does not draw as clear a line to the location of a caller or a call’s recipient as the traditional phone system does. As Voice over IP (VoIP)’s low cost and flexibility convinces more and more organizations to adopt the technology, the surveillance for law enforcement agencies becomes more difficult. Voice over IP (VoIP) technology has also increased security concerns because Voice over IP (VoIP) and similar technologies have made it more difficult for the government to determine where a target is physically located when communications are being intercepted, and that creates a whole set of new legal challenges.
Quality of service
Communication on the IP network is perceived as less reliable in contrast to the circuit-switched public telephone network because it does not provide a network-based mechanism to ensure that data packets are not lost, and are delivered in sequential order. It is a best-effort network without fundamental quality of service (QoS) guarantees. Therefore, Voice over IP (VoIP) implementations may face problems with latency, packet loss, and jitter.
By default, network routers handle traffic on a first-come, first-served basis. Network routers on high volume traffic links may introduce latency that exceeds permissible thresholds for Voice over IP (VoIP). Fixed delays cannot be controlled, as they are caused by the physical distance the packets travel; however, latency can be minimized by marking voice packets as being delay-sensitive with methods such as DiffServ.
Voice over IP (VoIP) endpoints usually have to wait for completion of transmission of previous packets before new data may be sent. Although it is possible to preempt (abort) a less important packet in mid-transmission, this is not commonly done, especially on high-speed links where transmission times are short even for maximum-sized packets. An alternative to preemption on slower links, such as dialup and digital subscriber line (DSL), is to reduce the maximum transmission time by reducing the maximum transmission unit. But every packet must contain protocol headers, so this increases relative header overhead on every link traversed, not just the bottleneck (usually Internet access) link.
DSL modems provide Ethernet (or Ethernet over USB) connections to local equipment, but inside they are actually Asynchronous Transfer Mode (ATM) modems. They use ATM Adaptation Layer 5 (AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission, reassembling them back into Ethernet frames at the receiving end. A virtual circuit identifier (VCI) is part of the 5-byte header on every ATM cell, so the transmitter can multiplex the active virtual circuits (VCs) in any arbitrary order. Cells from the same VC are always sent sequentially.
However, a majority of DSL providers use only one VC for each customer, even those with bundled Voice over IP (VoIP) service. Every Ethernet frame must be completely transmitted before another can begin. If a second VC were established, given high priority and reserved for Voice over IP (VoIP), then a low priority data packet could be suspended in mid-transmission and a Voice over IP (VoIP) packet sent right away on the high priority VC. Then the link would pick up the low priority VC where it left off. Because ATM links are multiplexed on a cell-by-cell basis, a high priority packet would have to wait at most 53 byte times to begin transmission. There would be no need to reduce the interface MTU and accept the resulting increase in higher layer protocol overhead, and no need to abort a low priority packet and resend it later.
ATM has substantial header overhead: 5/53 = 9.4%, roughly twice the total header overhead of a 1500 byte Ethernet frame. This “ATM tax” is incurred by every DSL user whether or not they take advantage of multiple virtual circuits – and few can.
ATM’s potential for latency reduction is greatest on slow links, because worst-case latency decreases with increasing link speed. A full-size (1500 byte) Ethernet frame takes 94 ms to transmit at 128 kbit/s but only 8 ms at 1.5 Mbit/s. If this is the bottleneck link, this latency is probably small enough to ensure good Voice over IP (VoIP) performance without MTU reductions or multiple ATM VCs. The latest generations of DSL, VDSL and VDSL2, carry Ethernet without intermediate ATM/AAL5 layers, and they generally support IEEE 802.1p priority tagging so that Voice over IP (VoIP) can be queued ahead of less time-critical traffic.
Voice, and all other data, travels in packets over IP networks with fixed maximum capacity. This system may be more prone to congestion and DoS attacks 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. They are especially problematic when satellite circuits are involved because of the long distance to a geostationary satellite and back; delays of 400–600 ms are typical.
When the load on a link grows so quickly that its switches experience queue overflows, congestion results and data packets are lost. This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion. But Voice over IP (VoIP) usually uses UDP not TCP because recovering from congestion through retransmission usually entails too much latency. So QoS mechanisms can avoid the undesirable loss of Voice over IP (VoIP) packets by immediately transmitting them ahead of any queued bulk traffic on the same link, even when that bulk traffic queue is overflowing.
The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all. Jitter results from the rapid and random (i.e. unpredictable) changes in queue lengths along a given Internet path due to competition from other users for the same transmission links. Voice over IP (VoIP) receivers counter jitter by storing incoming packets briefly in a “de-jitter” or “playout” buffer, deliberately increasing latency to improve the chance that each packet will be on hand when it is time for the voice engine to play it. The added delay is thus a compromise between excessive latency and excessive dropout, i.e. momentary audio interruptions.
Although jitter is a random variable, it is the sum of several other random variables that are at least somewhat independent: the individual queuing delays of the routers along the Internet path in question. Thus according to the central limit theorem, we can model jitter as a Gaussian random variable. This suggests continually estimating the mean delay and its standard deviation and setting the playout delay so that only packets delayed more than several standard deviations above the mean will arrive too late to be useful. In practice, however, the variance in latency of many Internet paths is dominated by a small number (often one) of relatively slow and congested “bottleneck” links. Most Internet backbone links are now so fast (e.g. 10 Gbit/s) that their delays are dominated by the transmission medium (e.g. optical fiber) and the routers driving them do not have enough buffering for queuing delays to be significant.
It has been suggested to rely on the packetized nature of media in Voice over IP (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 quality of service (QoS) and quality of experience (QoE) for Voice over IP (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 Voice over IP (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 Voice over IP (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 Voice over IP (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.
Rural areas in particular are greatly hindered in their ability to choose a Voice over IP (VoIP) system over PBX. This is generally down to the poor access to superfast broadband in rural country areas. With the release of 4G data, there is a potential for corporate users based outside of populated areas to switch their internet connection to 4G data, which is comparatively as fast as a regular superfast broadband connection. This greatly enhances the overall quality and user experience of a Voice over IP (VoIP) system in these areas. This method was already trialled in rural Germany, surpassing all expectations.
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 Voice over IP (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 per second) Local area network (LAN) 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 Voice over IP (VoIP) call) which require QoS and which have negotiated a “contract” with the network controllers.