Voice over Internet Protocol (also voice over IP , VoIP or IP telephony ) is the methodology and technology group for delivery voice communications and multimedia sessions over the Internet Protocol (IP) network, such as the Internet. The terms Internet calls , broadband phones , and broadband phone services specifically refer to the provision of communication services (voice, fax, SMS, voice-messaging) through the public Internet, rather than through the public switched telephone network (PSTN).
The steps and principles involved in VoIP phone calls originate similar to traditional digital phones and involve signaling, channel settings, digitizing analog voice signals, and encoding. Instead of being transmitted over circuit-switched networks, digital information is packaged, and transmission occurs as IP packets through a packet-switched network. They carry media streams using a special media delivery protocol that encodes audio and video with audio codecs, and video codecs. Various codecs exist that optimize media flow based on application requirements and network bandwidth; some implementations rely on narrowband speech and compression, while others support high-fidelity stereo codecs. Some popular codecs include legal and legal versions of G.711, G.722, an open source sound codec known as iLBC, a codec that only uses 8 kbit/dt per path called G.729, and many others.
Initial providers of voice-over-IP services offer business models and technical solutions that reflect the old telephone network architecture. Second-generation providers, such as Skype, built a closed network for a private user base, offering the benefits of free calls and convenience while possibly charging for access to other communications networks, such as PSTN. This restricts users' freedom to mix-and-match third-party hardware and software. Third-generation providers, such as Google Talk, adopt the concept of the VoIP federation - which is a departure from the legacy network architecture. This solution usually allows dynamic interconnection between users on two domains on the Internet when the user wants to make a call.
In addition to VoIP phones, VoIP is also available on many personal computers and other Internet access devices. SMS calls and text messages can be sent via mobile data or Wi-Fi.
Video Voice over IP
Pronunciation
VoIP is variously pronounced as initialism, VOIP , or as an acronym, usually/'v ?? p/( voyp ), as in voice , but full-word pronunciation, voice over Internet Protocol , or voice over IP , sometimes used.
Maps Voice over IP
Protocol
Voice over IP has been implemented in various ways using proprietary protocols and protocols based on open standards. This protocol can be used by VoIP phones, special purpose software, mobile applications or integrated into web pages. VoIP protocols include:
- Session Initiation Protocol (SIP), the connection management protocol developed by IETF
- H.323, one of the first VoIP call signaling and control protocols that invented widespread implementation. Due to the development of newer and less complex protocols such as MGCP and SIP, the deployment of H.323 is increasingly limited to carrying existing long-distance network traffic.
- Media Gateway Control Protocol (MGCP), connection management for media gateways
- H.248, a control protocol for media gateways across converged Internet networks consisting of traditional switched telephone networks (PSTN) and modern packet networks
- Real-time Transport Protocol (RTP), transport protocol for real-time audio and video data
- The Real-Time Transport Control Protocol (RTCP), sister protocol for RTP provides flow statistics and status information
- Secure Real-time Transport Protocol (SRTP), RTP encrypted version
- Session Description Protocol (SDP), the file format used primarily by SIP to describe VoIP connections
- Inter-Asterisk eXchange (IAX), the protocol used between VoIP servers
- Expandable Messaging and Presence Protocols (XMPP), instant messaging, attendance information, and maintenance of contact lists
- Jingle, adding a peer-to-peer session control to XMPP
- Skype protocol, patented Internet telephony protocol protocol based on peer-to-peer architecture
Adoption
Consumer market
Mass market VoIP services use existing broadband Internet access, where customers place and receive phone calls in the same way as through public telephone (PSTN) networks. The full-service VoIP phone company provides incoming and outgoing services with direct incoming calls. Many offer unlimited domestic calls and sometimes international calls for a fixed monthly subscription fee. Phone calls between customers from the same provider are usually free when the fixed cost service is not available.
VoIP phones are required to connect to VoIP providers. This can be implemented in several ways:
- Special VoIP phones connect directly to IP networks using technologies such as Ethernet cable or Wi-Fi. These are usually designed in the style of traditional digital business phones.
- The analog phone adapter is connected to the network and implements electronics and firmware to operate a conventional analog phone installed through a modular telephone jack. Some Internet gateways and cablemodem housing have this function.
- A softphone software program installed on a network computer with a microphone and speakers, or a headset. This app typically presents a dial pad and display field to the user to operate the app with a mouse click or keyboard input.
PSTN and mobile network provider
It is increasingly common for telecommunications providers to use VoIP phones over dedicated and public IP networks as backhaul to connect switching centers and to interconnect with other telephone network providers; this is often referred to as IP backhaul .
The smartphone may have a SIP client built into the firmware or available as an app download.
Corporate usage
Due to the bandwidth and low cost efficiency that VoIP technology can deliver, businesses are migrating from traditional copper wire telephone systems to VoIP systems to reduce their monthly call charges. In 2008, 80% of all newly installed International Private Branch (PBX) exchange lines are VoIP.
Business VoIP solutions have evolved into integrated communications services that handle all communications - phone calls, faxes, voice mail, emails, Web conferences, and more - as discrete units that can all be delivered via any means and to the handset, including phone. Two types of competitors compete in this space: a set is focused on VoIP for medium to large companies, while others target small and medium enterprise (SME) markets.
VoIP enables voice and data communications to run over a single network, which can significantly reduce infrastructure costs.
The price of extensions on VoIP is lower than for PBX and lock systems. The VoIP switch can run on commodity hardware, such as a personal computer. Instead of closed architecture, this device relies on a standard interface.
The VoIP device has a simple and intuitive user interface, so users can often make simple system configuration changes. Dual-mode phones allow users to continue their conversation as they move between outside mobile service and internal Wi-Fi network, thus no longer need to carry desktop and mobile phones. Treatment becomes simpler because there are fewer devices to monitor.
Skype, which initially markets itself as a service among friends, has started serving the business, providing free connections between every user on the Skype network and connecting to and from regular PSTN phones for a fee.
In the United States, Social Security Administration (SSA) transforms its field offices into 63,000 workers from traditional phone installations to VoIP infrastructure through existing data networks.
Service quality
Communication on an IP network is considered less reliable than a circuit-switched telephone network because it does not provide network-based mechanisms to ensure data packets are not lost, and are sent in sequence. This is the best network of unwarranted Quality of Service (QoS) efforts. Sound, and all other data, runs in packets over an IP network with a fixed maximum capacity. These systems may be more susceptible to jamming and DoS attacks than traditional circuit switched systems; Inadequate circuit switched capacity systems will refuse new connections while carrying the rest uninterrupted, while real-time data quality such as telephone conversations on packet-switched networks drops dramatically. Therefore, VoIP implementations may face problems with latency, packet loss, and jitter.
By default, network routers handle traffic based on the first coming, first served. Increased delay can not be controlled because it is caused by physical distance of travel package. They are very problematic when satellite circuitry is involved because of long distances to the geostationary and return satellites; 400-600 ms delay is typical. Latency can be minimized by marking voice packets as delay-sensitive with QoS methods like DiffServ.
Network routers on high volume traffic can introduce latency that exceeds the allowed threshold for VoIP. When the load on the link grows so fast that the switch is experiencing excessive queuing, the result of congestion and data packets is lost. This signifies a transport protocol such as TCP to reduce its transmission rate to reduce congestion. But VoIP typically uses UDP instead of TCP because recovery from congestion through retransmissions usually requires too much latency. So QoS mechanisms can avoid the loss of unwanted VoIP packets by sending them immediately before mass traffic is queued at the same link, even when the bulk traffic queue overflows.
VoIP endpoints usually have to wait for packet transmission to be completed before new data can be sent. Although it is possible to precede (cancel) less important packets on the middle transmission, this is not common, especially at high speed connections 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 maximum transmission time by reducing the maximum transmission unit. But each packet must contain a protocol header, so this increases the relative header overhead on each link traversed, not just a bottleneck link (usually Internet access).
The recipient must recover IP packets that are suddenly damaged and recovered gracefully when the packet arrives late or not at all. Jitter results from fast and random changes (ie unpredictable) in the queue length along the given Internet path due to competition from other users for the same transmission connection. The VoIP receiver overcomes jitter by storing packets that enter briefly in "de-jitter" or "playout" buffers, deliberately increasing the latency to increase the likelihood that each packet will be in hand when it is time for the sound engine to play it. The added delay is a compromise between excessive latency and excessive collapse, which is a momentary audio interruption.
Although jitter is a random variable, it is the sum of some other random variables that are at least somewhat independent: the individual queue delay from the router along the Internet path is concerned. According to the central limit theorem, jitter can be modeled as a gaussian random variable. This indicates a continuous estimate of the average delay and standard deviation and the delay delay setting so that only delayed packets more than some above-average deviation standard will arrive late to be useful. In practice, the variance in the latency of many Internet paths is dominated by a relatively small number of (often one) "bottleneck" links. Most internet backbones are now so fast (eg 10 Gbit/s) whose delays are dominated by transmission mediums (eg optical fibers) and routers that encourage them not to have enough buffering because queuing delays are significant.
It has been suggested to depend on the packetized nature of the media in VoIP communications and to send packet flow from the source phone to the destination phone simultaneously across different routes (multi-path routing). In such a way, temporary failure has less impact on the quality of communication. In capillary routing at packet level, Fountain code or especially raptor code, it is recommended to transmit additional redundant packages that make communication more reliable.
A number of protocols have been established to support quality of service reporting (QoS) and quality of experience (QoE) for VoIP calls. These include RTCP Extended Report (RFC 3611), RTCP SIP Summary Report, H.460.9 Appendix B (for H.323), H.248.30 and MGCP extensions. Block RFC 3611 VoIP Metrics is generated by IP phones or gateways during direct calls and contains information about packet loss, packet unloading rate (due to jitter), packet loss/burst length/density, network delay , late system delay, signal level/echo, Mean Opinion Scores (MOS) and R factors and configuration information associated with the jitter buffer.
The VoIP RFC 3611 metrics report is exchanged between occasional IP endpoints during a call, and a final calling message sent via the RTCP SIP Summary Report or one of the other signaling protocol extensions. The RFC 3611 VoIP metrics report is intended to support real-time feedback regarding QoS issues, exchange of information between end points for improved call quality calculations and various other applications.
The rural areas in particular are severely hampered in their ability to choose VoIP systems through PBX. This generally leads to poor access to super-fast broadband in rural areas. With the release of 4G data, there is potential for corporate users based outside the resident area to switch their internet connection to 4G data, which is relatively as fast as a regular superfast broadband connection. This greatly improves the quality and overall user experience of VoIP systems in this area.
DSL and ATM
The DSL modem provides an Ethernet (or Ethernet over USB) connection to local equipment, but it is actually an Asynchronous Transfer Mode (ATM) modem (Note: Non-ATM technologies like 802.3ah also provide this capability). They use ATM Adaptation Layer 5 (AAL5) to segment each Ethernet packet into a series of 53-byte ATM cells for transmission, rearranging them back to the Ethernet frame at the receiving end. The virtual circuit identifier (VCI) is part of a 5-byte header on each ATM cell, so the transmitter can multiply active virtual circuits (VCs) in an orderly manner. Cells from the same VC are always sent in sequence.
The majority of DSL providers use only one VC for each customer, even those with bundled VoIP services. Each Ethernet frame must be fully transmitted before anything else can be started. If a second VC is established, given a high priority and is reserved for VoIP, then low priority data packets may be delayed in mid-transmission and VoIP packets sent immediately on high priority VCs. Then the link will take the low-priority VC it ​​left behind. Since ATM links are multiplied cell-by-cell, high priority packets must wait at most 53 bytes to initiate transmission. There will be no need to reduce the MTU interface and receive the enhancements generated on the higher layer protocol overhead, and no need to undo the low priority packets and then send them back.
The ATM has a substantial header overhead: 5/53 = 9.4%, roughly doubling the total overhead header of the 1500 byte Ethernet frame. These "ATM taxes" are issued by every DSL user whether they take advantage of some virtual circuits and some can.
The ATM potential for the highest latency reduction in slow links, since the worst case latency decreases with increasing link speed. The full-size Ethernet framework (1500 bytes) requires 94Ã, ms to transmit at 128Ã, kbit/s but only 8Ã, ms at 1.5Ã, Mbit/s. If this is a bottleneck link, this latency may be small enough to ensure good VoIP performance without a reduced MTU or some VC ATMs. The latest generation of DSL, VDSL and VDSL2, carry Ethernet without ATM/AAL5 intermediate layers, and they generally support IEEE 802.1p priority tagging so VoIP can be queued before less time-critical traffic.
Layer 2
A number of protocols related to the data link layer and physical layer include a service quality mechanism that can be used to ensure that applications such as VoIP work well even in crowded scenarios. Some examples include:
- IEEE 802.11e is an amendment approved for the IEEE 802.11 standard that defines a series of service quality improvements for wireless LAN applications through modifications to the Media Access Control (MAC) layer. Standards are considered very important for time sensitive applications, such as voice over wireless IP.
- IEEE 802.1p defines 8 different service classes (including those dedicated to voice) for traffic on Ethernet cable layer-2.
- ITU-T G.hn Standard, which provides a way to create high speeds (up to 1 gigabit per second) Local area network (LAN) using existing home wiring (power lines, telephone lines and coaxial cables). G.hn provides QoS with "Flow-Free Option Transmission Opportunities" (CFTXOPs) allocated for flow (such as VoIP calls) that require QoS and who have negotiated "contracts" with network controllers.
Performance metrics
The quality of voice transmission is characterized by several metrics that can be monitored by network elements, by hardware or user agent software. These metrics include loss of network packets, packet jitter, packet latency (delays), post-call delays, and echoes. Metrics are determined by VoIP performance testing and monitoring.
PSTN Integration
The VoIP media gateway controller (aka Class 5 Softswitch) works with the media gateways (aka IP Business Gateway) and connects the stream of digital media, thus completing the creation of pathways for voice and data media. They include interfaces to connect standard PSTN networks with ATM and Inter Protocol networks. The Ethernet interface is also included in modern systems, designed specifically for connecting calls passed through VoIP.
E.164 is a global FGFnumbering standard for PSTN and PLMN. Most VoIP implementations support E.164 to allow calls to be transferred to and from VoIP and PSTN/PLMN subscribers. VoIP implementation can also allow other identification techniques to be used. For example, Skype allows customers to choose "Skype name" (username) whereas SIP implementation can use URIs that are similar to email addresses. Often VoIP implementations use non-E.164 identifier translation methods to E.164 and vice versa, such as Skype-In service provided by Skype and ENUM services in IMS and SIP.
Echo can also be a problem for PSTN integration. Common causes of echoes include impedance mismatches in analog circuits and acoustic coupling from sending and receiving signals at the receiving end.
Portability number
Local number portability (LNP) and mobile number portability (MNP) also impact VoIP business. In November 2007, the Federal Communications Commission in the United States released an order to extend the amount of portability liability to interconnected VoIP providers and carriers that support VoIP service providers. Portability number is a service that allows customers to choose a new phone operator without requiring a new number to be issued. Normally, it is the responsibility of the former operator to "map" the old number to an undisclosed number set by the new operator. This is achieved by maintaining a database of numbers. The number dialed was initially received by the original operator and quickly redirected to the new operator. Some porting references must be maintained even if the customer returns to the original operator. The FCC mandates operator compliance with these consumer protection provisions.
Voicemail calls from VoIP environments also face the challenge of achieving the goal if the number is diverted to a mobile phone number on a traditional mobile carrier. VoIP has been identified in the past as the Least Cost Routing (LCR) system, which is based on checking the purpose of each phone call as it is made, and then sending calls over the network that will harm the customer. This ranking is subject to some debate given the complexity of routing calls made by portability numbers. With GSM number portability now in place, LCR providers can no longer rely on the use of network root prefixes to determine how to route calls. Instead, they now have to specify the actual network of each number before routing the call.
Therefore, VoIP solutions also need to handle MNP when routing voice calls. In countries without a central database, such as the UK, it may be necessary to ask the GSM network about which home network is the mobile number. Due to VoIP's increasing popularity in the enterprise market due to the lowest cost routing option, it is necessary to provide a certain level of reliability when handling calls.
MNP checks are important to ensure that the quality of these services are met. Handling MNP searches before route routes provides a guarantee that voice calls will actually work.
Emergency call
Phones connected to telephone lines have a direct connection between phone numbers and physical locations, which are managed by the telephone company and available to emergency responders through the national emergency response service center in the form of an emergency customer list. When an emergency call is received by the center, the location is automatically determined from its data base and displayed in the console of the operator.
In IP telephony, there is no direct connection between the location and the end point of communication. Even providers with a hardware infrastructure, such as a DSL provider, may only know the approximate location of the device, based on the IP address allocated to the network router and the known service address. Some ISPs do not track automatic assignment of IP addresses to customer equipment.
IP communication provides device mobility. For example, a residential broadband connection may be used as a link to a virtual private network of a corporate entity, in which case the IP address used for customer communications may belong to the company, not the residential IP address of the ISP. Extensions outside these locations may appear as part of an upstream IP PBX. On a mobile device, for example, a 3G handset or a USB wireless broadband adapter, the IP address has no connection to any physical location known to the telephone service provider, as mobile users may be anywhere in the region with network coverage, even roaming via other devices. mobile companies.
At the VoIP level, the phone or gateway can identify itself by listing Session Initiation Protocol (SIP) by its account credentials. In such cases, Internet telephony service providers (ITSP) know only that certain user equipment is active. Service providers often provide emergency response services based on agreements with users who register physical locations and agree that emergency services are only provided to the address if the emergency number is called from the IP device.
Such emergency services are provided by VoIP vendors in the United States by a system called Enhanced 911 (E911), under the Wireless Communications and Public Safety Act 1999. The E911 VoIP emergency call system links physical addresses with the calling party telephone. amount. All VoIP service providers that provide access to the publicly diverted telephone network are required to implement E911, a service where customers may be charged. "VoIP providers may not allow customers to" opt out "of 911 services."
The E911 VoIP system is based on static table search. Unlike on cell phones, where the location of E911 calls can be tracked using GPS assistance or other methods, E911 VoIP information is accurate only if customers, who have legal liability, maintain their current emergency address information.
Fax Support
Sending faxes over a VoIP network is sometimes referred to as Fax over IP (FoIP). The transmission of fax documents is problematic in early VoIP implementations, as most voice digitization and compression codecs are optimized for human voice representation and the exact timing of modem signals can not be guaranteed in networks without packet-based connections. The standard-based solution for reliable fax-over-IP delivery is the T.38 protocol.
The T.38 protocol is designed to compensate for differences between traditional packet-less communications over analog channels and packet-based transmissions that are the basis for IP communications. The fax machine can be a standard device connected to an analog phone adapter (ATA), or perhaps a special software application or network device operating through an Ethernet interface. Initially, T.38 was designed to use UDP or TCP transmission methods across IP networks. UDP provides near real-time characteristics because "no recovery rules" when the UDP packet is lost or an error occurs during transmission.
Some of the newer high-end fax machines have built-in T.38 capabilities that connect directly to a network switch or router. In T.38 each packet contains a portion of the data stream sent in the previous packet. Two consecutive packets must be lost to completely lose data integrity.
Power requirements
Phones for traditional residential analog services usually connect directly to telephone company telephone lines that provide direct current to power the most basic analog handsets independently of locally available power.
IP phones and VoIP phone adapters connect to a router or cable modem that usually depends on the availability of primary power or locally generated power. Some VoIP providers use equipment where customers (eg cablemodems) with battery powered support to ensure uninterrupted service for several hours in case of local power failure. Such battery-backed devices are usually designed for use with analog handsets.
Some VoIP providers implement services to route calls to other phone services from customers, such as cell phones, if the customer's network device is inaccessible to end a call.
The vulnerability of telephone services to power failures is a common problem even with traditional analog services in areas where many customers purchase a modern telephone unit that operates with a wireless handset to a base station, or that has other modern phone features, such as a built-in mailbox or phonebook feature.
Security
The security issue of VoIP phone systems is similar to other devices connected to the Internet. This means that hackers with knowledge of VoIP vulnerabilities can perform denial-of-service attacks, harvest customer data, record conversations, and compromise voicemail messages. A compromised VoIP user account or session credential may allow an attacker to incur substantial fees from third-party services, such as long distance or international calls.
The technical details of many VoIP protocols create challenges in driving VoIP traffic through firewall and network address translators, which are used for interconnection to a transit network or the Internet. Personal session border controllers are often used to enable VoIP calls to and from protected networks. Other methods of traversing NAT devices include auxiliary protocols such as STUN and Interactive Connectivity Establishment (ICE).
Although many consumer VoIP solutions do not support signal or media path encryption, securing a VoIP phone is conceptually easier to apply than on traditional telephone circuits. Due to lack of encryption is relatively easy eavesdropping VoIP calls when access to data networks is possible. Free open-source solutions, such as Wireshark, facilitate the retrieval of VoIP conversations.
Standards for securing VoIP are available in the Real-Time Secure Transport Protocol (SRTP) and the ZRTP protocol for analog phone adapters, as well as for some softphones. IPsec is available to secure point-to-point VoIP at the transport level by using opportunistic encryption.
Government and military organizations use various security measures to protect VoIP traffic, such as secure voice over IP (VoSIP), secure voice over IP (SVoIP), and secure voice over secure IP (SVoSIP). The difference lies in whether the encryption is applied at the end point of the phone or on the network. Secure voice over secure IP can be implemented by encrypting media with protocols such as SRTP and ZRTP. Secure voice over IP uses Type 1 encryption on a secret network, such as SIPRNet. Public Secure VoIP is also available with free GNU software and in many popular commercial VoIP programs through libraries, such as ZRTP.
Caller ID
The Voice over IP protocol and equipment provide caller ID support compatible with the facilities provided on the public switched telephone network (PSTN). Many VoIP providers also allow callers to configure arbitrary caller ID information.
Compatibility with traditional analog phone sets
Most analog phone adapters do not decode the call pulses generated by the calling phone, but only support tone-touch signaling, but pulse-to-tone converters are commercially available.
Support for other telephony devices
Some specialized telephone services, such as those that operate in conjunction with digital video recorders, satellite television receivers, alarm systems, conventional modems via PSTN lines, may be disrupted when operated through VoIP services, due to mismatch in design.
Operational costs
VoIP drastically reduces communication costs by sharing network infrastructure between data and voice. A broad-band connection has the ability to send more than one phone call. Secured calls use standard protocols, such as the Secure Real-time Transport Protocol, since most facilities for establishing secure phone lines through traditional phone lines, such as digitizing and digital transmission, already exist with VoIP. You just need to encrypt and authenticate the existing data stream. Automated software, such as virtual PBX, can eliminate the need for personnel to welcome and divert incoming calls.
Legal and regulatory issues
As VoIP's popularity grew, the government became more interested in organizing VoIP in a manner similar to PSTN services.
Across developing countries, countries where regulation is weak or captured by dominant operators, restrictions on the use of VoIP apply, including in Panama where VoIP is taxed, Guyana where VoIP is banned and India where its retail commercial sales are allowed but only for service distance far. In Ethiopia, where the government nationalizes telecommunication services, it is a criminal offense to offer services using VoIP. This country has installed a firewall to prevent international calls from using VoIP. These steps are taken after the popularity of VoIP reduces revenue generated by state-owned telecommunications companies.
Canada
In Canada, the Canadian Radio-television and Telecommunications Commission regulates telephone services, including VoIP phone services. VoIP services operating in Canada must provide 9-1-1 emergency services.
European Union
In the EU, the treatment of VoIP providers is a decision for every national telecommunications regulator, which must use competition law to determine the relevant national market and then determine whether any service provider in the national market has "significant market power" and so on should be subject to certain obligations). Common differences are usually made between VoIP services that function through managed networks (via broadband connections) and VoIP services that function through unmanaged networks (basically, the Internet).
The relevant EU directives are not explicitly set out on obligations that can exist independently of market forces (eg, the obligation to offer access to emergency calls), and it is not possible to say for certain whether the VoIP service providers of both types are bound by it. A review of the EU Guidelines is in progress and should be completed by 2007.
Arabic countries in GCC
Oman
In Oman, it is illegal to provide or use unauthorized VoIP services, as long as unlicensed VoIP provider web sites have been blocked. Violations can be punishable by a fine of 50,000 Rial Oman (about 130,317 US dollars) or spend two years in jail or both. In 2009, police raided 121 internet cafes across the country and arrested 212 people for using or providing VoIP services.
Saudi Arabia
In September 2017, Saudi Arabia lifted a ban on VoIP, in an effort to reduce operational costs and spur digital entrepreneurship.
United Arab Emirates
In the United Arab Emirates (UAE), it is illegal to provide or use unauthorized VoIP services, as long as unlicensed VoIP provider web sites have been blocked. However, some VoIP like Skype is allowed. In January 2018, UAE internet service providers blocked all VoIP apps, including Skype, but only allowed 2 "government-approved" VoIP (C'ME and BOTIM) VoIP apps with a flat rate of Dh52.50 per month for use on mobile devices. , and Dh105 a month to use more than one connected computer. "In the opposition, a petition on Change.org collects more than 5000 signatures, in response to blocked websites in the UAE.
India
In India, it is legal to use VoIP, but it is illegal to have a VoIP gateway inside India. This effectively means that people who own a PC can use it to make VoIP calls to any number, but if the far side is a regular phone, the gateway that converts VoIP calls to POTS calls is not permitted by law to reside within India. Foreign based VoIP server service is illegal for use in India.
For the benefit of International Remote Access and Service Providers, Internet telephony is permitted to ISPs with restrictions. Internet Telephony is considered as a distinct service in scope, nature and type of real-time voice as offered by other Access Service Providers and Remote Operators. Therefore the following types of Internet Telephony are allowed in India:
- (a) PC to PC; inside or outside India
(b) PC/device/Adapter compatible with international institutional standards such as ITU or IETF etc. in India to PSTN/PLMN abroad.
(C) Each device/Adapter conforms to International institutional standards such as ITU, IETF etc. connect to an ISP node with a static IP address to a similar device/Adapter; inside or outside India (d) Except for anything described under (ii) above, no other form of Internet Telephony is permitted (e) In India, there is no Separate Numbering Scheme provided for Internet Telephony. Currently 10 digit Numbering allocations based on E.164 are allowed for fixed telephone service, GSM, CDMA wireless. For Internet Telephony, the numbering scheme will only conform to the IP Addressing Scheme of the Internet Assignment Authority (IANA). Translation number E.164/private number to the IP address assigned to any device and vice versa, by the ISP to indicate compliance with the IANA numbering scheme is not permitted.
(f) The Internet Service Licensee is not permitted to have PSTN/PLMN connectivity. Voice communications to and from phones connected to PSTN/PLMN and following E.164 numbering are prohibited in India.
South Korea
In South Korea, only registered government providers are authorized to offer VoIP services. Unlike many VoIP providers, most of which offer fixed rates, Korean VoIP services are generally measured and charged the same rate as terrestrial calls. Foreign VoIP providers face high barriers to government registrations. This problem came to mind in 2006 when Internet service providers provide private Internet services with contracts to the United States Army. Korean members living on USFK bases threatened to block access to VoIP services used by USFK members as an economical way to keep in touch with them. family in the United States, on the grounds that the member VoIP service provider is not listed. A compromise was reached between USFK and Korean telecommunications officials in January 2007, in which USFK service members arrived in Korea before June 1, 2007, and subscribed to the ISP services provided at the base could continue to use US-based VoIP subscriptions, but then arrival had to use providers Korean-based VoIP, which by contract will offer a price similar to the fixed rate offered by US VoIP providers.
United States
In the United States, the Federal Communications Commission requires all interconnected VoIP service providers to meet requirements comparable to those of traditional telecommunications service providers. VoIP carriers in the US are required to support portability of local numbers; making services accessible to persons with disabilities; pay regulatory fees, universal service contributions, and other mandate payments; and allows law enforcement authorities to exercise oversight in accordance with the Communications Aid for the Law Enforcement Act (CALEA).
VoIP "Interconnect" operators (fully connected to PSTN) are mandated to provide Enhanced 911 services without special request, provide customer location updates, clearly expose any restrictions on their E-911 functionality to their customers, receive affirmative acknowledgment of this disclosure of all consumers, and 'may not allow their customers to "opt out" of 911 services.' VoIP carriers also benefit from certain US telecommunications regulations, including the right to interconnect and exchange traffic with incumbent local exchange operators through wholesale operators. VoIP service providers "nomadic" - those who can not determine the location of their users - are exempt from state telecommunication regulations.
Another legal issue is that the US Congress is debating fears of change in the Foreign Intelligence Surveillance Act. The questionable issue is the call between Americans and foreigners. The National Security Agency (NSA) is not authorized to tap into an American conversation without a warrant - but the Internet, and especially VoIP does not draw a clear line to the caller's location or the call recipient because the traditional phone system does not. Because the low cost and flexibility of VoIP convince more and more organizations to adopt technology, oversight for law enforcement agencies becomes more difficult. VoIP technology has also increased security concerns because VoIP and similar technologies have made it more difficult for governments to determine where targets are physically located when communication is being intercepted, and it creates a series of new legal challenges.
History
The early development of packet network design by Paul Baran and other researchers was motivated by the desire for higher levels of redundancy and network availability in the face of infrastructure failures than was possible in circuit-switched networks in telecommunications in the mid-twentieth century. In 1973, Danny Cohen first showed the packet voice form as part of the flight simulator application, which operates early in the ARPANET. Within the next span of about two decades, various forms of phone packages are developed and industry interest groups are formed to support new technologies. After the discontinuation of the ARPANET project, and the expansion of the Internet for commercial traffic, IP telephony became an established interest area in commercial laboratories of major IT issues, such as Microsoft and Intel, and open-source software, such as VocalTec, became available in the mid- an. In the late 1990s, the first softswitch became available, and new protocols, such as H.323, Media Gateway Control Protocol (MGCP) and Session Initiation Protocol (SIP) received widespread attention. In the early 2000s, the proliferation of high-speed, ever-active internet connections to residential and residential businesses, gave birth to the industry of Internet telephony service providers (ITSPs). The development of open source phone software, such as Asterisk PBX, sparked widespread interest and entrepreneurship in voice-over-IP services, implementing new Internet technology paradigms, such as cloud services to telephony.
Milestones
- 1973: Packet sound app by Danny Cohen
- 1974: The Institute of Electrical and Electronic Engineers (IEEE) publishes a paper entitled "Protocols for Network Interconnection Packages".
- 1974: Network Voice Protocol (NVP) tested via ARPANET in August 1974, carrying 16k CVSD voice coding. 1977: Danny Cohen and Jon Postel of the USC Information Science Institute, and Vint Cerf of the Agency for Advanced Defense Research Project (DARPA), agree to separate IP from TCP, and create UDP to bring real-time traffic.
- 1981: IPv4 is described in RFC 791.
- 1985: National Science Foundation makes NSFNET commission.
- 1986: Proposals from standard organizations for Voice over ATM, in addition to commercial voice packages from companies like StrataCom
- 1991: The first Voice-over-IP app, Speak Freely, is released to the public domain. Originally written by John Walker and further developed by Brian C. Wiles.
- 1992: Frame Relay Forum does standard development for Voice over Frame Relay.
- 1994: MTALK, a freeware VoIP app for Linux
- 1995: VocalTec released the first commercial Internet telephony software.
- Beginning in 1995, Intel, Microsoft, and Radvision embarked on standardization activities for VoIP communications systems.
- 1996:
- ITU-T initiated the development of standards for transmitting and signaling voice communications over an Internet Protocol network with the standard H.323.
- US telecommunications companies petition US Congress to ban Internet phone technology.
- 1997: Level 3 begins the first softswitch development, a term they invented in 1998.
- 1999:
- The Session Initiation Protocol (SIP) specification RFC 2543 is released.
- Mark Spencer from Digium developed the first open source private branch exchange (PBX) software (Asterisk).
- 2004: Commercial VoIP providers proliferate.
- 2007: Manufacturers and sellers of VoIP devices in Asia, especially in the Philippines where many families of foreign workers are located.
- 2011: Increase in WebRTC technology that enables VoIP directly in the browser
See also
References
External links
- The dictionary definition of VoIP in Wiktionary
- Internet phone travel guides from Wikivoyage
Source of the article : Wikipedia
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