PDF | This paper was written for an Independent Study course. Princy Mehta Overview of Voice over IP Professor Udani February 1 Table. Voice over Internet Protocol (VoIP) is a form of communication that allows you to make phone calls over a broadband internet connection instead of typical. "What Voice over Internet Protocol (VoIP) is going to do is start to weaken VoIP aspires to be a telephony substitute, it will invite the threat of.
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This guide has been sponsored by. The Technology Guide Series goudzwaard.info . Voice over IP. (VoIP). The Leading Provider of Embedded Communications. Media, Inc. Packet Guide to Voice over IP, the image of a green . Voice over the Internet Protocol, a.k.a. Voice over IP, or VoIP, is a huge topic. You will be able to: To understand the current state of the art in Internet. Telephony. To be able to describe the key protocols involved in VoIP and their roles in.
Excessive load on a link can cause congestion and associated queueing delays and packet loss. This signals a transport protocol like TCP to reduce its transmission rate to alleviate the congestion. 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. But since every packet must contain protocol headers, this increases relative header overhead on every link traversed. The receiver must resequence IP packets that arrive out of order and recover gracefully when packets arrive too late or not at all.
Packet delay variation results from changes in queuing delay along a given network path due to competition from other users for the same transmission links.
VoIP receivers accommodate this variation by storing incoming packets briefly in a 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. Although jitter is a random variable, it is the sum of several other random variables which are at least somewhat independent: Motivated by the central limit theorem , jitter can be modeled 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, 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. 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 capillary routing at the packet level Fountain codes or particularly raptor codes it is recommended 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 VoIP calls. RFC 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. 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 VoIP system in these areas.
Non-ATM technologies such as 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.
Every Ethernet frame must be completely transmitted before another can begin. If a second VC were established, given high priority and reserved for VoIP, then a low priority data packet could be suspended in mid-transmission and a 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: ATM's potential for latency reduction is greatest on slow links, because worst-case latency decreases with increasing link speed.
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 VoIP work well even in congested scenarios.
Some examples include:. The quality of voice transmission is characterized by several metrics that may be monitored by network elements, by the user agent hardware or software. Such metrics include network packet loss , packet jitter , packet latency delay , post-dial delay, and echo. The metrics are determined by VoIP performance testing and monitoring. A VoIP media gateway controller aka Class 5 Softswitch works in cooperation with a media gateway aka IP Business Gateway and connects the digital media stream, so as to complete creating the path for voice as well as data media.
The Ethernet interfaces are also included in the modern systems, which are specially designed to link calls that are passed via the VoIP.
Most VoIP implementations support E. Often VoIP implementations employ methods of translating non-E. Echo can also be an issue for PSTN integration. Typically, it is the responsibility of the former carrier to "map" the old number to the undisclosed number assigned by the new carrier. This is achieved by maintaining a database of numbers. A dialed number is initially received by the original carrier and quickly rerouted to the new carrier. Multiple porting references must be maintained even if the subscriber returns to the original carrier.
The FCC mandates carrier compliance with these consumer-protection stipulations. A voice call originating in the VoIP environment also faces challenges to reach its destination if the number is routed to a mobile phone number on a traditional mobile carrier.
VoIP has been identified in the past as a Least Cost Routing LCR system, which is based on checking the destination of each telephone call as it is made, and then sending the call via the network that will cost the customer the least.
This rating is subject to some debate given the complexity of call routing created by number portability. With GSM number portability now in place, LCR providers can no longer rely on using the network root prefix to determine how to route a call. Instead, they must now determine the actual network of every number before routing the call.
In countries without a central database, like the UK, it might be necessary to query the GSM network about which home network a mobile phone number belongs to. As the popularity of VoIP increases in the enterprise markets because of least cost routing options, it needs to provide a certain level of reliability when handling calls.
MNP checks are important to assure that this quality of service is met. Handling MNP lookups before routing a call provides some assurance that the voice call will actually work. A telephone connected to a land line has a direct relationship between a telephone number and a physical location, which is maintained by the telephone company and available to emergency responders via the national emergency response service centers in form of emergency subscriber lists.
When an emergency call is received by a center the location is automatically determined from its databases and displayed on the operator console. In IP telephony, no such direct link between location and communications end point exists.
Even a provider having hardware infrastructure, such as a DSL provider, may know only the approximate location of the device, based on the IP address allocated to the network router and the known service address. IP communication provides for 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 being used for customer communications may belong to the enterprise, not being the IP address of the residential ISP.
On mobile devices, e. In such cases, the Internet telephony service provider ITSP knows only that a particular user's equipment is active. Service providers often provide emergency response services by agreement with the user who registers a physical location and agrees that emergency services are provided to that address only if an emergency number is called from the IP device. The VoIP E emergency-calling system associates a physical address with the calling party's telephone number.
All VoIP providers that provide access to the public switched telephone network are required to implement E,  a service for which the subscriber may be charged. The VoIP E system is based on a static table lookup. Unlike in cellular phones, where the location of an E call can be traced using assisted GPS or other methods, the VoIP E information is accurate only if subscribers, who have the legal responsibility, keep their emergency address information current.
Transmission of fax documents was problematic in early VoIP implementations, as most voice digitization and compression codecs are optimized for the representation of the human voice and the proper timing of the modem signals cannot be guaranteed in a packet-based, connection-less network.
A standards-based solution for reliably delivering fax-over-IP is the T. The T. The fax machine may be a standard device connected to an analog telephone adapter ATA , or it may be a software application or dedicated network device operating via an Ethernet interface.
UDP provides near real-time characteristics due to the "no recovery rule" when a UDP packet is lost or an error occurs during transmission. Some newer high end fax machines have built-in T. Two successive packets have to be lost to actually lose data integrity.
Telephones for traditional residential analog service are usually connected directly to telephone company phone lines which provide direct current to power most basic analog handsets independently of locally available electrical power.
IP Phones and VoIP telephone adapters connect to routers or cable modems which typically depend on the availability of mains electricity or locally generated power. Such battery-backed devices typically are designed for use with analog handsets. Some VoIP service providers implement services to route calls to other telephone services of the subscriber, such a cellular phone, in the event that the customer's network device is inaccessible to terminate the call.
The susceptibility of phone service to power failures is a common problem even with traditional analog service in areas where many customers download modern telephone units that operate with wireless handsets to a base station, or that have other modern phone features, such as built-in voicemail or phone book features.
The security concerns of VoIP telephone systems are similar to those of other Internet-connected devices.
This means that hackers with knowledge of VoIP vulnerabilities can perform denial-of-service attacks, harvest customer data, record conversations, and compromise voicemail messages. Compromised VoIP user account or session credentials may enable an attacker to incur substantial charges from third-party services, such as long-distance or international calling.
The technical details of many VoIP protocols create challenges in routing VoIP traffic through firewalls and network address translators , used to interconnect to transit networks or the Internet.
Private session border controllers are often employed to enable VoIP calls to and from protected networks. Though many consumer VoIP solutions do not support encryption of the signaling path or the media, securing a VoIP phone is conceptually easier to implement than on traditional telephone circuits.
A result of the lack of encryption is that it is relatively easy to eavesdrop on VoIP calls when access to the data network is possible. IPsec is available to secure point-to-point VoIP at the transport level by using opportunistic encryption. Voice over IP protocols and equipment provide caller ID support that is compatible with the facility provided in the public switched telephone network PSTN. Most analog telephone adapters do not decode dial pulses generated by rotary dial telephones, but rather support only touch-tone signaling, but pulse-to-tone converters are commercially available.
Some special telephony services, such as those that operate in conjunction with digital video recorders, satellite television receivers, alarm systems, conventional modems over PSTN lines, may be impaired when operated over VoIP services, because of incompatibilities in design. VoIP has drastically reduced the cost of communication by sharing network infrastructure between data and voice.
Secure calls using standardized protocols, such as Secure Real-time Transport Protocol , as most of the facilities of creating a secure telephone connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is necessary only to encrypt and authenticate the existing data stream.
Automated software, such as a virtual PBX , may eliminate the need of personnel to greet and switch incoming calls. Throughout the developing world, particularly in countries where regulation is weak or captured by the dominant operator, restrictions on the use of VoIP are often imposed, including in Panama where VoIP is taxed, Guyana where VoIP is prohibited and India where its retail commercial sales is allowed but only for long distance service.
The country has installed firewalls to prevent international calls being made using VoIP. These measures were taken after the popularity of VoIP reduced the income generated by the state owned telecommunication company. VoIP services operating in Canada are required to provide emergency service.
In the European Union , the treatment of VoIP service providers is a decision for each national telecommunications regulator, which must use competition law to define relevant national markets and then determine whether any service provider on those national markets has "significant market power" and so should be subject to certain obligations.
A general distinction is usually made between VoIP services that function over managed networks via broadband connections and VoIP services that function over unmanaged networks essentially, the Internet. The relevant EU Directive is not clearly drafted concerning obligations which can exist independently of market power e.
A review of the EU Directive is under way and should be complete by In Oman , it is illegal to provide or use unauthorized VoIP services, to the extent that web sites of unlicensed VoIP providers have been blocked. Violations may be punished with fines of 50, Omani Rial about , US dollars or spend two years in jail or both. In , police raided Internet cafes throughout the country and arrested people for using or providing VoIP services. In September , Saudi Arabia lifted the ban on VoIPs, in an attempt to reduce operational costs and spur digital entrepreneurship.
However, some VoIPs such as Skype were allowed.
Foreign based VoIP server services are illegal to use in India. Internet Telephony is considered to be different service in its scope, nature and kind from real time voice as offered by other Access Service Providers and Long Distance Carriers. Hence the following type of Internet Telephony are permitted in India: In South Korea , only providers registered with the government are authorized to offer VoIP services.
Unlike many VoIP providers, most of whom offer flat rates, Korean VoIP services are generally metered and charged at rates similar to terrestrial calling. Foreign VoIP providers encounter high barriers to government registration.
This issue came to a head in when Internet service providers providing personal Internet services by contract to United States Forces Korea members residing on USFK bases threatened to block off access to VoIP services used by USFK members as an economical way to keep in contact with their families in the United States, on the grounds that the service members' VoIP providers were not registered.
In the United States, the Federal Communications Commission requires all interconnected VoIP service providers to comply with requirements comparable to those for traditional telecommunications service providers. Providers of "nomadic" VoIP service—those who are unable to determine the location of their users—are exempt from state telecommunications regulation.
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 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 VoIP's low cost and flexibility convinces more and more organizations to adopt the technology, the surveillance 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 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.
The early developments of packet network designs by Paul Baran and other researchers were motivated by a desire for a higher degree of circuit redundancy and network availability in face of infrastructure failures than was possible in the circuit-switched networks in telecommunications in the mid-twentieth century. Following the termination of the ARPANET project, and expansion of the Internet for commercial traffic, IP telephony became an established area of interest in commercial labs of the major IT concerns, such Microsoft and Intel , and open-source software, such as VocalTec , became available by the mids.
By the late s, the first softswitches became available, and new protocols, such as H. In the early s, the proliferation of high-bandwidth always-on Internet connections to residential dwellings and businesses, spawned an industry of Internet telephony service providers ITSPs. The development of open-source telephony software, such as Asterisk PBX , fueled widespread interest and entrepreneurship in voice-over-IP services, applying new Internet technology paradigms, such as cloud services to telephony.
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