Voice over IP

[The#Danger]

INTRODUCTION

In this paper outlined the work of research on the technology Voice over IP, which combines two historically separate worlds: the transmission of voice and data. It is carrying voice, data previously converted, between distant points. This would allow use network data to make phone calls, and develop a single network that is responsible for issuing all types of communication, whether voice or data. It is clear that having a network instead of two, is beneficial for any operator to offer both services. The strong growth and implementation of network IP in both local and remote, the development of technical advanced voice digitization, mechanisms of control and traffic prioritization, protocol transmission time real, and the study of new standards enable quality of service in IP networks, have created an environment where it is possible to transmit telephony over IP so no way signify the disappearance of telephone networks circuit mode, but should, at least temporarily, a phase coexistence between. This monograph is comprised of four chapters, which exhibits information relevant to Voice Over IP technology.

As technology, Voice over IP ( VoIP ) has spent several years in the market. However, it was not until the emergence of innovative new services based on this technology as the integration of voice and data has become a reality, which, for businesses, has meant a saving of costs and a communications more efficient and effective. It is expected that in 2005 the VoIP market will move more than 10,000 million dollars, especially given that perceptions of the corporate world are more favorable.

Definition

The products of telephony Internet is called: IP Telephony (IP telephony) Voice Over Internet, Voice over the Internet (VOI) - or Voice over IP, Voice over IP (VOIP).

Voice over IP (VoIP, Voice over IP) is a technology that allows transmission of voice over IP networks as data packets. IP Telephony is an immediate application of this technology in a manner that allows the use of ordinary telephone calls over IP networks or other packet networks using a PC, gateways and standard telephones. In general, communication services - voice, fax, voice messaging applications - that are transported via IP networks, Internet normally, instead of being transported via the telephone network. The VoIP (Voice over IP) this acronym refers to the technology used to send voice information in digital form in discrete packets over Internet protocol (IP stands for Protocol Internet), rather than through the telephone network usual. Before proceeding, you may want to clarify what is a connection protocol. A connection protocol is a set of rules, a "language in common" that both parties agree to use to be able to communicate, is like saying: Now we will communicate in English, and we agree that this is English, or it is a convention.

The industry of Voice over IP is in a stage of rapid growth. The evolution of the use of Voice over IP will come with the development of infrastructure and communication protocols. In 2010, a quarter of global calls will be based on IP. Over time, the voice and data applications have required different networks using different technologies. However, lately there have been numerous efforts to find a solution that provides a satisfactory support for both types of transmission over a single network. Voice over IP telephony is a technology that can be enabled through a data network packet switching, protocol via IP (Internet Protocol). The real advantage of this technology is the transmission of free speech as it travels as data.

VoIP technology can revolutionize internal communications by offering:

• Access to corporate networks from small offices through integrated networks of voice and data attached to branches.
• Corporate directories based on the Intranet with messaging services and personal numbers for those who must travel.
• Directory Services-based conferencing and graphics from the system desktop.
• Virtual private networks and gateways managed to replace voice Virtual Private Networks ( VPN ).

[The#Danger]

VoIP (Voice over IP) provides new opportunities for those able to anticipate and act quickly enough to overcome the confusion that surrounds this remarkable technology as used Voice over IP. It is important to know how to use the VoIP (Voice over IP), you basically buy a device that visually is a black box that connects one side to the telephone equipment and the other to the PC ( computer ), but also IP phones are available. Of course you need to install a software for that device works. This device is almost always sold in the same stores that sell computers. There are two connection options:

• One party has VoIP (Voice over IP) and the other not.
• Both parties are VoIP (Voice over IP)

If both parties are VoIP (Voice over IP) call is free, it is called VoIP (Voice over IP) VoIP (Voice over IP), only have to dial the phone number and nothing else. If the caller only has VoIP (Voice over IP), then uses a card that is purchased online (online). The above card is a card of plastic or cardboard such as those sold in stores, rather it is a virtual card you buy and load on the Internet.

The model of Voice over IP consists of three main elements:

• The client - This element establishes and terminates voice calls. Encodes, packages and transmits the output information generated by the user's microphone. It also receives, decodes and reproduces the voice information input by the user's speakers or headphones. Note that the client component is available in two basic forms: the first is a suite of software running on a PC that you control through a graphical user interface (GUI) and the second client may be a "virtual" who resides in the gateway.
• Servers - The second element of Voice over IP is based on servers, which handle a wide range of operations, complex databases, in real time and outside it. These operations include user authentication, appraisal, accounting , pricing, collection, distribution of profits, routing, management 's overall service, charging customers, service control, registration of users and directory services among others.
• Gateways - The third element is formed by the Voice over IP gateways, which provide a communication bridge between users. The role of a main gateway is to provide the interfaces with appropriate traditional telephony, serving as a platform for virtual clients.

These teams also play an important role in access security, accounting, control of quality of service (QoS, Quality of Service) and improve it.


Characteristics of Voice over IP

For its structure provides the following standard features:

• Allows control of network traffic, so less chance of the occurrence of significant declines in the performance of data networks.
• Provides links to the traditional telephone network.
• Being supported on IP technology has the following advantages:
  
  • Is independent of the type of network physics that supports it. Allows integration with existing large IP networks.
  • Is independent of the hardware used.
  • Lets be implemented in both software and hardware, with the particularity that the hardware would eliminate the initial impact for the common user.

Voice over IP Protocols

Today, there are two protocols for transmitting voice over IP, both define the way in which such devices must communicate with each other, and includes specifications for codecs (coder-decoder) audio to convert an audio signal to a digitized pad and vice versa.

H.323

H.323 is the standard established by the International Telecommunications Union (ITU) which is comprised of a highly complex and extensive protocol, which also include voice over IP, provides specifications for video-conferencing and real-time applications, including other variants.

Session Initiation Protocol (SIP)

Session Initiation Protocol (SIP) was developed by the IETF (Internet Engineering Task Force) specifically for IP telephony, which in turn takes advantage of other existing protocols to handle part of the process of conversion, a situation that applies to H.323 as it defines its own protocols bases.

The Standard Voice over IP

Long, those responsible for communications companies have in mind the possibility of using their data infrastructure for the transport of internal voice traffic the company . However, the emergence of new standards and the improvement and cheapening of voice compression technology, which is ultimately causing its implementation.

Having found that from a PC with elements multimedia , you can make phone calls over the Internet, we believe that IP telephony is little more than a toy, because the quality of voice that we obtain through Internet is very poor . However, if in our company have a data network that has a large enough bandwidth, we can also think of using that network for voice traffic between the various delegations of the company. The advantages we would get to use our network to transmit both voice and data are evident:

• Communications cost savings because calls between the various delegations of the company would go free.

Actually the integration of voice and data in one network is an old idea, have long since emerged solutions from different manufacturers, using multiplexers, WAN networks allow the use of business data (typically point connections to-point and frame-relay) for the transmission of voice traffic. The lack of standards and long-term depreciation of such solutions has not allowed a wide introduction of the same.

Example network connecting PBXs to Cisco routers that have VoIP support

[The#Danger]

Undeniably, the definitive implementation of the IP protocol from business to domestic and the emergence of a standard, VoIP (Voice over IP) could not be expected. The emergence of VoIP (Voice over IP) together with the lowering of the DSP's ( Processor Digital Signal), which are key in the compression and decompression of voice, are the elements that have made possible the launch of these technologies. For this boom there are other factors, such as the emergence of new applications or definitive commitment for VoIP (Voice over IP) vendors such as Cisco Systems and Nortel, Bay Networks. Moreover mobile operators are offering or plan to offer in the near future, IP services to enterprises. It said so far, we see that we can find three different types of networks IP:

• Internet. The state 's current network does not allow professional use for voice traffic.
• Public IP network. The operators offer businesses the connectivity to interconnect their local area networks as IP traffic is concerned. Can be considered as similar to the Internet, but with a higher quality of service and significant security enhancements. Operators are even offering guarantees low delay and / or bandwidth, making them very interesting for voice traffic.
• Intranet. IP network implemented by the company. Usually consist of several LAN networks ( Ethernet switching, ATM , etc.) that are interconnected by WAN Frame-Relay/ATM type, point to point lines, ISDN remote access, etc.. In this case the company has under its control virtually all network parameters, making it ideal for use in the transport of voice.

The IMTC VoIP Forum has reached an agreement that allows interoperability of different elements in a network can be integrated VoIP (Voice over IP). Due to the existence and the ITU H.323 standard, which covered most of the requirements for the integration of voice, it was decided that was the basis H.323 VoIP (Voice over IP). Thus, the VoIP (Voice over IP) should be considered as a clarification of H.323, so that in case of conflict , in order to avoid differences between the standards, it was decided that H.323 would have priority over VoIP (Voice over IP). The VoIP (Voice over IP) has as main objective to ensure interoperability between equipment from different manufacturers, fixing issues such as silence suppression, encoding of the voice and direction, establish new elements to enable connectivity to the traditional telephone infrastructure. These items relate primarily to the directory services and the transmission of signaling tone multifrequency (DTMF).

So far we have seen the possibility of using our IP network to connect the PBX to the same, but the fact that VoIP is supported in a level 3 protocol such as IP, allows flexibility in configurations that in many cases is undiscovered. One idea that appears immediately is that the traditional role of the switchboard would be distributed among the various elements of the VoIP network. In this scenario, technologies such as CTI (computer-telephony integration) will have a much simpler implementation. Will the passage of time and imagination of the people involved in these environments, which will be defining applications and services based on VoIP. We can now from a number of items already available in the market and, according to different designs allow us to build VoIP applications. These elements are:

• IP phones.
• Adapters PC.
• Hubs Telephone.
  
• Gateways (gateways RTC / IP).
• Gatekeeper.
• Multiple audio conference units. (MCU Voice)
  
• Directory Services.

The functions of the various components are easily understandable in view of the previous figure, but it is worth emphasizing some ideas. The Gatekeeper is an optional element in the network, but when present, all other items that contact the network should make use of it. Its function is to manage and control the resources of the network, so that there is no saturation conditions thereof. The Gateway is an essential element in most networks because its mission is to connect the VoIP network to the analog telephone network or ISDN. We can consider the Gateway as a box on one side has an interface LAN and the other has one or more of the following interfaces:

• FXO. For connection to or extensions on the basic telephone network.
• FXS. For connection to link analog phones or PBX.
• E & M. For specific connection to PBXs.
• BRI. ISDN basic access (2B + D).
• PRI. ISDN primary rate access (30B + D).
• G703/G.704. (E & M digital) specific connection to switchboards to 2 Mbps

The various components can reside on separate physical platforms, or we can find several elements coexisting in the same platform. Thus it is quite common to find together Gatekeeper and Gateway. We can also see how Cisco has implemented the Gateway functions in the router. An important aspect to highlight is that of delays in transmission of voice. Keep in mind that the voice is not very tolerant with them. In fact, if the delay introduced by the network is more than 300 milliseconds, it is almost impossible to have a fluent conversation. Because local area networks are not prepared in principle to this type of traffic, the problem may seem severe. Keep in mind that IP packets are variable length data traffic is often bursty. To try to ignore situations in which the voice is lost because we have a burst of data on the network, has developed the RSVP protocol, whose main function is to chop the large data packets and give priority to voice packets when congestion in a router. While this protocol will greatly assist the multimedia traffic over the network, it should be noted that RSVP does not guarantee quality of service as in advanced networks such as ATM to provide QoS as standard.

We can summarize by saying that VoIP is a technology that has all the elements for rapid development. As shown we can see that companies like Cisco, have incorporated it into its product portfolio, IP phones are already available and major global operators, as well as Telefonica, are actively promoting IP services to businesses, offering voice quality through thereof. On the other hand we already have a standard that guarantees interoperability between different manufacturers. The conclusion seems logical : we must consider how we can implement VoIP on our network.

[The#Danger]

To establish a voice communication using the Internet, the first thing you need is to establish the connection between the two user terminals equipped with the same or compatible software, you want to communicate, ie to establish an IP session, from Hence, the voice is digitized, compressed to occupy less bandwidth, and transmitted over the network like a data flow. Communication can be multimedia and transferred files or watch video while chatting. The appeal represents this approach is that in this case the rates that apply are those of the Internet, local fare is always at both ends and in many cases flat rate instead of the phone, depending on distance and airtime. The user accepts the lower quality of communication, which is offset by the cost savings you get. There are two other modes that occur in the case of establishing communication between a phone and a PC or between two telephones, using the Internet. In the first case it is necessary to have a gateway connection to the Internet on one side and one on the RTC, which digitize the voice and if not, to compress and package and perform the translation between IP addresses and numbers the RTC, making the process simultaneously in both directions. In the case of calls between phones via the Internet, the process is similar, using two gateways, one at each end, with several companies offering these services taking advantage of the economic advantages enjoyed by normal route voice calls through the network.

[zee789]

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[Benito]

Although in that year (1996) New software proliferated VoIP (Voice over IP) for clients, the lack of standards and the need to use a crude PC (computer) and end-user device discouraged the first potential supporters expect quality and efficiency and originality. The VoIP (Voice over IP) for the business market was virtually nonexistent and the first gateways (devices that pass access calls to and from the Internet or other IP networks, which allow you to use conventional telephones) were far from the "class carrier". But there is no doubt that things have changed. Several years of intense research and development in all areas of the industries of telecommunications networks and resulted in a market where the big traditional phone companies not only recognize that IP telephony is viable but also are adopting. Today, IP telephony is not just a potential source of revenue for service providers of all shapes and sizes, analysts and industry players increasingly consider a new paradigm for voice communications and data next century.

[DarenHawk]

Everyone has a phone in our house. And in general, we know that the phone has a cable card (RJ-11) similar to the network cable, and within two cables of copper, which is called twisted pair. This is a twisted pair that runs to the telephone, to a plate that is often called subscriber plate. Is the board that controls our line? In FACT, you CAN control lines Many, Not One, and has a density of ports depend on the manufacturer, ranges Between 8 and 16 subscribers (sometimes more, sometimes less). The exact value depends on the particular computer. The telephone is a set of related equipment. All these together form a very large take up May several rooms. As Mentioned, the call centers Often Are Designed to have a Very High Availability (They Say They Are carrier class, since it's available Are says 99.9% of the time s, which accounts for about 5 minutes per year of service interruption). To Achieve this goal, Have redundancy at multiple levels (processors, links, etc. ..) and in general Are connected to an Uninterrupted Power systems, which has a number of that Are batteries connected to a generator is activated That When you cut the light. Telephone service, together with the grid, one of the most reliable we know and use, as everything is very redundant and is designed to run forever. A telephone system is designed to minimize service downtime. It is a technology in which the interface is very important, people know it, hopes that when the tube up the tone, and if not the same as that expected to hear, annoying, it is very universal and widespread. All this is taken into account when providing telephone service.

[Pratim]

An important issue is the "routing" between plants, ie, as you know the subscriber's central core that has to be connected. We will call signaling information related to a call that is transmitted between two computers (the actual definition is broader, but this is particularly the most relevant for that matter). Can be divided into two groups : one that refers to the subscriber and calls itself (up, marc or, cut), and partly from plants (you drop something and you want to tell, etc). Through signage, the plant can locate central else has to call, to pay within that unit there to call, know who cut the communication, which he occupied, etc. Stations communicate with each other using different protocols, which are generally open standards, although in many cases the specifications are not easy (or cheap) to get. The most common protocols are three: R2, PRI and SS7. R2 is one of the oldest and has many different variants there-even-a variant, and spend all their information using 4 bits. SS7 is, moreover, one of the new complexes.

[tomstonsils]

Stations communicate with each other using different protocols, which are generally open standards, although in many cases the specifications are not easy (or cheap) to get. The most common protocols are three: R2, PRI and SS7. R2 is one of the oldest and has many different variants there-even-a variant Argentina, and spend all their information using 4 bits. SS7 is, moreover, one of the new complexes.

It is necessary that the two plants that are trying to communicate can speak the same protocol, so if you want to intercommunicate two plants that do not support the same protocols, it is necessary to use an intermediate central to translate the information. About the link which passes both the signaling and the voice itself, there are many types. The most common are known and E1 or E3 (European), with its variants T1 and T3 (used primarily in the United States). Copper cables are very similar to coaxial cable, which can be 75 or 120 ohms. The E1 has 2 Mbps (32 channels of 64kbps), the E3 is 32Mbps (512 channels of 64kbps). In many areas when it comes to this type of link is given importance only to the bandwidth, but in our case we are also interested in the number of times-lots in which they can divide. However, you cannot occupy all channels to pass all subscribers. In necessary to warn that no calls and that kind of information. For example, in the case of a 30 E1 channels are often used for the passage of the voice, 1 for framing (0) and 1 for signaling (the 15). The framing is usually found (among other things) the CRC of the other 31 (although it depends on the configuration), so that if a given frame is corrupted, it can be noticed and acted upon.

[Sukhvinder]

This convergence of voice, data and video over a single network means for a company to adopt a lower capital cost, simplified support and network configuration and better integration of remote locations and branch offices in corporate network installations. IP telephony uses the data network to provide voice communications throughout the enterprise through a single network for voice and data. It is clear that having a network instead of two, is beneficial for any organization. VoIP provide the branches of a company, free communication between them, with the cost savings that would entail. Not only among its branches, but between suppliers, dealers and end, communications could be made completely free. In addition, the communications network of the company would be greatly simplified, and should not duplicate the network wiring, because it would use the data network for voice. This capability enables companies to reduce the cost of fax and telephone, gathering data services, voice, fax and video, and build new network infrastructure for advanced e-commerce applications.

[DayanandPatil]

Working with IP is that you cannot have certain bandwidth, packet collisions there, among other disadvantages. Beyond that, controlling massive traffic to IP level is not easy, there are many complex variables, and it is very difficult to do a large scale. Also, if you are using a link on which you have no control (the Internet being a prime example) and cannot guarantee anything. And it's not something to dismiss, because the phone is something that always likes to tell. The problem with VOIP (Voice over IP), then, is not the protocol, but the IP network. The IP network is not intended to give any guarantees is a best effort network. The worst problem is not only the bandwidth (and that with good compression, it can transmit up modem), but the latency. You cannot use VOIP (Voice over IP) with 200ms latency. When the user is speaking and the other one is listening, is not acceptable to have 200ms pause. And, as mentioned above, the telephone, with TDM technology avoids these problems.

[kyosang]

While the average time allowed for a phone call is about 3 minutes, Internet access to the user is typically connected from 20 to 30 minutes, which means that during this time no one can make use of the telephone line the disadvantages that entails. To find a solution to this problem, some manufacturers have developed a system that converts voice calls into IP data stream that can be sent directly to users that they are addressed. The operation is as follows: when an incoming call is received on the telephone, the network is able to detect whether the target line is busy in an Internet session and then immediately rerouted to a dedicated server that scans and makes a data frame, converts the phone number to the destination Internet address and immediately sends a message that represents an icon on the terminal screen indicating that a call is waiting, asking for their acceptance. For outgoing calls the reverse process takes place. If the user has the required minimum bandwidth, you can talk and keep the session Internet at the same time neglect the time spent surfing the Internet, be reassured that he will not miss any calls. In this way, it generates extra business for the network operator and Internet service provider (ISP).