Department of Computer Science & Engineering University of W
Course # 350 VOICE OVER INTERNET PROTOCOL (VoIP)
Submitted to:
Dr. A
Presented by:
John Doe Jane Doe
Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
Content
1. Introduction 1.1. 1.2. What is VoIP How does it work
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2. VoIP: History and Growth
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3. Understanding VoIP Networks 3.1 Functions 3.2 Components 3.3 Signaling Protocols 3.4 VoIP Service Consideration 4. The World – Implementing VoIP 5. Future Directions & Conclusion
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Department of Computer Science, Windsor, Canada
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
VoIP (Voice over Internet Protocol)
1.1 What is VoIP? A category of hardware and software that enables people to make telephone calls via the Internet. Voice signals are converted to packets of data, which are transmitted on shared public lines, hence avoiding the tolls of the traditional, public-switched telephone network (PSTN).
1.2 How does it work? a) At the source, analog audio signals are converted into digital data that can be transmitted over the Internet. The analog signals at first reach a gateway. b) At the gateway, analog signals are compressed into IP packets. These IP packets are routed over the Internet through various other gateways. Each packet is given an address that tells the gateways, to which other gateway the packet needs to be routed. IP packets contain a header (to control communication) and a payload to transport data: VoIP use it to travel across the network and reach the destination. c) At the destination gateway, the packets are decompressed and reassembled into analog audio signals and delivered to the receiver.
VoIP can be deployed in various ways. There are basically 4 main ways:
Telephone to Telephone Computer to Computer Telephone to Computer Computer to Telephone
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
Telephone to Telephone: When a connection from one telephone to another is to be established, it starts with connecting to a gateway. The picture above shows the scenario where the analog phone is connected to the “Analog Telephone Adapter”. This ATA is connected to the Internet. When a destination number is dialed, the ATA converts analog signals into digital data and routes it to another router which is nearest to the destination telephone number.
Computer to Computer: When a connection from one computer to another computer is to be established, at the source, the analog audio signals are converted to digital data by the computer. The digital data is then routed through the Internet till the destination computer is reached. The
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
destination computer then converts the digital data into analog signals and delivers it to the receiver.
Telephone to Computer: When a connection from a telephone to a computer is to be established, at the source, the analog audio signals reach the Analog Telephone Adapter (ATA). The ATA converts analog signals to digital data. It routes the digital data to the destination through the Internet and reaches the destination computer. The computer then converts the digital data into analog signals and delivers it to the receiver.
Computer to Telephone: When a connection from one computer to a telephone is to be established, at the source, the analog audio signals are converted to digital data by the computer. The digital data is then routed through the Internet till the destination Analog Telephone Adapter(ATA) which then delivers the analog signals to the destination telephone. 1.3 Why use VoIP? More efficient than POTS: For routing of data packets, VoIP employs Packet switching. It is very efficient because it minimizes the amount of time that a connection must be maintained between two sources and thus reduces the load on a network. Digital signal is more noise tolerant than the analog signal. Also, it can be better controlled: we can compress it, route it, and convert it to a new better format.
Lower cost: In general, phone service via VoIP costs less than equivalent service from traditional sources. VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, we are bypassing the phone company (and its charges) entirely. There are also some cost savings due to using a single network to carry voice and data.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
In the most extreme case, users see VoIP phone calls (even international) as FREE. While there is a cost for their Internet service, using VoIP over this service may not involve any extra charges, so the users view the calls as free. There are a number of services that have sprung up to facilitate this type of "free" VoIP call. E.g. Free World Dialup, Skype, and Google etc. Increased Functionality: A VoIP enabled phone is portable. It just needs to be connected to the Internet. We can make and receive calls anywhere we go without being charged more for the „roaming facility‟. It is possible to exchange data (like images, graphs, videos etc.) simultaneously as we are talking.
2. VoIP: History and Growth
Voice over IP began as the result of work done by some hobbyists in Israel in 1995 when only PC-to-PC communication was available. Later in 1995, Vocaltec, Inc. released Internet Phone Software. This software was designed to run on a home PC. It utilized sound cards, microphones and speakers, much like the PC phones used today. The software was called "Internet Phone" and used the H.323 protocol instead of the SIP protocol that is more prevalent today. Vocaltec had initial success with Internet Phone, and had a successful IPO in 1996. A major drawback in 1995 was the lack of broadband availability, and as such, this software used modems which resulted in poor voice quality when compared to a normal telephone call. However, this was still a major milestone as it represented the first ever IP phone. It is also worthwhile to mention that one of Vocaltec‟s key employees is also a founder of WhichVoIP.com. By 1998 VoIP had reached some potential. A number of entrepreneurs started setting up gateways to allow first PC-to-Phone and later Phone-to-Phone connections. Some of these entrepreneurs started by providing customers a facility to make free phone calls using the
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
regular phone. Every phone call which the user made had an advertisement at the beginning and at the end of the call. This service was only available to users in North America and allowed the users to make free long distance calls. This “free to the customer” marketing model, was sponsored by various advertising companies or agencies. These services often required the services of a PC to originate the call, although the actual communication was from „phone to phone‟. At this stage, VoIP traffic represented rather less than 1% of voice traffic. In 1998 three IP switch manufacturers introduced equipment capable of switching. At present, most IP switching and routing equipment suppliers offer VoIP as either a standard or as an option on their mid-range and up equipment. VoIP traffic exceeded 3% of voice traffic by 2000, and is forecast to grow rapidly to between 25% and 40% of all international voice traffic by end of 2005. Today there are two standards for VoIP switching and gateways: SIP and H.323. The former primarily relates to end-user IP Telephony applications, whereas the latter is a new ITU standard for routing between the circuitswitched and packet-switched worlds used for termination of an IP originated call on the PSTN, but the converse is increasingly becoming common. Now, in 2005, major voice quality issues have long since been addressed and VoIP traffic can be prioritized over data traffic to ensure reliable, clear sounding, unbroken telephone calls. Revenue from VoIP equipment sales alone are projected to reach around $3 billion this year and are being forecast to be over $8.5 billion by the end of 2008. This is primarily being driven by low cost unlimited calling plans and the abundance of enhanced and useful telephony features associated with VoIP technology. Voice is the latest core function making its way into the IP world. In the years since VoIP has been introduced, a growing list of technology providers have begun to offer PC telephony software. There is a spate of gateway manufacturers entering the market. Until recently, VoIP provided PC-to-PC telephony primarily over intranets typically found in a business environment. With the introduction of gateway infrastructure outfitted with VoIP technology, users can now look forward to the widespread increase in the usage of Internet telephony.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
This is a phenomenal growth rate and with the rapid introduction of Video over IP fueling demand, the future of this technology is truly exciting. Video over IP follows the same concept as VoIP but in this case enables the transmission of video signals. As such, video phones are becoming common and many companies are already offering attractive packages. Voice over Internet Protocol, VoIP or Broadband phone service as it is often referred to, is changing the telephony world. Traditional phone lines are slowly being phased out as businesses and households around the world embrace the benefits and features that VoIP technology has to offer.
3. Understanding VoIP Networks
To understanding VoIP Networks, one must have to have a clear idea about VoIP functions, its major components and protocols to use and of course have a knowledge of factors to consider while deploying a VoIP network. During the last few years of development phase, today‟s VoIP has achieved these standards and attributes which are briefly discussed below. Due to the limited scope of the report details explanation of the technical terms and factors could not be explained in details.
3.1 VoIP Functions Before going into a discussion of the components that make up a VoIP solution, it is important to understand the basic functions of VoIP, particularly as they compare to current PSTNs. As mentioned above, in order to enable organizations to adopt VoIP as a viable solution, its components must be able to perform the same functions as the PSTN network. These are:
Signaling Database services Call connect and disconnect (bearer control) CODEC operations
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
Signaling Signaling is the way that devices communicate within the network, activating and coordinating the various components needed to complete a call. In a PSTN network, phones communicate with a Class 5 switch (analog) or traditional private branch exchange (PBX) (digital) for call connection and call routing purposes. In a VoIP network, Signaling is accomplished by the exchange of IP datagram messages between the VoIP components.
Database Services Database services are a way to locate an endpoint and translate the addressing that two networks use. A PSTN uses phone numbers to identify endpoints. A VoIP network uses an IP address and port number to identify an endpoint.
Call Connect and Disconnect (Bearer Control) The connection of a call is made by two endpoints opening a communication session between one another. In the PSTN, the public (or private) switch connects logical (Digital Signal) DS-0 channels through the network to complete the calls. In a VoIP implementation, this connection is a multimedia stream (audio, video, or both) transported in real time. When a communication is complete, the IP sessions are released and optionally network resources are freed.
CODEC Operations Traditional voice communication is analog, while data networking is digital, as a result, the network needs a way to be able to convert the voice into a format that it can transport. Since the PSTN is often analog, this is not necessarily a major function, however, for VoIP, it is necessary for “packetiz-ing” the voice. The process of converting analog waveforms to digital information is done with a coder-decoder (CODEC, which is also known as a voice coder-decoder [VOCODER]). There are many ways an analog voice signal can be transformed, all of which are governed by various standards.
The output from the CODECs is a data stream that is put into IP packets and transported across the network to an endpoint. These endpoints must use the standards, as well as a
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
common set of CODEC parameters. If two endpoints use different standards or parameters then the communication will be unintelligible. Table 1 lists some of the more important encoding standards covered by the International Telecommunications Union (ITU).
3.2 VoIP Components The major components of a VoIP network, while different in approach, deliver very similar functionality to that of a PSTN and enable VoIP networks to perform all of the same tasks that the PSTN does. The one additional requirement is that VoIP networks must contain a gateway component that enables VoIP calls to be sent to a PSTN, and visa versa. There are basically four major components to a VoIP network.
Call Processing Server/IP PBX User End-Devices Media/VOIP Gateways IP network
Call Processing Server / IP PBX The call processing server, otherwise known as an IP PBX, is the heart of a VoIP phone system, managing all VoIP control connections. Call processing servers are usually software-based and can be deployed as a single server, cluster of servers, or a server farm with distributed functionality.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
VoIP communications require a Signaling mechanism for call establishment, known as control traffic, and actual voice traffic, known as voice stream or VoIP payload. VoIP control traffic follows the client-server model, with VoIP terminals, including messaging servers that hold voice-mail messages representing the clients that communicate to the call processing servers. VoIP payload flows in a peer-to-peer fashion – from every VoIP terminal to every other VoIP terminal. In this case, the VoIP terminals determine traffic flows and the call processing servers negotiate those flows within the control messages. A typical VoIP setup with Call Processing Server is shown in Figure 1. 12
3456 Figure 1
User End-Devices The user end-devices consist of VoIP phones and desktop-based devices. VoIP phones maybe software based (“soft phones”) or hardware based (“hard phones” or “handsets”, like traditional phones). VoIP phones use the TCP/IP stack to communicate with the IP network, as such, they are allocated an IP address for the subnet on which they are installed. Typically, VoIP phones use DHCP to auto-configure themselves, with the DHCP server telling the phone about the location of the configuration server, which most of the time is identical to the call processing server.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
Soft phones are software application running on computers, usually targeted towards mobile users. They have the same base features as VoIP phones.
Consoles, on the other hand, are applications with certain control characteristics. Consoles usually include a Soft phone, but may also interact with a legacy phone, via a voice gateway or a VoIP phone. Consoles are special-purpose applications to control call distribution. This includes receptionist consoles with the ability to connect calls, executive consoles with the ability to see call states of special groups of phones, and customer relations consoles with the ability to support call distribution. Consoles should be installed on dedicated desktop computers, with no access to the Internet and only controlled access to data network services, in order not to expose the voice network. Consoles are usually static and should be confined to their own network within the module.
Media/VOIP Gateways/Gatekeepers The terms gateway and gatekeeper are sometimes used interchangeably. Traditionally gatekeepers have been mainly used for Call Admission and control and bandwidth management. But this has changed recently, as technology has allowed this functionality to co-exist within traditional gateways.
The major function of media gateways is analog-to-digital conversion of voice and creation of voice IP packets (CODEC functions). In addition, media gateways have optional features, such as voice (analog and/or digital) compression, echo cancellation, silence suppression, and statistics gathering. The media gateway forms the interface that the voice content uses so it can be transported over the IP network. Media gateways are the sources of bearer traffic. Typically, each conversation (call) is a single IP session transported by a Real-time Transport Protocol (RTP) that runs over UDP or TCP.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
Media gateways exist in several forms. For example, media gateways could be a dedicated telecommunication equipment chassis, or even generic PC running VoIP software. Their features and services can include some or all of the following:
Trunking gateways that interface between the telephone network and a VoIP network. Such gateways typically manage a large number of digital circuits. Residential gateways that provide a traditional analog interface to a VoIP network. Examples of residential gateways include cable modem/cable set-top boxes, xDSL devices and broadband wireless devices.
Access media gateways that provide a traditional analog or digital PBX interface to a VoIP network. Examples include small-scale (enterprise) VoIP gateways. Business media gateways that provide a traditional digital PBX interface or an integrated soft PBX interface to a VoIP network. Network access servers that can attach a modem to a telephone circuit and provide data access to the Internet.
IP Network VoIP network can be viewed as one logical switch. However, this logical switch is a distributed system, rather than that of a single switch entity; the IP backbone provides the connectivity among the distributed elements. Depending on the VoIP protocols used, this system as a whole is sometimes referred to as a soft switch architecture. The IP infrastructure must ensure smooth delivery of the voice and Signaling packets to the VoIP elements. Due to their dissimilarities, the IP network must treat voice and data traffic differently. If an IP network is to carry both voice and data traffic, it must be able to prioritize the different traffic types, as VoIP traffic is extremely sensitive to latency. IP networks are quite different from the circuit-switch infrastructure in that it is a packetnetwork, and it is based on the idea of statistical availability. Thus network resources are not completely tied up for the duration of the call, unlike in a circuit-switched environment. Class of service (CoS) ensures that packets of a specific application are given priority. This prioritization is required for real-time VoIP applications to ensure that the voice service is unaffected by other traffic flows.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
3.3 VoIP Signaling Protocols VoIP Signaling protocols are the enablers of the VoIP network. The protocols determine what types of features and functionality are available, as well as how all of the VoIP components interact with one another.
There are a variety of VoIP protocols and implementations, with a wide range of features that are currently deployed. Two major standards bodies govern multimedia delivery (voice being one type) over packet-based networks: International Telecommunications Union (ITU) and Internet Engineering Task Force (IETF). H.323 is the ITU‟s standard for establishing VOIP connections, while IETF uses Session Initiation Protocol (SIP) as its standard. The protocols listed below are the most prevalent; there are a few other protocol options available, but due to limited scope of this report only H.323 and SIP are brought into lights.
H.323 Real-time Transport Protocol (RTP) Real-time Transport Control Protocol (RTCP) Media Gateway Control Protocol (MGCP Session Initiation Protocol (SIP) Megaco/H.248
H.323 H.323 is the ITU recommendation. It is a packet-based multimedia communication system that is a set of specifications. These specifications define various Signaling functions, as well as media formats related to “packetized” audio and video services.
H.323 networks consist of Call Processing Servers, (media) gateways and gatekeepers. Call Processing Servers provide call routing, and communication to VOIP gateways and end devices. Gateways serve as both the H.323 termination endpoint and interface with nonH.323 networks, such as the PSTN. Gatekeepers function as a central unit for call admission control, bandwidth management and call Signaling. Although the gatekeeper is
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
not a required element in H.323, it can help H.323 networks to scale to a larger size, by separating call control and management functions from the gateways. Let‟s look at the main H323 process:
With each call that is initiated, a TCP session (H.225.0 protocol) is created. This TCP connection is maintained for the duration of the call. A second session is established using the H.245 protocol. This TCP-based process is for capabilities exchange, master-slave determination, and the establishment and release of media streams (explanation for which are beyond this reports scope).
The H.323 quality of service (QoS) delivery mechanism of choice is the Resource Reservation Protocol (RSVP). This protocol is not considered to have good scaling properties due to its focus and management of individual application traffic flows.
Although H.323 may not be well suited in service provider spaces, it is well positioned to deploy enterprise VoIP applications.
Session Initiation Protocol (SIP) The Session Initiation Protocol (SIP, RFC 2543) is part of IETF's multimedia data and control protocol framework. SIP is a powerful client-server Signaling protocol used in VoIP networks. SIP handles the setup and tears down of multimedia sessions between speakers; these sessions can include multimedia conferences, telephone calls, and multimedia distribution.
SIP is basically a text-based Signaling protocol transported over either TCP or UDP, and is designed to be lightweight. It uses invitations to create Session Description Protocol (SDP) messages to carry out capability exchange and to setup call control channel use. These invitations allow participants to agree on a set of compatible media types. SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can inform the server of their current location (IP address or URL) by sending a registration message to a registrar. This function is powerful and often needed for a highly mobile
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
voice user base. The SIP client-server application has two modes of operation; SIP clients can ether signal through a proxy or redirect server.
Using proxy mode, SIP clients send requests to the proxy and the proxy either handles requests or forwards them on to other SIP servers. Proxy servers can insulate and hide SIP users by proxying the Signaling messages; to the other users on the VoIP network, the Signaling invitations look as if they are coming from the proxy SIP server.
Under redirect operation, the Signaling request is sent to a SIP server, which then looks up the destination address. The SIP server returns the destination address to the originator of the call, who then signals the SIP client
3.4 VoIP Service Considerations Now that the functions, components and protocols related to VoIP traffic are clear, let‟s take a quick look at some of the issues that must carefully be consider when deploying VoIP solutions. The important considerations are as follows: o o o o o o o Latency Jitter Bandwidth Packet loss Reliability Security Interoperability
Latency Latency (or delay) is the time that it takes a packet to make its way through a network endto-end. In telephony terms, latency is the measure of time it takes the talker's voice to reach the listener's ear. Large latency values do not necessarily degrade the sound quality of a phone call, but the result can be a lack of synchronization between the speakers, such that there are hesitations in the speaker' interactions. Generally, it is accepted that the end-toend latency should be less than 150 ms for toll quality phone calls.
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Voice over Internet Protocol (VoIP)
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Jitter Jitter is the measure of time between when a packet is expected to arrive to when it actually arrives. Although some amount of jitter is to be expected, severe jitter can cause voice quality issues because the media gateway might discard packets arriving out of order. In this condition, the media gateway could starve its play-out buffer and cause gaps in the reconstructed waveform.
Bandwidth In order to communicate Telco-grade voice (or similarly, other real-time applications such as moving video) two different approaches can be attempted. To transmit information of the highest quality over unrestricted bandwidth or to reduce the bandwidth required for transmitting information (voice) of a given quality. Compression and decompression (CODEC) of digital signals is a means of reducing the required bandwidth or transmission bit rate. If, for example, a digital signal contains a string of zeroes, it will be economical to transmit a code indicating that a string of zero follows along with the length of the string. Many different algorithms for compression and decompression of digital codes have been constructed for these kinds of purposes.
Packet Loss It is important that bearer and Signaling packets not be discarded, otherwise, voice quality or service disruptions might occur. By configuring CoS (Class of Service) parameters, administrators can give packets of greater importance a higher priority in the network, thus ensuring packet delivery for critical applications, even during times of network congestion As long as the amount of packet loss is less than five percent for the total number of calls, the quality generally is not adversely affected. It is best to drop a packet, versus increasing the latency of all delivered packets by further buffering them.
Reliability Traditional data communication strives to provide reliable end-to-end communication between two peers. They use checksum and sequence numbering for error control and some
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Voice over Internet Protocol (VoIP)
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form of negative acknowledgement with a packet retransmission handshake for error recovery. VoIP networks typically are supposed to leave the proper error control and error recovery scheme to higher communication layers. They can thus provide the level of reliability required, taking into account the impact of the delay characteristics. Therefore, UDP is the transport level protocol of choice for voice and like communications.
Security On the Internet, since anybody can capture packets meant for someone else, security of voice communication becomes an important issue. Some measure of security can be provided by using encryption and tunneling. Usually, the common tunneling protocol used is Layer 2 Tunneling protocol, and the common encryption mechanism used is Secure Sockets Layer (SSL).
4. The World – Implementing VoIP
Organizations are increasingly looking to VoIP as an attractive alternative to traditional PSTN. Vonage, Skype etc. comes as very well known names as we talk about VoIP and its implementation. The brief part discussed below should give a precise idea about how VoIP is being implemented in Enterprises and in the commercial world.
SKYPE It‟s a global p2p telephony company that is changing the telecommunication world by consumer free, superior quality calling worldwide with the help of VoIP network. All that Skype requires at the user end are a PC, microphone and speakers. It‟s simple to install regardless of pc environments and can set up without server and workstation configuration. It can work behind most firewalls and gateways without providing new security risks. Skype calls are encrypted always.
VoIP CALCULATORS These are free online tools for performing a variety of technical calculations relating to the design of VoIP systems. They help to assist engineers and managers involved in the deployment of VoIP network. They include VoIP Select, VoIP Traffic Calculator and
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Voice over Internet Protocol (VoIP)
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others. There are more of them like VoIP Lite, Website Traffic Calculator, dual gatekeepers.
NETSCAPE COOLTALK Another widely used VoIP implementation in Netscape Cool Talk. Its hardware setup is same like SKYPE. It has many other different features like:
Audio conferencing High quality Answering machine Shared tool Chat tool Dial-up IP support Multimedia enabled Multiplatform supported
NET2PHONE Net2Phone is a provider of low-cost, high-quality, retail voices over services, either directly or through partners. The company is comprised of two wholly owed subsidiaries Net2Phone Global Services (NGS) and Net2Phone Cable Telephony (NCT). NCT offers cable operators the ability to deliver a viable cable telephony service to their video and high speed data customers. For many operators, offering voice enables the „triple play‟ combination of VDO, high-speed of data and telephony, which not only delivers profitable top-line revenue but also aids in the reduction of subscriber churn. NCT manages the cable operators‟ networks for them, enabling them to assure quality services from all inceptions to completion.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
5. Future Directions
Before we can understand the future need and vision for this emerging technology and start making predictions, we must look a bit deep into the pros and cons of it and also the remaining barriers.
There are some nice advantages of VoIP. As it is digital, it may offer features and services not available in a traditional phone. With the connection of high speed internet, we need not maintain and pay the cost for a line just to make telephone calls. Depending on plans we can talk for as long as we want with any person in the world. We can also talk with many people at the same time without any additional cost.
One of the great advantages that VoIP offers is the capability to use a single network to support a wide range of applications, including data, voice and VDO. A single multifunctional network, as opposed to multiple discreet networks, means lower capital cost and lower operating cost.
One can easily exploit the technologies and partnerships to their own advantage like cable competitors and can be easily benefited. VoIP is easy to understand as well as to operate hence used by many business houses and individual also it minimize cost because it cost only about half the cost of traditional phone services. Also it happens in making long distance calls for free and keeps a track of all the calls, which are coming and outgoing even recording the minutes.
But still there are some facts shadowing the development and popularity of this blessed technology. Like some VoIP services don‟t work during power outages and the service provider may not provide back up power. Not all VoIP services connect directly to emergency service through 9-1-1. VoIP providers may or may not offer directory assistance or white page listings. And a lot of other issues rise too as we talk about the future prediction and vision of this technology.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
Vision and Predictions The future of VoIP is exactly not decided yet. Though we can say it is here to stay probably but we don‟t yet know which direction it is turning up to. We can imagine its existence with many other computers, internet and with people almost everywhere communicating in a real time fashion.
May be in the next few years the researchers and workers on VoIP will overcome the problems and trouble that we are facing now. Unfortunately we have to report some problem with the integration between VoIP architecture and internet. As we can easily imagine, voice data communication must be a real time stream (we couldn‟t speak, wait for many seconds, and then hear other side answering). This is in contrast with the Internet heterogeneous architecture that can be made of many routers, about 20-30 or more and can have a very high round trip time, so we need to modify something to get it properly working. As we can see that the vision of the researchers will be to overcome these problems and may be introduce more features.
We see that Voice over Internet Protocol is a very interesting and helpful protocol based on which so many varieties of soft wares are serving us in our day to day life and making our world even shorter and bringing people more close to each other. When the researchers will come up with solutions with the problems that they are facing now with this for sure they will make it more efficient and colorful features.
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Voice over Internet Protocol (VoIP)
Course offered by: Dr. A.K Aggarwal
Reference: 1. Understanding VoIP networks – Stefan Brunner (2004) 2. Network Convergence and Voice over IP – Debashish Mitra (March 2001) 3. Voice over IP – Wikipedia (http://en.wikipedia.org/wiki/Voice_over_IP) 4. VoIP Review - http://www.voipreview.org
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