Voice Over Internet Protocol
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23-09-2008, 12:37 AM


Definition
Using an ordinary phone for most people is a common daily occurrence as is listening to your favorite CD containing the digitally recorded music. It is only a small extension to these technologies in having your voice transmitted in data packets. The transmission of voice in the phone network was done originally using an analog signal but this has been replaced in much of the world by digital networks. Although many of our phones are still analog, the network that carries that voice has become digital.

In todays phone networks, the analog voice going into our analog phones is digitized as it enters the phone network. This digitization process, shown in Figure 1 below, records a sample of the loudness (voltage) of the signal at fixed intervals of time. These digital voice samples travel through the network one byte at a time.
At the destination phone line, the byte is put into a device that takes the voltage number and produces that voltage for the destination phone. Since the output signal is the same as the input signal, we can understand what was originally spoken.
The evolution of that technology is to take numbers that represent the voltage and group them together in a data packet similar to the way computers send and receive information to the Internet. Voice over IP is the technology of taking units of sampled speech data .
So at its most basic level, the concept of VoIP is straightforward. The complexity of VoIP comes in the many ways to represent the data, setting up the connection between the initiator of the call and the receiver of the call, and the types of networks that carry the call.

Using data packets to carry voice is not just done using IP packets. Although it won't be discussed, there is also voice over Frame Relay (VoFR) and Voice over ATM (VoATM) technologies. Many of the issues VoIP being discussed also apply to the other packetized voice technologies.
The increasing multimedia contents in Internet have reduced drastically the objections to putting voice on data networks. Basically, the Internet objections to putting voice on data networks. Basically, the Internet Telephony is to transmit multimedia information in discrete packets like voice or video over Internet or any other IP-based Local Area Network (LAN) or Wide Area Network (WAN). The commercial Voice Over IP (Internet Protocol) was introduced in early 1995 when VocalTec introduced its Internet telephone software. Because the technologies and the market have gradually reached their maturity, many industry leading companies have developed their products for Voice Over IP applications since 1995
VoIP, or "Voice over Internet Protocol" refers to sending voice and fax phone calls over data networks, particularly the Internet. This technology offers cost savings by making more efficient use of the existing network.

Traditionally, voice and data were carried over separate networks optimized to suit the differing characteristics of voice and data traffic. With advances in technology, it is now possible to carry voice and data over the same networks whilst still catering for the different characteristics required by voice and data.
Voice-over-Internet-Protocol (VOIP) is an emerging technology that allows telephone calls or faxes to be transported over an IP data network.
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Aarti88
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11-01-2010, 05:57 PM

please post me a 20 pages documentation on this topic..it will b very helpful...thanqSmile
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13-02-2010, 05:09 PM


.doc   Voice Over Internet Protocol.doc (Size: 241 KB / Downloads: 171)

ABSTRACT
The increasing multimedia contents in Internet have reduced drastically the objections to putting voice on data networks. Basically, the Internet objections to putting voice on data networks. Basically, the Internet Telephony is to transmit multimedia information in discrete packets like voice or video over Internet or any other IP-based Local Area Network (LAN) or Wide Area Network (WAN). ,
Voice over IP is the technology of taking units of sampled speech data .So at its most basic level, the concept of VOIP is straightforward.
Traditionally, voice and data were carried over separate networks optimized to suit the differing characteristics of voice and data traffic. With advances in technology, it is now possible to carry voice and data over the same networks whilst still catering for the different characteristics required by voice and data.
Voice-over-Internet-Pro to co I (VOIP) is an emerging technology that allows telephone calls or faxes to be transported over an IP data network.
INTRODUCTION
Using an ordinary phone for most people is a common daily occurrence as is listening to your favorite CD containing the digitally recorded music. It is only a small extension to these technologies in having your voice transmitted in data packets. The transmission of voice in the phone network was done originally using an analog signal but this has been replaced in much of the world by digital networks. Although many of our phones are still analog, the network that carries that voice has become digital.
In today's phone networks, the analog voice going into our analog phones is digitized as it enters the phone network. This digitization process, shown in Figure 1 below, records a sample of the loudness (voltage) of the signal at fixed intervals of time. These digital voice samples travel through the network one byte at a time.
TIME ”p-
Figure 1. Digital Sampling of an analog voice signal
-'¦: the destination phone line, the byte is put into a device that takes the voltage number and produces that voltage for the destination phone. Since the output signal is the same as the input signal, we can understand what was originally spoken. The evolution of that technology is to take numbers that represent the voltage and group them together in a data packet similar to the way computers send and receive information to the Internet. Voice over IP is the technology of taking units of sampled speech data .
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So at its most basic level, the concept of VoIP is straightforward. The complexity of VoIP comes in the many ways to represent the data, setting up the connection between the initiator of the call and the receiver of the call, and the types of networks that carry the call.
Using data packets to carry voice is not just done using IP packets. Although it won't be discussed, there is also voice over Frame Relay (VoFR) and Voice over ATM (VoATM) technologies. Many of the issues VoIP being discussed also apply to the other packetized voice technologies.
The increasing multimedia contents in Internet have reduced drastically the objections to putting voice on data networks. Basically, the Internet objections to putting voice on data networks. Basically, the Internet Telephony is to transmit multimedia information in discrete packets like voice or video over Internet or any other IP-based Local Area Network (LAN) or Wide Area Network (WAN). The commercial Voice Over IP (Internet Protocol) was introduced in early 1995 when VocalTec introduced its Internet 'alephone software. Because the technologies and the market have gradually reached :neir maturity, many industry leading companies have developed their products for Voice Over IP applications since 1995.
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Voice over internet protocol
VoIP, or "Voice over Internet Protocol" refers to sending voice and fax phone calls over data networks, particulariy the Internet. This technology offers cost savings by making more efficient use of the existing network.
Traditionally, voice and data were carried over separate networks optimized to suit the differing characteristics of voice and data traffic. With advances in technology, it is now possible to carry voice and data over the same networks whilst still catering for the different characteristics required by voice and data.
Voice-over-Internet-Protocol (VOIP) is an emerging technology that allows telephone calls or faxes to be transported over an IP data network. The IP network could be
¢ A local area network in an office
¢ A wide area network linking the sites of a large international organization
¢ A corporate intranet
¢ The internet
¢ Any combination of the above
There can be no doubt that IP is here to stay. The explosive growth of the Internet, making IP the predominate networking protocol globally, presents a huge opportunity to dispense with separate voice and data networks and use IP technology for voice traffic as well as data. As voice and data network technologies merge, massive infrastructure cost savings can be made as the need to provide separate networks for voice and data can be eliminated.
Most traditional phone networks use the Public Switched Telephone Network(PSTN), this system employs circuit-switched technology that requires a dedicated voice channel to be assigned to each particular conversation. Messages are sent in analog format over this network.
Today, phone networks are on a migration path to VoIP. A VoIP system employs a packet-switched network, where the voice signal is digitized, compressed and
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packetized. This compressed digital message no longer requires a voice channel. Instead, a message can be sent across the same data lines that are used for the Intranet or Internet and a dedicated channels is no longer needed. The message can now share bandwidth with other messages in the network.
Normal data traffic is carried between PC's, servers, printers, and other networked devices through a company's worldwide TCP/IP network. Each device on the network has an IP address, which is attached to every packet for routing. Voice-over-IP packets are no different.
Users may use appliances such as Symbol's NetVision phone to talk to other IP phones or desktop PC-based phones located at company sites worldwide, provided that a voice-enabled network is installed at the site. Installation simply involves assigning an IP address to each wireless handset.
VOIP lets you make toll-free long distance voice and fax calls over existing IP data networks instead of the public switched telephone network (PSTN). Today business that implement their own VOIP solution can dramatically cut long distance costs between two or more locations.
SCENARIOS IN INTERNET TELEPHONY
PC to PC
PC to Phone
Phone to Phone
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SCENARIO 1: PC TO PC
Scenario 1: PC to PC
¢ Need a PC with sound card.
¢ IP Telephony software: Cuseeme, Internet Phone, Video optional
SCENARIO 2 : P C TO PC
Scenario 2: PC to Phono
# Need a gateway that connects IP network to phone network (Router to PBX)
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What are VoIP Gateways
Many companies still have separate voice and data networks but would like to take advantage of the benefits of using Internet Telephony. A gateway therefore converts a telephone conversation into the correct format as data packets to enable it to travel across a data network as Internet Telephony. Gateways are required at both ends of a telephone conversation so that voice can be converted then reconverted back into intelligible language at the other end. VegaStream VoIP Gateways are available to cater for different sized companies according to the number of simultaneous conversations that would typical take place. For example, the Vega 50 is available with 8 ports to allow up to 8 simultaneous conversations.
SCENARIO 3: PHONE TO PHONE
Scenario 3: Phone to Phone
Gateway C iateway
¢ Need more gateways that connect IP network to phone networks.
¢ The IP network could be dedicated intra-net or the Internet.
« The phone networks could be intra-company PBXs or the carrier switches.
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WORKING
In VoIP, analog voice signals is digitized using PCM. These digital voice samples are then buffered on an IP gateway. This device converts the PCM data stream into a compressed IP packet stream using DSP's (Digital Signal Processors). DSP's are responsible for converting from analog to digital as well as compression. The set of PCM samples are analysed as a discrete set of binary data, it checks the speech for all the moments of silence, which are a lot. Even when we speak, there are pauses in between that go unnoticed to the human ear, but are quiet discernible to the sampling device. The length and beginning of these pauses is noticed, while the remaining silence is removed from the data set. Similarly redundant data is also removed, making the data set more compact. Finally, an IP header is attached to this compressed data, which is then sent out on the network as discrete data packets. Once the voice
packet is sent out, it finds its way to the destination just like any other data packet. It passes through various routers and switches to reach the destination gateway. Here, it gets
decompressed, meaning all the periods of silence and redundant data are reinserted and is finally decoded to produce an approximation of the original sound. The compression algorithm
used in the process can compress the voice signals and can even carry voice over a little as 5.3 kbps bandwidth.
APPLICATIONS OF VOICE OVER INTERN ET PROTOCOL
Voice communications will certainly remain a basic form of interaction for all of us. The PSTN simply cannot be replaced, or even dramatically changed, in the short term (this may not apply to private voice networks, however). The immediate goal for VoIP service providers is to reproduce existing telephone capabilities at a significantly lower "total cost of operation" and to offer a technically competitive alternative to the PSTN. It is the combination of VoIP with point-of-service applications that shows great promise for the longer term. Some of the example are:
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¢ PSTN gateways: Interconnection of the internet to the PSTN can be accomplished
using a gateway, either integrated into a PBX (the iPBX) or provided as a separate device. A PC-based telephone, for example, would have access to the public network by calling a gateway at a point close to the destination (thereby minimizing long distance charges).
¢ Internet-aware telephones: Ordinary telephones (wired or wireless) can be enhanced to serve as an Internet access device as well as providing normal telephony. Directory services, for example, could be accessed over the Internet by submitting a name and receiving a voice (or text) reply.
¢ Inter-office trunking over the corporate intranet: Replacement of tie trunks between company-owned PBXs using an Intranet link would provide economies of scale and help to consolidate network facilities.
* Remote access from a branch (or home) office: A small office (or a home office) could gain access to corporate voice, data, and facsimile services using the company's Intranet
(emulating a remote extension for a PBX, for example). This may be useful for home-based agents working in a call center, for example.
¢ Voice calls from a mobile PC via the Internet: Calls to the office can be achieved using a multimedia PC that is connected via the Internet. One example would be using the Internet to
call from a hotel instead of using expensive hotel telephones. This could be ideal for submitting or retrieving voice messages.
¦ Internet call center access: Access to call center facilities via the Internet is emerging as a valuable adjunct to electronic commerce applications. Internet call center access would enable a customer who has questions about a product being offered over the Internet to access customer service agents online. Another VoIP
application for call centers is the interconnection of multiple call centers.
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H.323
The H.323 standard provides a foundation for audio, video, and data communications across IP-based networks, including the Internet. H.323 is an umbrella recommendation from the International Telecommunications Union (ITU) that sets standards for multimedia communications over Local Area Networks (LANs) that do not provide a guaranteed Quality of Service (QoS). These networks dominate today's corporate desktops and include packet-switched TCP/IP and IPX over Ethernet, Fast Ethernet and Token Ring network technologies. Therefore, the H.323 standards are important building blocks for a broad new range of collaborative, LAN-based applications for multimedia communications. It includes parts of H.225.0 - RAS, Q.931, H.245 RTP/RTCP and audio/video codecs, such as the audio codecs (G.711, G.723.1, G.728, etc.) and video codecs (H.261, H.263) that compress and decompress media streams.
Media streams are transported on RTP/RTCP. RTP carries the actual media and RTCP carries status and control information. The signalling is transported reliably over TCP. The following protocols deal with signalling:
¢ RAS manages registration, admission, status.
¢ Q.931 manages call setup and termination.
¢ H.245 negotiates channel usage and capabilities.
¢ H.235 security and authentication.
VOICE CODING ALGORITHMS
There are several approaches to digitizing the voice samples. These approaches vary by the information that is transmitted, the complexity of the algorithm, and the assumptions of the sound being transmitted (e.g. voice, fax, music). Broadly classified, the various coding algorithms fall into two broad categories: coding of waveform and modeling of the vocal track.
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Compelling VoIP Applications
The VoIP technology only becomes useful when compelling applications meet the needs of customers. Three such applications are corporations that replace PBX systems, cable operators offering telephony services using their plant, and video conferencing. These existing applications are the driving factors in allowing manufactures to make equipment, service providers to offer services, and customers to increase their productivity. Having an established market and a profitable business model allows VoIP to begin addressing the next generation of applications which then will allow the market to continue to grow.
Corporate
LAN-based telephony applications offer attractive business models to consumers today. The most important applications are the replacement of the
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Compelling VoIP Applications
The VoIP technology only becomes useful when compelling applications meet the needs of customers. Three such applications are corporations that replace PBX systems, cable operators offering telephony services using their plant, and video conferencing. These existing applications are the driving factors in allowing manufactures to make equipment, service providers to offer services, and customers to increase their productivity. Having an established market and a profitable business model allows VoIP to begin addressing the next generation of applications which then will allow the market to continue to grow.
Corporate
LAN-based telephony applications offer attractive business models to consumers today. The most important applications are the replacement of the
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traditional PBX, new client side applications using the PC, and the reduction in maintenance expenses for wiring changes.
LAN based PBX systems provide superior return on investment to traditional PBX systems. Although initial equipment costs are comparable, LAN PBXs typically cost much less than PBXs to install because they use the existing data infrastructure (Category 5 cabling) rather than separate voice wiring. Administration is also less burdensome because LAN and server administrators can manage the system without the need for dedicated telephony technicians.
The most compelling reason businesses consider IP telephony-type applications is for the integration of applications with voice. Over the years, a significant amount of work has gone into computer telephony integration (CTI) in traditional PBXs. These systems began to offer application-programming interfaces such as Telephony API (TAPI), Telephony Services API (TSAPI), and Java Telephony API (JTAPI). This work has resulted in advanced call center functions, including screen pops for agents and active call routing between call centers.
Client (end station) products offer LAN telephony services through the use of a software client on the user's PC, while others offer telephone instruments that plug into the LAN. When there is a need to relocate the equipment, it needs only to be unplugged from one data port and plugged in at the new destination. This process avoids the wiring changes typically done for convention phones and thus VoIP clients reduce the cost of ownership. Over time, these savings add up to the point that a LAN-based telephony system can offer considerable savings over traditional PBXs.
PacketCable
PacketCable„¢ is the telephony architecture for using VoIP over the Cable TV system. With buried Cable TV plant passing hundreds of millions of homes .-.orldwide, it is logical to assume that cable operators desire to offer new services that make use of their installed system.
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The PacketCable standards (1.0, 1.1 and 1.2) were developed by CableLabs®, the research group of the cable operators and makes use of the same equipment used for the cable modem services. The cable modem architecture, known as DOCSIS (Data over Cable Service Interface Specification), is able to meet the requirements of transporting VoIP packets. DOCSIS version 1.1 provides Quality of Service (QoS), security features, and the prioritization of packet traffic that is necessary for voice communication.
The PacketCable specification also incorporates the Network-based Call Signaling (NCS) protocol for signaling voice calls over cable networks. NCS leverages the existing Media Gateway Control Protocol (MGCP) and the protocol is sometimes referred to as MGCP NCS. NCS uses network-based call agents to negotiate cable-based IP telephony calls.
Traditional telephones draw all the power they need from the phone lines. Part of the reason that the public phone system has evolved to such a reliable state, is that it is essentially immune from the effects of power outages. Electrical utilities in most areas do not offer this degree of unfailing reliability and this reality is an important issue for the cable television plant.
Normally, head-end and customer-premises cable equipment rely solely on the ocal electric company for their power and this puts users at risk of losing phone service should a power outage occur. There is not universal agreement among the :abfe operators about offering "lifeline" telephony service.
Some Time Warner systems focus their telephony service towards the "second" phone line in the home. They suggest that the customer keep the existing primary line and then add the Time Warner telephony service for a business line, fax line, ;r children's phone. AT&T has taken a different approach by enhancing many of their networks with alternate power sources thereby allowing lifeline telephony services to their customers.
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Video Conferencing
Video conferencing includes the use of packetized voice and is an important application for the home and business. There are several video conferencing standards and one of the significant standards is known as H.32x where x can be 0, 1, 2, 3, or 4 representing video conferencing on different types of communications links. The H.32x model includes establishing connections between the source and destination devices and this method connection establishment has been adapted for use in VoIP gateways.
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Table 1. The ITU-T H.32x Video Conferencing Standards
H.320 H.321 H.322 H.323 H.324
Approval Date 1990 1995 1995 1996 1996
Network Narrowban d switched digital ISDN Broadband ISDN ATM LAN QoS packet switched networks Non-QoS networks
(Ethernet) The analog
phone
system
G.711
G.711 G.711 G.711 G.722
Audio G.722 G.722 G.722 G.728 G.723
G.728 G.728 G.728 G.723 G.729
Video H.261 H.261/H.26 3 H.261/H.26
3 H.261/H.26 3 H.261/H.26 3
Control H.230/H.24
2 H.242 H.230/H.24 2 H.245 H.245
Multiplex in
g H.221 H.221 H.221 H.225 H 213
AAL
Comm. Interface 1.400 1.363
AJM 1.361 PHY 1.400 I.400& TCP/IP TCP/IP V.34 Modem
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The H.323 recommendation supports the largest number of audio coding standards. In fact, the audio codecs of the other recommendations are a subset of the audio codecs outlined in the H.323 recommendation. In addition, the H.323 is the recommendation that is made for a non-guaranteed bandwidth, packet switched networks such as the Internet.
H.323 Video Conferencing
The H.323 recommendation covers the technical requirements for audio and video communications services in LANs that do not provide a guaranteed Quality of Service (QoS). The scope of H.323 does not include the LAN itself or the transport layer that may be used to connect various LANs. Only elements needed for interaction with the Switched Circuit Network (SCN) are within the scope of H.323.
H.323 defines four major components for a network-based communications system: Terminals, Gateways, Gatekeepers, and Multipoint Control Units (MCUs). The terminal must support voice transmission with data and video transfer being an option.
Figure 1. The H.323 Protocol Stack
H.323 uses the following standards as part of the video conferencing protocol stack (Figure 1):
¦ Q.931 QoS - Quality of Service. When the connection between devices is being established, the end stations and each data link in the route
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negotiate to determine what bandwidth is available, how much delay the
application can tolerate, and how much jitter there will be in the packet arrival. Once the links agree to the QoS message, they must guarantee those parameters for the duration of the video conferencing session.
¦ H.245 - Control Channel Protocol. Provides capability negotiation between
the two end-points such as voice compression algorithm to use, conferencing requests, etc. The H.245 channels transport must be reliable (e.g. TCP, SPX)
¦ H.225 RAS - Registration, Admission, and Status (RAS) Protocol. Used to convey the registration, admissions, bandwidth change and status messages between IP Telephone devices and servers called Gatekeepers that provide address translation and access control to devices.
¦ RTCP - Real-time Transport Control Protocol (RTCP). Provides statistics
information for monitoring the quality of service of the voice call. Control messages (Q.931 signaling, H.245 capability exchange and the RAS protocol) are carried over the reliable TCP layer. This ensures that important messages get retransmitted if necessary so they can make it to the other side. Media traffic is transported over the unreliable UDP layer and includes two protocols: RTP (Real-Time Protocol) that carries the actual media and RTCP (Real-Time Control Protocol) that includes periodic status and control messages. Audio and Video information is carried over UDP because it need not be retransmitted because if a sound packet is
lost and then transmitted, it would most probably arrive too late to be used for the real-time conferencing.
GATEWAY PROTOCOLS
H.323
The most widely embraced standard is the ITU-T's H.323 umbrella standard. ~: z rally designed as an end to end communications standard for video conferencing over packet networks. H.323 was adapted for VoIP applications. The result was a r= : = 'd that defines far more functionality than is necessary for most VoIP
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environments, its complexity is hard to implement efficiently and causes problems in interoperably since there are various ways to interpret the standard.
MGCP (MEDIA GATEWAY CONTROL PROTOCOL)
MGCP is a protocol that addresses control of media gateways, but it does not, as H.323 does, specify a complete end-to-end communication. MGCP uses simple endpoints called Media gateways (MGs). An intelligent media gateway controller (MGC) or call agent (CA) provides services. The endpoints provides user interactions and interfaces, while the MGC provides centralized call intelligence.
A master/slave relationship is presented at all times between the MGC and the MGs. Typically MGCP message are sent over IP/UDP between the MG and MGC. The media connection itself is usually over IP/RTP. For security, MGCP uses IP sec to protect the signaling information.
SIP (SESSION INITIATION PROTOCOL)
SIP is an application layer signaling protocol that specifies call control for multiparty sessions, IP phone calls or multimedia distribution. Unlike H.323, which is based on binary encoding, SIP is a text based protocol that is much easier to implement. Much like H.323, SIP is a peer-to-peer architecture ( vs, master/slave for MGCP).
SIP depends on relatively intelligent endpoints, which require littfe or no interaction with servers. Each endpoints manages its own signaling both to the user and to other endpoints. SIP is more scalable than H.323 because it Is inherently a distributed and stateless call model.
Perhaps the key advantage of SIP is that it is an Internet-mode! protocol from inception. It uses simple ASCII messaging based on HTTP/1.1. This means that SIP messaging is easy to decode and troubleshoot.
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SIP ARCHITECTURE
¢ SIP teriininal: Supports real-time, 2 way communication with another SIP entity. Supports both signaling and media, similar to H.323 terminal.
¦ Proxy: Contacts one or more clients or next-hop servers and passes the call request further.
¢ Redirect Server: Accepts SIP requests, maps the address into zero or
more new addresses and returns those addresses to the client. Does not initiate SIP requests or accept calls.
¦ Location Server: Provides information about a callers possible locations to redirect and proxy servers. May be co-located with a SIP server.
THE IMPORTANCE OF VOICE OVER IP
Of the key emerging technologies for data, voice, and video integration, voice over IP (Internet Protocol) is arguably very important. The most quality of service (QoS) sensitive of all traffic, voice is the true test of the engineering and quality of a network. Demand for Voice over IP is leading the movement for QoS in IP environments, and will ultimately lead to use of the Internet for fax, voice telephony, and video telephony services. Voice over IP will ultimately be a key component of the migration of telephony to the LAN infrastructure.
Significant advances in technology have been made over the past few years that enable the transmission of voice traffic over traditional public networks such as Frame Relay (Voice over Frame Relay) as well as Voice over the Internet through the efforts of the Voice over IP Forum and the Internet Engineering Task Force (IETF). Additionally, the support of Asynchronous Transfer Mode (ATM) for different traffic types and the ATM Forum's recent completion of the Voice and Telephony over ATM specification will quicken the availability of industry-standard solutions.
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MGCP COMMANDS
¢Endpoint Configuration (EPCF): Specify coding
¢Notification Request (RQNT): Watch for event
¢Notify (NTFY): Used by gateway to inform Call agent
¢Create Connection (CRCX)
¢Modify Connection (MDCX)
¢Delete Connection (DLCX)
¢Audit Endpoint (AUEP): Give me status
¢Audit Connection (AUCX)
¢Restart in Progress (RSIP): Used by gateway to indicate initialization/shutdown of end points/gateway
BENEFITS OF VOIP
Widespread deployment of a new technology seldom occurs without a clear and sustainable justification, and this is also the case with VoIP. Demonstrable benefits to end users are also needed if VoIP products (and services) are to be a long-term success. Generally, the benefits of technology can be divided into the following four categories:
¢ Cost Reduction: Although reducing long distance telephone costs is always a popular topic and would provide a good reason for introducing VoIP, the actual savings over the long term are still a subject of debate in the industry. Flat rate pricing is available with the
Internet and can result in considerable savings for both voice and facsimile. It has been estimated that up to 70% of all calls to Asia are to send faxes, most of which could be replaced by FolP.
¢ Simplification: An integrated infrastructure that supports all forms of communication
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allows more standardization and reduces the total
equipment complement. This combined infrastructure can support dynamic bandwidth optimization and a fault tolerant design.
¢ Consolidation: Since people are among the most significant cost elements in a network, any opportunity to combine operations, to eliminate points of failure, and to consolidate accounting systems would be beneficial. In the enterprise, SNMP-based management can be provided for both voice and data services using VoIP.
* Advanced Applications: Even though basic telephony and facsimile are the initial applications for VoIP, the longer term benefits are expected to be derived from multimedia and multiservice applications. For example, Internet commerce solutions can combine WWW access to information with a voice call button that allows immediate access to a call center agent from the PC.
ADVANTAGES OF IP TELEPHONY
Ubiquity
The IP standard is by far the world's most popular network protocol. It is developing fast and is accepted by every major vendor. Users can benefit from end-to-end connectivity to every data-networking device available, a tremendous amount of research from firms focused on IP, and a unique global addressing scheme which allows an IP device to address the entire network, regardless of size or location.
With more than 80 per cent of personal computers in business networks on a local area network (LAN), IP is an obvious transport of choice. It also seems to make sense to carry all forms of communication (data, voice, and video) over a common, ubiquitous medium.
Value Added Applications
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Once an IP infrastructure is in place there is a seemingly unending number of add value applications that are available or being designed that are either far simpler than those enabled by regular telephony systems, or totally new applications like user registration that allows phone users to be identifiable and fully functional no matter which handset they're using.
Cost
For most large businesses, overhauling a PBX-based telecom infrastructure is a long-term investment decision. However, with telephony requirements becoming more demanding and new technology creating business advantage in shorter timeframes, IP telephony has become a more and more realistic alternative. And while cost project and implimentationions will inevitably vary enormously from one company to another, most equipment vendors and IP telephony integration specialists would claim return on investment within one or two years.
Single Infrastructure
Putting the voice and data traffic on one set of wires instead of two also seems to make good commercial sense when compared to the cost of developing and supporting two separate infrastructures.
Overhead
A large part of that cost is a human one - and again there is a clear case for creating one pool of skills rather than two, so that voice calls essentially become just another application running on the network. In reality the network will only see information packets, some of this information originates and terminates within a data edge device whilst other packets of information on the network will originate and terminate in voice edge devices.
Call centre
Several advantages arise from a Call Centre environment based on an IP infrastructure. Web based forms, text chat sessions, email, and web collaboration are
all examples of modes of communication, which are based on the IP standard. Voice
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is transformed to the IP standard in IP call centres. An IP infrastructure therefore facilitates integration of both web-based media and voice into the call centre.
IP also puts companies in a better position to take advantage of future developments in terms of new channels because once an IP 'skeleton' is in place, functionality is just a question of what software is developed.
Department (J] Computer Science ii hiiginecrmv
ST7UUE
MAIN JUSTIFICATIONS FOR DEVELOPM -ENT OF VOIP
¢Cost Reduction. ¦Simplification. ¢Consolidation. ¢Advanced Applications.
FUTURE TRENDS
¢Price is the key driver of the VoIP market today. End-user features such as multimedia conferencing, multicast, call centers, IP call waiting, and message unification are the benefits that will drive the VoIP market well into the future.
¢The growing competition between ISPs is causing declining margins. ISPs are seeking value-added services to increase revenues per subscriber. Becoming an ITSP is the solution.
¢The demand for convergent networks is evolving into a requirement for new network/telephone orders and upgrades.
SUMMARY
1. Voice over IP products and services are being rolled out.
2. Ideal for computer-based communications.
3. IP needs QoS for acceptable quality.
4. A number of working group at IETF are working on it.
5. H.323 provides interoperability.
Department Oj Computer Science & tngmeenng
SMJCE
References
1. cis.ohio-state.edu/~iain/refs/ref voip.htm
2. protocolsvoip/testinQ.hima mi ^3 ¦
3. httpV/webprofoaimvfoip/inde^.htrflfUnt-*
CONTENTS
1. Introduction 1
2. Voice Over Internet Protocol 3
3. Scenarios in Internet Telephony 4
i. PC to PC 5
ii. PC to Phone 5
iii. Phone to Phone 6
4. Working 7
5. Applications of Voice over Internet Protocol 7
6. Voice Coding Algorithms 9
7. Coding Standards 10
8. Compelling VoIP Applications 11
i. Corporate 11
ii. PacketCable 12
iii. Video Conferencing 14
iv. H.323 Video Conferencing 15
9. Gateway Protocols 16
i. H.323 16
ii. MGCP (Media Gateway Control Protocol) 17
Hi. SIP (Session Initiation Protocol) 17
10. SIP Architecture 18
11. The Importance of Voice over IP 18
12. MGCP Commands 19
13. Benefits of VoIP 19
14. Advantages of IP Telephony 20
15. Main Justifications for development of VoIP 23
16. Future Trends 23
17. Summary 23
18. References 24

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ABSTRACT
Traditionally,voice and data were carried over separate networks optimized to suit the differing characteristics of voice and data traffic.with advances in technologyjt is now possible to carry voice and data over the same networks. . VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems.
Today,phone networks are on a migration path to VoIP.Most traditional phone networks use the Public Switched Telephone Network(PSTN),this system employs circuit-switched technology that requires a dedicated voice channel to be assigned to each particular coversation.Messages are sent in analog format over this network.A VoIP system employs a packet-switched betwork where the voice signal is digitized,compressed and packetized.This compressed digital message no longer reqires a voice channel.Instead,a message can be sent across the same data lines that are used for the Internet and a dedicated channel is no longer needed. The message can now share the bandwidth with other messages in the network.
1 .INTRODUCTION
Voice over Internet Protocol, also called VoIP, IP Telephony, Internet telephony, Broadband telephony, Broadband Phone and Voice over Broadband is the routing of voice conversations over the Internet or through any other IP-based network.Protocols which are used to carry voice signals over the IP network are commonly referred to as Voice over IP or VoIP protocols.This technology offers cost savings by making more efficient use of the existing network.
Traditionally,voice and data were carried over separate networks optimized to suit the differing characteristics of voice and data traffic.with advances in technology,it is now possible to carry voice and data over the same networks. Some services using VoIP may only allow you to call other people using the same service, but others may allow you to call anyone who has a telephone number - including local, long distance, mobile, and international numbers. Also, while some services only work over your computer or a special VoIP phone, other services allow you to use a traditional phone through an adaptor.

Fig: IP Telephony
The IP network could be
¢ A LAN in an office
¢ A WAN linking the sites of a large international organization
¢ A corparate intranet
¢ The internet
¢ Any combination of the above
.Today,phone networks are on a migration path to VoIP.Most traditional phone networks use the Public Switched Telephone Network(PSTN),this system employs circuit-switched technology that requires a dedicated voice channel to be assigned to each particular coversation.Messages are sent in analog format over this network.A VoIP system employs a packet-switched betwork where the voice signal is digitized,compressed and packetized.This compressed digital message no longer reqires a voice channel.Instead,a message can be sent across the same data lines that are used for the Internet and a dedicated channel is no longer needed.The message can now share the bandwidth with other messages in the network.
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, you are bypassing the phone company (and its charges) entirely. VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. A broadband (high speed Internet) connection is required. This can be through a cable modem, or high speed services such as DSL or a local area network. You can hook up an inexpensive microphone to your computer and send your voice through a cable modem or connect a phone directly to a telephone adaptor.
Some VoIP providers offer their services for free, normally only for calls to other subscribers to the service. Your VoIP provider may permit you to select an area code different from the area in which you live. It also means that people who call you may incur long distance charges depending on their area code and service.Some VoIP providers charge for a long distance call to a number outside your calling area, similar to existing, traditional wireline telephone service. Other VoIP providers permit you to call anywhere at a flat rate for a fixed number of minutes.
Depending upon your service, you might be limited only to other subscribers to the service, or you may be able to call any phone number, anywhere in the world. The call can be made to a local number, a mobile phone, to a long distance number, or an international
number. You may even utilize the service to speak with more than one person at a time. The person you are calling does not need any special equipment, just a phone.
Most VoIP companies are offering minute-rate plans structured like cell phone bills for as little as $30 per month. On the higher end, some offer unlimited plans for $79. With the elimination of unregulated charges and the suite of free features that are included with these plans, it can be quite a savings.
Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan. VoIP includes:
. Caller ID
¢ Call waiting
¢ Call transfer
¢ Repeat dial
¢ Return call
¢ Three-way calling
There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled. You can:
¢ Forward the call to a particular number
¢ Send the call directly to voicemail
¢ Give the caller a busy signal
¢ Play a "not-in-service" message
¢ Send the caller to a funny rejection hotline
2. FLAVORS OF VoIP
The interesting thing about VoIP is that there is not just one way to place a call. There are three different "flavors" of VoIP service in common use today:
¢ ATA - The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you're ready to make VoIP calls
¢ IP Phones - These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Soon, Wi-Fi IP phones will be available, allowing subscribing callers to make VoIP calls from any Wi-Fi hot spot.
¢ Computer-to-computer - This is certainly the easiest way to use VoIP. You don't even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the software, a microphone, speakers, a sound card and an Internet connection , preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.
3.THE STANDARD PHONE SYSTEM VS. THE VOIP SYSTEM
Existing phone systems are driven by a very reliable but somewhat inefficient method for connecting calls called circuit switching. Circuit switching is a very basic concept that has been used by telephone networks for more than 100 years. When a call is made between two parties, the connection is maintained for the duration of the call. Because you are connecting two points in both directions, the connection is called a circuit. This is the foundation of the Public Switched Telephone Network (PSTN).
Here's how a typical telephone call works:
1. You pick up the receiver and listen for a dial tone. This lets you know that you have a connection to the local office of your telephone carrier.
2. You dial the number of the party you wish to talk to.
3. The call is routed through the switch at your local carrier to the party you are calling.
4. A connection is made between your telephone and the other party's line using several interconnected switches along the way.
5. The phone at the other end rings, and someone answers the call.
6. The connection opens the circuit.
7. -You talk for a period of time and then hang up the receiver.
8. When you hang up, the circuit is closed, freeing your line and all the lines in between.
4.HOW VoIP WORKS?
VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes.
How packet switching works.
While circuit switching keeps the connection open and constant, packet switching opens a brief connection ” just long enough to send a small chunk of data, called a packet, from one system to another. It works like this:
¢ The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.
¢ Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet.
¢ The sending computer sends the packet to a nearby router and forgets about it. The nearby router send the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.
¢ When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.
Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well.

Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched network:
1. You pick up the receiver, which sends a signal to the ATA.
2. The ATA receives the signal and sends a dial tone. This lets you know that you have a connection to the Internet.
3. You dial the phone number of the party you wish to talk to. The tones are converted by the ATA into digital data and temporarily stored.
4. The phone number data is sent in the form of a request to your VoIP company's call processor. The central call processor is a piece of hardware running a specialized database/mapping program called a soft switch .The call processor checks it to ensure that it is in a valid format.
5. The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend's ATA, telling it to ask the connected phone to ring.
6. Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.
7. You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.
8. You finish talking and hang up the receiver.
9. When you hang up, the circuit is closed between your phone and the ATA.
10. The ATA sends a signal to the soft switch connecting the call, terminating the
session. Probably one of the most compelling advantages of packet switching is that
data networks already understand the technology. By migrating to this technology,
¢telephone networks immediately gain the ability to communicate the way
computers do.
5. PROTOCOLS
As we've seen, on each end of a VoIP call we can have any combination of an analog, soft or IP phone as acting as a user interface, ATAs or client software working with a codec to handle the digital-to-analog conversion, and soft switches mapping the calls. So how do you get all of these completely different pieces of hardware and software to communicate efficiently to pull all of this off? The answer is protocols.
There are several protocols currently used for VoIP. These protocols define ways in which devices like codecs connect to each other and to the network using VoIP. They also include specifications for audio codecs. The most widely used protocol is H.323, a standard created by the International Telecommunication Union (ITU). H.323 is a comprehensive and very complex protocol that was originally designed for video conferencing. It provides specifications for real-time, interactive videoconferencing, data sharing and audio applications such as VoIP. Actually a suite of protocols, H.323 incorporates many individual protocols that have been developed for specific applications.

H.323 Protocol Suite
Video Audio Data Transport
H.261 G.711 T.122 H.225
H.263 G.722 T.124 H.235
G.723.1 T.125 H.245
G.728 T.126 H.450.1
G.729 T.127 H.450.2 H.450.3
RTP X.224.0
As you can see, H.323 is quite a large collection of protocols and specifications. That's what allows it to be used for so many applications. The problem with H.323 is that it is not specifically tailored to VoIP.
An alternative to H.323 emerged with the development of Session Initiation Protocol (SIP). SIP is a much more streamlined protocol, developed specifically for VoIP applications. Smaller and more efficient than H.323, SIP takes advantage of existing protocols to handle certain parts of the process. Media Gateway Control Protocol (MGCP) is a third commonly used VoIP protocol that focuses on endpoint control. MGCP is geared toward features like call waitingOne of the challenges facing the worldwide use of VoIP is that these three protocols are not always compatible. VoIP calls going between several networks may run into a snag if they hit conflicting protocols. Since VoIP is a relatively new technology, this compatibility issue will continue to be a problem until a governing body creates a standard universal protocol for VoIP.
The overall hurdle facing VoIP is that there are currently no overriding standards. This includes hardware, protocols and virtually every aspect of the system. In the end. VoIP is a vast improvement over the current phone system in terms of efficiency, cost and flexibility. Like any emerging technology, VoIP has some challenges to overcome, but it is clear that developers will keep refining this technology until it eventually replaces the current phone system.
6. ADVANTAGES
Cost
In general, phone service via VoIP is free or costs less than equivalent service from traditional sources but similar to alternative Public Switched Telephone Network (PSTN) service providers. Some cost savings are due to utilizing a single network to carry voice and data, especially where users have existing underutilized network capacity they can use for VoIP at no additional cost. VoIP to VoIP phone calls on any provider are typically free, whilst VoIP to PSTN calls generally costs the VoIP user.
Functionality
VoIP can facilitate tasks that may be more difficult to achieve using traditional phone networks:
¢ Incoming phone calls can be automatically routed to your VoIP phone, regardless of where you are connected to the network. Take your VoIP phone with you on a trip, and wherever you connect to the Internet, you can receive incoming calls.
¢ Free phone numbers for use with VoIP are available in the USA, UK and other countries from organizations such as VoIP User.
¢ Call center agents using VoIP phones can work from anywhere with a sufficiently fast and stable Internet connection.
¢ Many VoIP packages include PSTN features that most telcos normally charge extra for, or may be unavailable from your local telco, such as 3-way calling, call forwarding, automatic redial, and caller ID.
Mobility
VoIP allows users to travel anywhere in the world and still make and receive phone calls:
¢ Subscribers of phone-line replacement services can make and receive local phone calls regardless of their location. For example, if a user has a New York City phone number and is traveling in Europe and someone calls the phone number, it will ring in Europe. Conversely, if a call is made from Europe to New York City, it will be

treated as a local call. Of course, there must be a connection to the Internet e.g. WiFi to make all of this possible.
Users of Instant Messenger based VoIP services can also travel anywhere in the world and make and receive phone calls.
VoIP phones can integrate with other services available over the Internet, including video conversation, message or data file exchange in parallel with the conversation, audio conferencing, managing address books and passing information about whether others (e.g. friends or colleagues) are available online to interested parties.
7. DISADVANTAGES
VoIP technology still has a few shortcomings that have led some to believe that it is not ready for widespread deployment. However, many industry analysts predicted that 2005 was the "Year of Inflection," where more IP PBX ports shipped than conventional digital PBX ports. This date has been moved on an annual basis and only now (mid 2006) is it beginning to happen. However, many purchasers of VOIP ports just want a phone, so the statistics can be misleading when interpreted by marketeers.
Faxes
One drawback is the difficulty in sending faxes due to software and networking restraints in most home systems. However, an effort is underway to define an alternate IP-based solution for delivering Fax-over-IP, namely the T.38 protocol. Another possible solution to overcome the drawback is to treat the fax system as message switching system which does not need real time data transmission. The end system can buffer completely the incoming fax data before displaying or printing the fax image.
Internet Connection
Another drawback of VOIP service is its reliance upon another separate service - an internet connection. The quality and overall reliability of the phone connection is entirely reliant upon the quality, reliability, and speed of the internet connection which it is using. Shortcomings with internet connections and Internet Service Providers (ISPs) can cause a lot of grief with VOIP calls. Higher overall network latencies can lead to significantly reduced call quality and cause certain problems such as echoing.
A few business VOIP Providers (like Unity Business Networks) overcome this challenge using dedicated connections (point to point Tls) between a client location and the VOIP Providers gateway facility where VOIP is converted back to traditional local phone service.
Using a point to point connection with specialized routers which prioritize packets, the highest quality of service can be achieved.
Many VOIP users still maintain a traditonal analog voice line (business line) which allows them to utilize a traditional fax machine when needed and can also be used to call 911 service if you have an analog phone on the fax machine as well. .
Power Outages
Another drawback of VOIP is the inability to make phone calls during a power outage, but this problem also exists with many phones used with conventional land lines and can be remedied with a battery backup. During a power outage you also have the choice to forward your phone to your cell phone or another phone number so you would still be able to receive calls. Although you can't call out on your home phone system during a power outage, at least you can still receive calls.
If VoIP is used in solitary LAN (with no internet connection), it would consume more resources compared to a PABX.Modems are now Available with lithium ion battery backup so that you can use the service with no power.
Phone Jacks
With VoIP you must set a telephone near your DSL connection, or rewire your telephone jack(s) to accommodate VoIP standards. This will limit the number of telephones you can use.
8. IMPLEMENTATION CHALLENGES
Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round trip propagation delay (400 milliseconds to 600 milliseconds for geostationary satellite). The receiving node must restructure IP packets that may be out of order, delayed or missing, while ensuring that the audio stream maintains a proper time consistency. This functionality is usually accomplished by means of a jitter buffer.
Another challenge is routing VoIP traffic through firewalls and address translators. Private Session Border Controllers are used along with firewalls to enable VoIP calls to and from a protected enterprise network. Skype uses a proprietary protocol to route calls through other Skype peers on the network, allowing it to traverse symmetric NATs and firewalls. Other methods to traverse firewalls involve using protocols such as STUN or ICE.
VOIP challenges:
¢ Delay/Network Latency
¢ Packet loss
¢ Jitter
¢ Echo
¢ Security
Fixed delays cannot be controlled but some delays can be minimized by marking voice packets as being delay-sensitive.
The principal cause of packet loss is congestion, which can be controlled by congestion management and avoidance. Carrier VoIP networks avoid congestion by means of traffic engineering.
Variation in delay is called jitter. The effects of jitter can be mitigated by storing voice packets in a buffer (called a play-out buffer) upon arrival, before playing them out. This avoids a condition known as buffer underrun, in which the playout process runs out of voice data to play because the next voice packet has not yet arrived, but increases delay by the length of the buffer.
Common causes of echo include impedance mismatches in analog circuitry, and acoustic coupling of the transmit and receive signal at the receiving end.
9.CONCLUSION
The international long distance market in India was opened to competition on April 1, 2004. This brings forth a whole range of opportunities in the technology of Voice over Internet Protocol.Many industry analysts predicted that 2005 was the "Year of Inflection," where more IP PBX ports shipped than conventional digital PBX ports. This date has been moved on an annual basis and only now (mid 2006) is it beginning to happen. However, many purchasers of VOIP ports just want a phone, so the statistics can be misleading when interpreted by marketeers. Because UDP does not provide a mechanism to ensure that data packets are delivered in sequential order, or provide Quality of Service guarantees, VoIP implementations face problems dealing with latency and jitter. This is especially true when satellite circuits are involved, due to long round trip propagation delay
10. FUTURE TRENDS
In the not too distant past many businesses were using typewriters instead of PCs. Today one would struggle to find a business that is not using PCs. Businesses rely on the vast array of applications that have been developed for PCs. These applications help businesses accomplish their goals, increase productivity, and provide a competitive advantage. Voice over IP (VoIP) based communications systems are replacing all of the old TDM based PBX systems in enterprises today for the same reason that PCs replaced typewriters. Businesses require a communications system that will increase productivity, reduce costs, and provide them with a competitive advantage.
Moving further into the 21st Century, more suppliers will base their products on open standards so that enterprises can have flexibility and choice when deploying their communications solution. Just as new applications are continually developed for the PC, new applications will continue to appear for IP based communications systems. The workforce is going to become increasingly mobile and applications for multimedia wireless communications devices will become increasingly abundant. People within the enterprise
are going to be reachable anywhere by using a single address.

11.BIBLIOGRAPHY
1. en.Wikipedia.com
2. seminar and presentationtopics.com
3. seminar and presentations4u.com
4. Howstuffworks.com
5. Computer Networks By Andrew S.Tanenbaum
CONTENTS
1) INTRODUCTION 1
2) FLAVORS OF VOIP 4
3) THE STANDARD PHONE SYSTEM VS. THE VOIP SYSTEM 5
4) HOW VOIP WORKS? 6
5) PROTOCOLS 9
6) ADVANTAGES 11
7) DISADVANTAGES 13
8) IMPLEMENTATION CHALLENGES 15
9) CONCLUSION 17
10) FUTURE TRENDS 18
11) BIBLIOGRAPHY 19
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Voice Over Internet Protocol


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INTRODUCTION

Voice over internet protocol is one of the modern communication methods that has crept into the world of telecommunication. This makes transfer of voice calls over the internet framework, thereby, effectively reducing the cost of calls and also reducing/avioding the need for new infra-structures to be built for expanding the telecommunications, which would be highly essential in case of telephones.
VOIP works by converting our standard voice(analog) signals into digital signals and putting them into packets and sending them over the internet. This way, any person who has a reasonably good internet connection, and the right hardware, or the software that has to be installed into the computer, can use it to make phone calls over the internet to any other similar VOIP user.
This report gives an overview into the technical details of VoIP, the concerns raised, and also the basic hardware and software requirements.


VoIP COMMUNICATION SYSTEM

The VoIP communication system is deployed on the existing Internet framework to provide communication between end systems. The basic technology that is involved in the VoIP communication are
• ADC, which is used to convert analog voice to digital signals (bits)
• Compression, used to reduce the band-width necessary
• Transmission, where the compressed data packets are embedded into IP packets and sent over the internet
• Signaling protocol, which is used to set-up the communication links
• DAC, which should be present at the receiver’s side to decode the incoming digital signal back to analog voice
These steps are not as easy. The main difficulty lies in the fact that these things have to be done at real time speed, which is difficult with the current architecture of the internet domain.


ADC
ADC stands for Analog-to-Digital Conversion. This is done at the senders end so that the input voice signal can be digitized to perform various operations on them, such as compression, quality improvement, etc.

Compression
This is one of the main steps in VoIP. It is used to reduce the pay-load of the IP datagram packets, ie, to make the datagram packet to carry more data. This helps in improving the speed of transmission of the data. One of the most commonly used compression technique is the PCM G.711 compression algorithm.



RTCP
RTCP stands for Realtime Transport Control Protocol. This is used for ontrolling the data flow in RTP. The RTCP packets are similar to the data packets of the RTP. These packets are split into different sections, as given below.
• Sender Report - Byte send, estimate rate, time stamp, ...
• Receiver Report - number of packets sent and expected, loss, round trip delay, ...
• Source Description - name, email, location, ...
• Explicit Leave - used in addition to Timeout
• Extensions(application specific and optional)
The RTCP sender report gives the user the details about identity of the source of data, the
sending time-stamp and also the sender’s packet counts. This will be followed by one or more receiver reports. The receiver report will contain the fields used to identify who’s being reported on, the fraction of packet lost, the total number of packet lost, the highest sequence number received, inter arrival jitter, time since last source report and also the delay since last source report.


Other Signaling Protocols
SCTP, TALI, MGCP, and SCCP are other protocols that perform signaling functions on a VoIP network. It is important to note thatmultiple signaling protocols can exist in some portions of the network.



CONCLUSION
In this report, I have discussed about the factors that play an important role in the use of VoIP as a major communication channel. When we consider VoIP applications in general, they will probably become more widely used in the future. Currently, the main problem for such applications is the lack of QoS guarantees. When QoS supporting protocols like RSVP are deployed on a larger scale, this will certainly make VoIP more popular since people can then communicate with the quality that they desire. The use of VoIP as a telephony alternative can reduce costs. Since voice and data traffic can be integrated, the necessary infrastructure to provide both services is reduced. With the advent of Wi-Fi access, and a major development in that field will help the use of VoIP Wi-Fi phones, just the way a person is now using him mobile phone, with the change that his account will be internationally valid. These changes can be expected in the near future with the advancement in technology growing at faster rates.


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Voice Over Internet Protocol



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what is voip

Voice Over Internet protocol (VoIP) is a protocol used for the transmission of voice and fax phone calls through the Internet or other packet-switched networks.

It is also called IP telephony, Internet telephony, voice over broadband, broadband telephony.



INTERNET


It uses packet switching.
No dedicated path between sender and receiver.

It acquires and releases bandwidth, as it is needed.
Cost is not dependent on time and distance.

application

Real time applications:
Telephony ( two-way)
Radio-TV Broadcast (one-way)
Non-real time applications:
e_mail
OTHERS:
Integration of data,voice and fax
Video telephony
Enhanced teleconferencing

conclusion

VoIP is predominately used for personal instead of enterprise-wide use.

The availability of high-quality audio using wideband codecs, video conferencing, and document sharing enables more effective and pleasant communication.





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