Voice Over Internet Protocol
Content
Voice Over Internet Protocol
Rinzam A
Department of Computer Science and Engineering M G College of Engineering
[email protected]
Abstract— The Voice-Over Internet Protocol (VoIP) technology allows the voice information to pass over IP data networks. This technology results in huge savings on the amount of physical resources required to communicate by voice over long distance. It does so by exchanging the information in packets over a data network. The basic functions performed by a VoIP include – signalling, data basing, call connect and disconnect, and coding/decoding. The steps involved in originating and internet telephone call are the conversion of the analogue voice signal to digital format and compression/translation of the signal into internet protocol (IP) packets for transmission over the internet; the process is reversed at the receiving end. VoIP software’s like Vocal TEC or Net 2 Phone are available for the user. With the exception of phone to phone, the user must posses an array of equipment which should at minimum include VoIP software, an internet connection, and a multimedia computer with a sound card, speakers, a microphone and a modem.
Fig.1 How VOIP works
The VoIP network acts as a gateway to the existing PSTN network. This gateway forms the interface for transportation of the voice content over the IP networks. Gateways are responsible for all call origination, call detection, analogue to digital conversion of voice, and creation of voice packets.
II. BASIC FLOW OF VOIP NETWORK The VoIP networks replace the traditional public-switched telephone networks (PSTNs), as these can perform the same functions as the PSTN networks. The functions performed include signaling, data basing, call connect and disconnect, and coding-decoding. A. Signaling Signaling in a VoIP network is accomplished by the exchange of IP datagram messages between the components. The format of these messages is covered by the standard datalink layer protocols. B. Database services
I. INTRODUCTION The development of very fast, inexpensive microprocessors and special-purpose switching chips, coupled with highly reliable fibre-optic transmission systems, has made it possible to build economical, ubiquitous, high-speed packet-based data networks. Similarly, the development of very fast, inexpensive digital signal processors (DSPs) has made it practical to digitise and compress voice and fax signals into data packets. The natural evolution of these two developments is to combine digitised voice and fax packets with packet data, creating integrated data-voice networks. The voice-overInternet protocol (VoIP) technology allows voice information to pass over IP data networks. Primarily, the cost savings that accrue from operating a single, shared network have motivated this convergence of telecommunications and data communications
Database services are a way to locate an endpoint and translate the addressing that two networks use; for example, the PSTN uses phone numbers to identify endpoints, while a VoIP network could use an IP address and port numbers to identify an endpoint. A call control database contains these mappings and translations. C. Calls connect and disconnect (bearer control) The connection of a call is made by two endpoints opening communication sessions between each other. In the PSTN, the public (or private) switch connects logical channels through the network to complete the calls. In a VoIP implementation, a multimedia stream (audio, video, or both) is transported in real time. The connection path is the bearer channel and represents the voice or video content being delivered. When communication is complete, the IP sessions are released and, optionally, network resources are freed. D. CODEC operations Voice communication is analogue, while data networking is digital. Analogue waveforms are converted into digital information by using a coder-decoder (CODEC).
lookup, and signaling. The extent of gateway functionalities is based on the VoIP-enabling products used. Fig. 1 shows the architecture of a typical gateway. The DSP in a gateway is responsible for signal processing functions such as analogue- to-digital conversion of voice signals, voice compression, echo cancellation, and voiceactivity detection. The functions like call origination, call detection, signaling, and phone number translations are performed by the microprocessor. Gateways exist in several forms; for example, the gateway could be a dedicated telecommunication equipment chassis, or even a generic PC running VoIP software. IV. BANDWIDTH AND CODECS In addition to performing the analogue-to digital conversion, CODECs compress the voice data stream. Compression of the voice waveform results in bandwidth savings. The output from the CODECs is a data stream that is put into IP packets and transported across the network to an endpoint. The endpoints must use the same standards as well as a common set of CODEC parameters. Use of different standards or parameters at the endpoints will lead to unintelligible communication. The table I shows some of the coding standards that are covered by the International Telecommunications Union (ITU). Use of complex coders with higher compression ratios reduces the bandwidth consumption. But there is a price to be paid for reduced bandwidth consumption: increased conversion delay. Another way to save bandwidth is the use of silence suppression, in which voice packets aren’t sent between the gaps in human conversations. The voice-activity detection technique allows the monitoring of silence in speech data.
TABLE I
Fig 2. Voice Gateway
ITU STANDARD CODECS
III. VOICE GATEWAY The VoIP network acts as a gateway to the existing PSTN network. This gateway forms the interface for transportation of the voice content over the IP network. Gateways are responsible for call origination; call detection, analogue-to-digital conversion of voice, and creation of voice packets (CODEC functions). Voice (analogue and/or digital) compression, echo cancellation, silence suppression, and statistics gathering are their optional features. The gateways must also perform some of the database services, such as phone number translations, host
ITU standard
Description
Bandwidth
Conversion delay
G.711 G.721 G.728 G.729 G.723.1
PCM ADPCM LD-CELP CS-ACELP Multirate CELP
64 32,16,24,40 16 8 6.3,5.3
<1.00 <1.00 2.50 15.00 30.00
V. PACKET DELAY VoIP quality is also affected by the packet delay. The endto-end packet delay in a network is a result of the incremental
delays in the connection path. The use of voice CODECs adds a small amount of processing delay. A delay greater than 100 ms will interfere with normal conversation. Longer delays can cause echoes. The network delays can be reduced through a careful network architecture, equipment selection, and configuration. The following are sources of delay in an end-to-end, voiceover-packet call: A. Accumulation Delay This delay is caused by the need to collect a frame of voice samples to be processed by the voice coder. It is related to the type of voice coder used and varies from a single sample time (.125 microseconds) to many milliseconds. A representative list of standard voice coders and their frame times follows: •G.726 adaptive differential pulse-code modulation (ADPCM) (16, 24, 32, 40 kbps)—0.125 microseconds •G.728 LD–code excited linear prediction (CELP)(16 kbps)— 2.5 milliseconds •G.729 CS–ACELP (8 kbps)—10 milliseconds •G.723.1 Multirate Coder (5.3, 6.3 kbps)—30 milliseconds B. Processing Delay This delay is caused by the actual process of encoding and collecting the encoded samples into a packet for transmission over the packet network. The encoding delay is a function of both the processor execution time and the type of algorithm used. Often, multiple voice-coder frames will be collected in a single packet to reduce the packet network overhead. For example, three frames of G.729 code words, equaling 30 milliseconds of speech, may be collected and packed into a single packet. C. Network Delay This delay is caused by the physical medium and protocols used to transmit the voice data and by the buffers used to remove packet jitter on the receive side. Network delay is a function of the capacity of the links in the network and the processing that occurs as the packets transit the network. The jitter buffers add delay, which is used to remove the packetdelay variation to which each packet is subjected as it transits the packet network. This delay can be a significant part of the
overall delay, as packet-delay variations can be as high as 70 to 100 milliseconds in some frame-relay and IP networks.
D. Jitter The delay problem is compounded by the need to remove jitter, a variable interpacket timing caused by the network a packet traverses. Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence. This causes additional delay. The two conflicting goals of minimizing delay and removing jitter have engendered various schemes to adapt the jitter buffer size to match the time-varying requirements of network jitter removal. This adaptation has the explicit goal of minimizing the size and delay of the jitter buffer, while at the same time preventing buffer underflow caused by jitter. E. Echo Compensation Echo in a telephone network is caused by signal reflections generated by the hybrid circuit that converts between a fourwire circuit (a separate transmit and receive pair) and a twowire circuit (a single transmit and receive pair). These reflections of the speaker's voice are heard in the speaker's ear. Echo is present even in a conventional circuit-switched telephone network. However, it is acceptable because the round-trip delays through the network are smaller than 50 milliseconds and the echo is masked by the normal side tone every telephone generates. Echo becomes a problem in voice-over-packet networks because the round-trip delay through the network is almost always greater than 50 milliseconds. Thus, echo-cancellation techniques are always used. ITU standard G.165 defines performance requirements that are currently required for echo cancellers. The ITU is defining much more stringent performance requirements in the G.IEC specification. A new concept for echo control was invented at Bell Laboratories in 1964, commonly called echo cancellation. Echo cancellation was a revolutionary departure from the previous technique of opening (temporarily disconnecting) the speech path to prevent echo signals from being returned over the long distance circuit. Echo is generated toward the packet network from the telephone network. The echo canceller compares the voice
data received from the packet network with voice data being transmitted to the packet network. The echo from the telephone network hybrid is removed by a digital filter on the transmit path into the packet network. VI. VOIP APPLICATIONS
Fig 4. Interoffice trunking application
Fig 3. Branch office application
A wide variety of applications are enabled by the transmission of VoIP networks. This tutorial will explore three examples of these applications. The first application, shown in Figure 1, is a network configuration of an organization with many branch offices (e.g., a bank) that wants to reduce costs and combine traffic to provide voice and data access to the main office. This is accomplished by using a packet network to provide standard data transmission while at the same time enhancing it to carry voice traffic along with the data. Typically, this network configuration will benefit if the voice traffic is compressed as a result of the low bandwidth available for this access application. Voice over packet provides the interworking function (IWF), which is the physical implementation of the hardware and software that allows the transmission of combined voice and data over the packet network. The interfaces the IWF must support in this case are analog interfaces, which directly connect to telephones or key systems. The IWF must emulate the functions of both a private branch exchange (PBX) for the telephony terminals at the branches, as well as the functions of the telephony terminals for the PBX at the home office. The IWF accomplishes this by implementing signaling software that performs these functions.
A second VoIP application, shown in Figure 2, is a trunking application. In this scenario, an organization wishes to send voice traffic between two locations over the packet network and replace the tie trunks used to connect the PBXs at the locations. This application usually requires the IWF to support a higher-capacity digital channel than the branch application, such as a T1/E1 interface of 1.544 or 2.048 Mbps. The IWF emulates the signalling functions of a PBX, resulting in significant savings to companies' communications costs. VII.STANDARDS Over the next few years, the industry will address the bandwidth limitations by upgrading the Internet backbone to asynchronous transfer mode (ATM), the switching fabric designed to handle voice, data, and video traffic. Such network optimization will go a long way toward eliminating network congestion and the associated packet loss. The Internet industry also is tackling the problems of network reliability and sound quality on the Internet through the gradual adoption of standards. Standards-setting efforts are focusing on the three central elements of Internet telephony: the audio codec format; transport protocols; and directory services. In May 1996, the International Telecommunications Union (ITU) ratified the H.323 specification, which defines how voice, data, and video traffic will be transported over IP– based local area networks; it also incorporates the T.120 dataconferencing standard (see Figure). The recommendation is based on the real-time protocol/real-time control protocol (RTP/RTCP) for managing audio and video signals. As such, H.323 addresses the core Internet-telephony applications by defining how delay-sensitive traffic, (i.e.,
voice and video), gets priority transport to ensure real-time communications service over the Internet. (The H.324 specification defines the transport of voice, data, and video over regular telephony networks, while H.320 defines the protocols for transporting voice, data, and video over integrated services digital network (ISDN). H.323 is a set of recommendations, one of which is G.729 for audio CODECs, which the ITU ratified in November 1995. Despite the ITU recommendation, however, the Voice over IP (VoIP) Forum in March 1997 voted to recommend the G.723.1 specification over the G.729 standard. The industry consortium, which is led by Intel and Microsoft, agreed to sacrifice some sound quality for the sake of greater bandwidth efficiency—G.723.1 requires 6.3 kbps, while G.729 requires 7.9 kbps. Adoption of the audio codec standard, while an important step, is expected to improve reliability and sound quality mostly for intranet traffic and point-to-point IP connections. To achieve PSTN–like quality, standards are required to guarantee Internet connections. F. VoIP market in India IDC Asia Pacific, a market research firm, estimates VoIP services in India to be worth US$2.8 billion by 2005, which will make it the second biggest VoIP market in the region after China. IDC expects revenues from VoIP from the whole of Asia Pacific to be worth US$13.8 billion. VIII.MERITS • Bypass long distance phone charges – The only cost incurred will result from our ISP’s standard, internet connection rates. VoIP is as easy to use as a telephone – Its state of the art Graphical User Interface (GUI) intensive and user friendly. The setup wizard makes installation practically effortless. Business telephone functionality – VoIP can be used in the network configuration of an organization with many branch offices that want to reduce costs and combine traffic to provide voice and data to main office. IX. DEMERITS • Audio quality – Although the phones have improved, none of the latest crop sounds as good as a regular phone line. The digitalized voice sounds tiny and there can be
gaps of several seconds as it makes its way across the internet • Degradation of quality – Packets is queued at routers during periods of congestion. If the congestion is significant, packets may even be dropped Security- Any H.323 IP-aware user can trap into any conversation on the system. Therefore the system is less secure. X. CONCLUSIONS VoIP technology offers broadband services and the integration of voice and data at all levels. One key factor that is driving the VoIP application development and deployment is reduced voice service charges. In addition to cost advantages, VoIP services have compelling technical advantages over circuit switching. VoIP networks are based more on an open architecture than their circuit-switched contemporaries. This open, standards-based architecture means that VoIP services are more interchangeable and more modular than those offered by a proprietary voice-based PSTN network. Open standards translate into the realization of new services that one can rapidly develop and deploy. Moreover, VoIP is suitable for computer telephony integration and other next generation applications.
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REFERENCES
[1] [2] [3] [4] [5] [6] [7] electronics for you – December 2002 www.iec.com N. O. Johannesson, “The ETSI computation model: a tool for transmission planning of telephone networks”, IEEE Communications Magazine, 35(1), pp. 70-79, 2002. www.webphone.com www.ieee.org www.voicendata.com voice&data magazine-September 2002
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Rinzam A
Department of Computer Science and Engineering M G College of Engineering
[email protected]
Abstract— The Voice-Over Internet Protocol (VoIP) technology allows the voice information to pass over IP data networks. This technology results in huge savings on the amount of physical resources required to communicate by voice over long distance. It does so by exchanging the information in packets over a data network. The basic functions performed by a VoIP include – signalling, data basing, call connect and disconnect, and coding/decoding. The steps involved in originating and internet telephone call are the conversion of the analogue voice signal to digital format and compression/translation of the signal into internet protocol (IP) packets for transmission over the internet; the process is reversed at the receiving end. VoIP software’s like Vocal TEC or Net 2 Phone are available for the user. With the exception of phone to phone, the user must posses an array of equipment which should at minimum include VoIP software, an internet connection, and a multimedia computer with a sound card, speakers, a microphone and a modem.
Fig.1 How VOIP works
The VoIP network acts as a gateway to the existing PSTN network. This gateway forms the interface for transportation of the voice content over the IP networks. Gateways are responsible for all call origination, call detection, analogue to digital conversion of voice, and creation of voice packets.
II. BASIC FLOW OF VOIP NETWORK The VoIP networks replace the traditional public-switched telephone networks (PSTNs), as these can perform the same functions as the PSTN networks. The functions performed include signaling, data basing, call connect and disconnect, and coding-decoding. A. Signaling Signaling in a VoIP network is accomplished by the exchange of IP datagram messages between the components. The format of these messages is covered by the standard datalink layer protocols. B. Database services
I. INTRODUCTION The development of very fast, inexpensive microprocessors and special-purpose switching chips, coupled with highly reliable fibre-optic transmission systems, has made it possible to build economical, ubiquitous, high-speed packet-based data networks. Similarly, the development of very fast, inexpensive digital signal processors (DSPs) has made it practical to digitise and compress voice and fax signals into data packets. The natural evolution of these two developments is to combine digitised voice and fax packets with packet data, creating integrated data-voice networks. The voice-overInternet protocol (VoIP) technology allows voice information to pass over IP data networks. Primarily, the cost savings that accrue from operating a single, shared network have motivated this convergence of telecommunications and data communications
Database services are a way to locate an endpoint and translate the addressing that two networks use; for example, the PSTN uses phone numbers to identify endpoints, while a VoIP network could use an IP address and port numbers to identify an endpoint. A call control database contains these mappings and translations. C. Calls connect and disconnect (bearer control) The connection of a call is made by two endpoints opening communication sessions between each other. In the PSTN, the public (or private) switch connects logical channels through the network to complete the calls. In a VoIP implementation, a multimedia stream (audio, video, or both) is transported in real time. The connection path is the bearer channel and represents the voice or video content being delivered. When communication is complete, the IP sessions are released and, optionally, network resources are freed. D. CODEC operations Voice communication is analogue, while data networking is digital. Analogue waveforms are converted into digital information by using a coder-decoder (CODEC).
lookup, and signaling. The extent of gateway functionalities is based on the VoIP-enabling products used. Fig. 1 shows the architecture of a typical gateway. The DSP in a gateway is responsible for signal processing functions such as analogue- to-digital conversion of voice signals, voice compression, echo cancellation, and voiceactivity detection. The functions like call origination, call detection, signaling, and phone number translations are performed by the microprocessor. Gateways exist in several forms; for example, the gateway could be a dedicated telecommunication equipment chassis, or even a generic PC running VoIP software. IV. BANDWIDTH AND CODECS In addition to performing the analogue-to digital conversion, CODECs compress the voice data stream. Compression of the voice waveform results in bandwidth savings. The output from the CODECs is a data stream that is put into IP packets and transported across the network to an endpoint. The endpoints must use the same standards as well as a common set of CODEC parameters. Use of different standards or parameters at the endpoints will lead to unintelligible communication. The table I shows some of the coding standards that are covered by the International Telecommunications Union (ITU). Use of complex coders with higher compression ratios reduces the bandwidth consumption. But there is a price to be paid for reduced bandwidth consumption: increased conversion delay. Another way to save bandwidth is the use of silence suppression, in which voice packets aren’t sent between the gaps in human conversations. The voice-activity detection technique allows the monitoring of silence in speech data.
TABLE I
Fig 2. Voice Gateway
ITU STANDARD CODECS
III. VOICE GATEWAY The VoIP network acts as a gateway to the existing PSTN network. This gateway forms the interface for transportation of the voice content over the IP network. Gateways are responsible for call origination; call detection, analogue-to-digital conversion of voice, and creation of voice packets (CODEC functions). Voice (analogue and/or digital) compression, echo cancellation, silence suppression, and statistics gathering are their optional features. The gateways must also perform some of the database services, such as phone number translations, host
ITU standard
Description
Bandwidth
Conversion delay
G.711 G.721 G.728 G.729 G.723.1
PCM ADPCM LD-CELP CS-ACELP Multirate CELP
64 32,16,24,40 16 8 6.3,5.3
<1.00 <1.00 2.50 15.00 30.00
V. PACKET DELAY VoIP quality is also affected by the packet delay. The endto-end packet delay in a network is a result of the incremental
delays in the connection path. The use of voice CODECs adds a small amount of processing delay. A delay greater than 100 ms will interfere with normal conversation. Longer delays can cause echoes. The network delays can be reduced through a careful network architecture, equipment selection, and configuration. The following are sources of delay in an end-to-end, voiceover-packet call: A. Accumulation Delay This delay is caused by the need to collect a frame of voice samples to be processed by the voice coder. It is related to the type of voice coder used and varies from a single sample time (.125 microseconds) to many milliseconds. A representative list of standard voice coders and their frame times follows: •G.726 adaptive differential pulse-code modulation (ADPCM) (16, 24, 32, 40 kbps)—0.125 microseconds •G.728 LD–code excited linear prediction (CELP)(16 kbps)— 2.5 milliseconds •G.729 CS–ACELP (8 kbps)—10 milliseconds •G.723.1 Multirate Coder (5.3, 6.3 kbps)—30 milliseconds B. Processing Delay This delay is caused by the actual process of encoding and collecting the encoded samples into a packet for transmission over the packet network. The encoding delay is a function of both the processor execution time and the type of algorithm used. Often, multiple voice-coder frames will be collected in a single packet to reduce the packet network overhead. For example, three frames of G.729 code words, equaling 30 milliseconds of speech, may be collected and packed into a single packet. C. Network Delay This delay is caused by the physical medium and protocols used to transmit the voice data and by the buffers used to remove packet jitter on the receive side. Network delay is a function of the capacity of the links in the network and the processing that occurs as the packets transit the network. The jitter buffers add delay, which is used to remove the packetdelay variation to which each packet is subjected as it transits the packet network. This delay can be a significant part of the
overall delay, as packet-delay variations can be as high as 70 to 100 milliseconds in some frame-relay and IP networks.
D. Jitter The delay problem is compounded by the need to remove jitter, a variable interpacket timing caused by the network a packet traverses. Removing jitter requires collecting packets and holding them long enough to allow the slowest packets to arrive in time to be played in the correct sequence. This causes additional delay. The two conflicting goals of minimizing delay and removing jitter have engendered various schemes to adapt the jitter buffer size to match the time-varying requirements of network jitter removal. This adaptation has the explicit goal of minimizing the size and delay of the jitter buffer, while at the same time preventing buffer underflow caused by jitter. E. Echo Compensation Echo in a telephone network is caused by signal reflections generated by the hybrid circuit that converts between a fourwire circuit (a separate transmit and receive pair) and a twowire circuit (a single transmit and receive pair). These reflections of the speaker's voice are heard in the speaker's ear. Echo is present even in a conventional circuit-switched telephone network. However, it is acceptable because the round-trip delays through the network are smaller than 50 milliseconds and the echo is masked by the normal side tone every telephone generates. Echo becomes a problem in voice-over-packet networks because the round-trip delay through the network is almost always greater than 50 milliseconds. Thus, echo-cancellation techniques are always used. ITU standard G.165 defines performance requirements that are currently required for echo cancellers. The ITU is defining much more stringent performance requirements in the G.IEC specification. A new concept for echo control was invented at Bell Laboratories in 1964, commonly called echo cancellation. Echo cancellation was a revolutionary departure from the previous technique of opening (temporarily disconnecting) the speech path to prevent echo signals from being returned over the long distance circuit. Echo is generated toward the packet network from the telephone network. The echo canceller compares the voice
data received from the packet network with voice data being transmitted to the packet network. The echo from the telephone network hybrid is removed by a digital filter on the transmit path into the packet network. VI. VOIP APPLICATIONS
Fig 4. Interoffice trunking application
Fig 3. Branch office application
A wide variety of applications are enabled by the transmission of VoIP networks. This tutorial will explore three examples of these applications. The first application, shown in Figure 1, is a network configuration of an organization with many branch offices (e.g., a bank) that wants to reduce costs and combine traffic to provide voice and data access to the main office. This is accomplished by using a packet network to provide standard data transmission while at the same time enhancing it to carry voice traffic along with the data. Typically, this network configuration will benefit if the voice traffic is compressed as a result of the low bandwidth available for this access application. Voice over packet provides the interworking function (IWF), which is the physical implementation of the hardware and software that allows the transmission of combined voice and data over the packet network. The interfaces the IWF must support in this case are analog interfaces, which directly connect to telephones or key systems. The IWF must emulate the functions of both a private branch exchange (PBX) for the telephony terminals at the branches, as well as the functions of the telephony terminals for the PBX at the home office. The IWF accomplishes this by implementing signaling software that performs these functions.
A second VoIP application, shown in Figure 2, is a trunking application. In this scenario, an organization wishes to send voice traffic between two locations over the packet network and replace the tie trunks used to connect the PBXs at the locations. This application usually requires the IWF to support a higher-capacity digital channel than the branch application, such as a T1/E1 interface of 1.544 or 2.048 Mbps. The IWF emulates the signalling functions of a PBX, resulting in significant savings to companies' communications costs. VII.STANDARDS Over the next few years, the industry will address the bandwidth limitations by upgrading the Internet backbone to asynchronous transfer mode (ATM), the switching fabric designed to handle voice, data, and video traffic. Such network optimization will go a long way toward eliminating network congestion and the associated packet loss. The Internet industry also is tackling the problems of network reliability and sound quality on the Internet through the gradual adoption of standards. Standards-setting efforts are focusing on the three central elements of Internet telephony: the audio codec format; transport protocols; and directory services. In May 1996, the International Telecommunications Union (ITU) ratified the H.323 specification, which defines how voice, data, and video traffic will be transported over IP– based local area networks; it also incorporates the T.120 dataconferencing standard (see Figure). The recommendation is based on the real-time protocol/real-time control protocol (RTP/RTCP) for managing audio and video signals. As such, H.323 addresses the core Internet-telephony applications by defining how delay-sensitive traffic, (i.e.,
voice and video), gets priority transport to ensure real-time communications service over the Internet. (The H.324 specification defines the transport of voice, data, and video over regular telephony networks, while H.320 defines the protocols for transporting voice, data, and video over integrated services digital network (ISDN). H.323 is a set of recommendations, one of which is G.729 for audio CODECs, which the ITU ratified in November 1995. Despite the ITU recommendation, however, the Voice over IP (VoIP) Forum in March 1997 voted to recommend the G.723.1 specification over the G.729 standard. The industry consortium, which is led by Intel and Microsoft, agreed to sacrifice some sound quality for the sake of greater bandwidth efficiency—G.723.1 requires 6.3 kbps, while G.729 requires 7.9 kbps. Adoption of the audio codec standard, while an important step, is expected to improve reliability and sound quality mostly for intranet traffic and point-to-point IP connections. To achieve PSTN–like quality, standards are required to guarantee Internet connections. F. VoIP market in India IDC Asia Pacific, a market research firm, estimates VoIP services in India to be worth US$2.8 billion by 2005, which will make it the second biggest VoIP market in the region after China. IDC expects revenues from VoIP from the whole of Asia Pacific to be worth US$13.8 billion. VIII.MERITS • Bypass long distance phone charges – The only cost incurred will result from our ISP’s standard, internet connection rates. VoIP is as easy to use as a telephone – Its state of the art Graphical User Interface (GUI) intensive and user friendly. The setup wizard makes installation practically effortless. Business telephone functionality – VoIP can be used in the network configuration of an organization with many branch offices that want to reduce costs and combine traffic to provide voice and data to main office. IX. DEMERITS • Audio quality – Although the phones have improved, none of the latest crop sounds as good as a regular phone line. The digitalized voice sounds tiny and there can be
gaps of several seconds as it makes its way across the internet • Degradation of quality – Packets is queued at routers during periods of congestion. If the congestion is significant, packets may even be dropped Security- Any H.323 IP-aware user can trap into any conversation on the system. Therefore the system is less secure. X. CONCLUSIONS VoIP technology offers broadband services and the integration of voice and data at all levels. One key factor that is driving the VoIP application development and deployment is reduced voice service charges. In addition to cost advantages, VoIP services have compelling technical advantages over circuit switching. VoIP networks are based more on an open architecture than their circuit-switched contemporaries. This open, standards-based architecture means that VoIP services are more interchangeable and more modular than those offered by a proprietary voice-based PSTN network. Open standards translate into the realization of new services that one can rapidly develop and deploy. Moreover, VoIP is suitable for computer telephony integration and other next generation applications.
•
REFERENCES
[1] [2] [3] [4] [5] [6] [7] electronics for you – December 2002 www.iec.com N. O. Johannesson, “The ETSI computation model: a tool for transmission planning of telephone networks”, IEEE Communications Magazine, 35(1), pp. 70-79, 2002. www.webphone.com www.ieee.org www.voicendata.com voice&data magazine-September 2002
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