Lakshmi Voip

Seminar 2004

VoIP-Voice over Internet Protocol

Department of Electronics & Communication Govt Engineering College Thrissur

VoIP (Voice over Internet Protocol)

Submitted On 15-10-04

Submitted by Lakshmi Menon S7 ECE 630 Co-ordinator:Muneera C R

Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

ACKNOWLEDGEMENT

First,and foremost I thank God Almighty for making this venture a success. I extend my sincere gratitude to Prof. Indiradevi, Head of Electronics

and

Communication

Department,

Govt

Engineering College, Thrissur for providing me with necessary infrastructure. I would like to convey a deep sense of gratitude to the seminar co-ordinator

Mrs. C R

Muneera for the timely advices. I also extend my sincere thanks to my friends for their help.

Dept of ECE

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Seminar 2004

VoIP-Voice over Internet Protocol

ABSTRACT 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 digitize and compress voice and fax signals into data packets. The natural evolution of these two developments is to combine digitized voice and fax packets with packet data, creating integrated data-voice networks. The voice-over-Internet 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. VoIP allows you to make telephone calls using a computer network, over a data network like the Internet. VoIP converts the voice signal from your telephone into a digital signal that travels over the Internet then converts it back at the other end so you can speak to anyone with a regular phone number.

Dept of ECE

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VoIP-Voice over Internet Protocol

INDEX S.No 1 2 3 4

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TOPIC BASIC FLOW OF VoIP NETWORK VOICE GATEWAY A TYPICAL VoIP NETWORK APPLICATIONS IDENTIFICATION OF MAJOR SYSTEM COMPONENTS • Gateways • Gatekeepers • IP Telephones • PC Software Phones VoIP PRODUCTS • Hard Phones • Soft Phones VoIP QoS (Quality of Service) ISSUES • Delay • Lost Packet Compensation • Echo Compensation ADVANTAGES OF USING VoIP TECHNICAL BARRIERS FUTURE OF VoIP TELEPHONY CONCLUSION REFERENCES

Dept of ECE

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Seminar 2004

VoIP-Voice over Internet Protocol

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, databasing, call connect and disconnect, and codingdecoding. Signaling. Signaling in a VoIP network is accomplished by the exchange of datagram messages between the components. The format of these messages is covered by the standard datalink layer protocols. Database services. 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. Call 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. CODEC operations. Voice communication is analogue, while data networking is digital. Analogue waveforms are converted into digital information by using a coder-decoder (CODEC). Dept of ECE

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VoIP-Voice over Internet Protocol

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 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.

FIG 1 ARCHITECTURE OF A TYPICAL GATEWAY

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Seminar 2004

VoIP-Voice over Internet Protocol

A TYPICAL VOIP NETWORK Fig. 2 shows a typical VoIP network. The IP network should ensure smooth delivery of voice and signaling information to the VoIP elements. Since the IP network is to carry both voice and data, it must be able to prioritize the voice traffic. This prioritization is required for real-time VoIP applications to ensure that voice traffic is unaffected by other network traffic. Without prioritization, the voice packets may be bogged down by heavy data traffic like large file transfers using file transfer protocol (FTP).The voice packets are encapsulated with real-time protocol (RTP) and real-time control protocol (RTCP) for real-time transfer. The resource reservation protocol (RSVP) is used at the networking gateways (such as the routers) to reserve a particular amount of bandwidth for real-time applications (VoIP, video multicasting, etc).

FIG 2 A TYPICAL FULL SERVICE VOIP NETWORK

Dept of ECE

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Seminar 2004

VoIP-Voice over Internet Protocol

Unlike the PCM data streams in circuitswitched telephony, VoIP data travels over the networks in packets. In VoIP digitized voice is bundled into IP packets and sent out into the network for delivery. Routers, switches, and other network equipment direct the packets to their destination IP address. This mode is called packetswitched telephony. The transport of voice packets is affected by several factors, such as the amount of bandwidth available in the network connection, the delay that the packet experiences, and any packet loss or corruption that occurs. The ability of the network to deliver the voice packets quickly and consistently is referred to as Quality of Service (QoS).

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VoIP-Voice over Internet Protocol

APPLICATIONS A wide variety of applications are enabled by the transmission of VoIP networks. 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. 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.

FIGURE 1. BRANCH OFFICE APPLICATION

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VoIP-Voice over Internet Protocol

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 signaling functions of a PBX, resulting in significant savings to companies' communications costs.

FIGURE 2. INTEROFFICE TRUNKING APPLICATION

A third application of VoIP software is interworking with cellular networks, as shown in Figure 3. The voice data in a digital cellular network is already compressed and packetized for transmission over the air by the cellular phone. Packet networks can then transmit the compressed cellular voice packet, saving a tremendous amount of bandwidth. The IWF provides the Dept of ECE

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VoIP-Voice over Internet Protocol

transcoding function required to convert the cellular voice data to the format required by the public switched telephone network (PSTN).

FIGURE 3. INTEROFFICE TRUNKING APPLICATION

Mainly the different types of communications that exist in a VoIP are : ƒ PC to PC connection. ƒ PC to PHONE connection. ƒ PHONE to PHONE connection

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VoIP-Voice over Internet Protocol

PC to PC Communication:

ƒ Need a PC with sound card ƒ IP Telephony software: Cuseeme, Internet Phone, ... ƒ Video optional PC to Phone Communication:

ƒ Need a gateway that connects IP network to phone network (Router to PBX)

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VoIP-Voice over Internet Protocol

``PC TO PHONE CONNECTION

Phone to Phone communication:

ƒ Need more gateways that connect IP network to phone Networks. ƒ The IP network could be dedicated intra-net or the Internet. ƒ The phone networks could be intra-company PBXs or the carrier switches.

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VoIP-Voice over Internet Protocol

Phone to Phone communication

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VoIP-Voice over Internet Protocol

IDENTIFICATION OF MAJOR SYSTEM COMPONENTS ƒ Gateways The gateways are the devices that communicate between the telephone signals and the IP endpoint. The IP endpoint usually speaks H.323 for media stream and more recently Session Initiation protocol (SIP). The gateways usually perform the following 6 functions: • Search function When an IP gateway is used to place a call across an IP network, it receives a called party phone number. It converts it into the IP address of the far end gateway, possibly through a table lookup in the originating gateway or in a centralized directory server. • Connection Function The originating gateway establishes a connection to the destination gateway, exchanges call setup, compatibility information and performs any option negotiation and security handshake. • Digitizing function Analog telephone signals coming into a trunk on the gateway are digitized by the gateway into a format useful to the gateway, usually 64 kbps PCM. This requires the gateway to interface to a variety of Telephone-signaling conventions.

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VoIP-Voice over Internet Protocol

• Demodulation functions With some gateways the gateway trunk can accept only a voice signal or a fax signal but not both. But sophisticated gateways handle both. When the signal is a fax,it is demodulated by the DSP back into the original 2.4-14.4 kbps digital format. This is then put into the IP packets for transmission. The demodulated information is remodulated back to the original analog fax signal by the remote • Compression functions When the signal is determined to be voice, it is usually compressed by a DSP from 64K PCM to a 5.3 Kbps signal, which is the G.723.1 standard. • Decompression and Remodulation functions At the same time that the gateway performs steps 15, it is also receiving packets. Hence this function is required

ƒ Gatekeepers Terminals are the L AN client endpoints that provide real time two-way communications. When an endpoint is switched on, it performs a multicast discovery for a gatekeeper and registers with it. Thus the gatekeeper knows how many users are connected and where they are located. The collection of a gatekeeper and its registered endpoints is called as a zone. A gatekeeper is required to perform the following functions: • Address translation Translation of an alias address to a Transport Address using a table updated via Registration messages. Dept of ECE

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VoIP-Voice over Internet Protocol

• Admissions control Authorization of LAN access, using Admissions Requests or Confirm and Reject (ARQ/ARC/ARJ), bandwidth or some other criteria. • Bandwidth management Support for Bandwidth Request, Confirm and Reject messages, or a null function that accepts all requests for bandwidth changes. • Zone management The Gatekeeper provides the above functions forterminals, MCUs, and Gateways, which are registered in its Zone of control.

FIG 1 ZONE MANAGEMENT IN A GATEKEEPER

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VoIP-Voice over Internet Protocol

FIG 2 INTERZONE GATEWAY GATEKEEPER COMMUNICATION

FIG 3 INTRAZONE GATEWAY GATEKEEPER COMMUNICATION

ƒ IP Telephones These are devices, which replace the existing telephones by providing enhanced services suited to VOIP. At the same time they should retain the capabilities of the original phones to keep the user comfortable.

ƒ PC Software phones This arrangement consists of a microphone connected to a PC interfaced by a card and running a software, which permits voice and multimedia transfer over the Internet. Microsoft NetMeeting is an example.

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VoIP-Voice over Internet Protocol

VOIP PRODUCTS ƒ Hard Phones • Broadband Hard Phones A broadband hard phone is a self contained IP telephone that looks just like a conventional phone but instead of a conventional phone jack, it has an Ethernet port through which it communicates directly with a VoIP server, VoIP gateway or another VoIP phone. Since a broadband hard phone communicates directly with a VoIP server, VoIP gateway or another VoIP phone it does not require any personal computer nor any software running on a personal computer to make or receive VoIP phone calls. It can be used independently, all that is required is an internet connection. While PC based software solutions are cheaper, a hard phone is the best solution for IP telephony.

• Dialup Hard Phones A dialup hard phone is a hard phone with a built-in modem instead of the Ethernet port. It will connect through the modem via a dialup internet service to a remote VoIP server and is therefore self contained. It does not require a personal computer nor any software to be run on a personal computer to make and receive VoIP phone calls. All that is required is a phone line and a dialup internet account. Dialup hard phones are popular in countries where there is very little broadband infrastructure yet.

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VoIP-Voice over Internet Protocol

• WLAN or WiFi Phones A WLAN or WiFi phone is a hard phone with a built-in WiFi transceiver unit instead of an Ethernet port to connect to a WiFi base station and from there to a remote VoIP server. It does not require a personal computer nor any software to be run on a personal computer to make and receive VoIP phone calls. All that is required is access to a WiFi base station.

ƒ Soft Phones A soft phone is an IP telephone in software. It can be installed on a personal computer and function as an IP phone. Soft phones require appropriate audio hardware to be present on the personal computer they run. This can either be a sound card with speakers or earphones and a microphone, or, alternatively a USB phone set. Soft phones are inferior to hard phones but cheaper to obtain, many are available as a free download.

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VoIP-Voice over Internet Protocol

VOIP QOS(QUALITY OF SERVICE) ISSUES The advantages of reduced cost and bandwidth savings of carrying voice-over-packet networks are associated with some quality-of-service (QoS) issues unique to packet networks.

ƒ Delay Delay causes two problems: echo and talker overlap. Echo is caused by the signal reflections of the speaker's voice from the farend telephone equipment back into the speaker's ear. Echo becomes a significant problem when the round-trip delay becomes greater than 50 milliseconds. As echo is perceived as a significant quality problem, voice-over-packet systems must address the need for echo control and implement some means of echo cancellation. Talker overlap (or the problem of one talker stepping on the other talker's speech) becomes significant if the one-way delay becomes greater than 250 milliseconds. The end-to-end delay budget is therefore the major constraint and driving requirement for reducing delay through a packet network. The following are sources of delay in an end-to-end, voiceover-packet call: • Accumulation Delay (Sometimes Called Algorithmic 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 Dept of ECE

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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

• 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.

• 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 Dept of ECE

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VoIP-Voice over Internet Protocol

packets transit the network. The jitter buffers add delay, which is used to remove the packet-delay 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. • 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. Two approaches to adapting the jitter buffer size are detailed below. The approach selected will depend on the type of network the packets are traversing. The first approach is to measure the variation of packet level in the jitter buffer over a period of time and incrementally adapt the buffer size to match the calculated jitter. This approach works best with networks that provide a consistent jitter performance over time, such as ATM networks.

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VoIP-Voice over Internet Protocol

The second approach is to count the number of packets that arrive late and create a ratio of these packets to the number of packets that are successfully processed. This ratio is then used to adjust the jitter buffer to target a predetermined, allowable late-packet ratio. This approach works best with the networks with highly variable packet-interarrival intervals—such as IP networks. In addition to the techniques described, the network must be configured and managed to provide minimal delay and jitter, enabling a consistent QoS.

ƒ Lost-Packet Compensation Lost packets can be an even more severe problem, depending on the type of packet network that is being used. Because IP networks do not guarantee service, they will usually exhibit a much higher incidence of lost voice packets than ATM networks. In current IP networks, all voice frames are treated like data. Under peak loads and congestion, voice frames will be dropped equally with data frames. The data frames, however, are not time sensitive, and dropped packets can be appropriately corrected through the process of retransmission. Lost voice packets, however, cannot be dealt with in this manner. Some schemes used by voice-over-packet software to address the problem of lost frames are as follows: •

interpolate for lost speech packets by replaying the last packet received during the interval when the lost packet was supposed to be played out; this scheme is a simple method that fills the time between noncontiguous speech frames; it works well when the incidence of lost frames is infrequent; it

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•

VoIP-Voice over Internet Protocol

does not work well if there are a number of lost packets in a row or a burst of lost packets send redundant information at the expense of bandwidth utilization; this basic approach replicates and sends the nth packet of voice information along with the (n+1)th packet; this method has the advantage of being able to correct for the lost packet exactly; however, this approach uses more bandwidth and also creates greater delay

•

use a hybrid approach with a much lower bandwidth voice coder to provide redundant information carried along in the (n+1)th packet; this reduces the problem of the extra bandwidth required but fails to solve the problem of delay.

ƒ Echo Compensation Echo in a telephone network is caused by signal reflections generated by the hybrid circuit that converts between a four-wire circuit (a separate transmit and receive pair) and a two-wire 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. Echo is generated toward the packet Dept of ECE

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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.

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VoIP-Voice over Internet Protocol

ADVANTAGES OF USING VOIP When you are using PSTN line, you typically pay for time used to a PSTN line manager company: more time you stay at phone and more you'll pay. In addition you couldn't talk with other that one person at a time. In opposite with VoIP mechanism you can talk all the time with every person you want (the needed is that other person is also connected to Internet at the same time), as far as you want (money independent) and, in addition, you can talk with many people at the same time. If you're still not persuaded you can consider that, at the same time, you can exchange data with people are you talking with, sending images, graphs and videos. • Integration of Voice and Data The integration of voice and data traffic will be demanded by multi application software. The inevitable evolution will be web servers capable of interacting with voice, data and images. • Simplification An integrated infra structure that supports all forms of communication allows more standardization and lesser equipment management. The result is a fault tolerant design. • Network Efficiency The integration of voice and data effectively fills up the data communication channels efficiently, thus providing bandwidth consolidation. The idea is to move away from the TDM scheme wherein the user is given bandwidth when he is not talking. Data networks do not do this. It is a big saving when one considers the Dept of ECE

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statistics that 50% of a conversation is silence. The network efficiency can be further boosted, by removing the redundancy in certain speech patterns. • Cost reduction The Public Switched Telephone Networks' toll services can be bypassed using the Internet backbone, which means slash in prices of the long distance calls. However these reductions may slightly decrease when the Federal communications Commission (FCC) removes the Enhanced Service Provider (ESP) status granted to Internet service providers (ISPs) by which they do not have to pay the local access fees to use the telephone company (TELCO) local access facilities.Access fees form a significant part of all long distance calls. But in spite of this, the circuit switched telephony would be expensive because of lack of bandwidth consolidation and speech compression techniques.

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VoIP-Voice over Internet Protocol

TECHNICAL BARRIERS The ultimate objective of Internet telephony is, of course, reliable, high-quality voice service, the kind that users expect from the PSTN. At the moment, however, that level of reliability and sound quality is not available on the Internet, primarily because of bandwidth limitations that lead to packet loss. In voice communications, packet loss shows up in the form of gaps or periods of silence in the conversation, leading to a clipped-speech effect that is unsatisfactory for most users and unacceptable in business communications. The Internet, a collection of more than 130,000 networks, is gaining in popularity as millions of new users sign on every month. The increasingly heavy use of the Internet's limited bandwidth often results in congestion, which, in turn, can cause delays in packet transmission. Such network delays mean packets are lost or discarded. In addition, because the Internet is a packet-switched or connectionless network, the individual packets of each voice signal travel over separate network paths for reassembly in the proper sequence at their ultimate destination. While this makes for a more efficient use of network resources than the circuit-switched PSTN, which routes a call over a single path, it also increases the chances for packet loss. Network reliability and sound quality also are functions of the voice-encoding techniques and associated voice-processing functions of the gateway servers. To date, most developers of Internet-telephony software, as well as vendors of gateway servers, have been using a variety of speech-compression protocols. The use of various speech-coding algorithms—with their different bit rates and mechanisms for reconstructing voice packets and Dept of ECE

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handling delays—produces varying levels of intelligibility and fidelity in sound transmitted over the Internet. The lack of standardized protocols also means that many Internet-telephony products do not interoperate with each other or with the PSTN.

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VoIP-Voice over Internet Protocol

FUTURE OF VOICE-OVER-INTERNET PROTOCOL (VOIP) TELEPHONY Several factors will influence future developments in VoIP products and services. Currently, the most promising areas for VoIP are corporate intranets and commercial extranets. Their IP– based infrastructures enable operators to control who can—and cannot—use the network. Another influential element in the ongoing Internet-telephony evolution is the VoIP gateway. As these gateways evolve from PC–based platforms to robust embedded systems, each will be able to handle hundreds of simultaneous calls. Consequently, corporations will deploy large numbers of them in an effort to reduce the expenses associated with high-volume voice, fax, and videoconferencing traffic. The economics of placing all traffic— data, voice, and video—over an IP–based network will pull companies in this direction, simply because IP will act as a unifying agent, regardless of the underlying architecture (i.e., leased lines, frame relay, or ATM) of an organization's network. Commercial extranets, based on conservatively engineered IP networks, will deliver VoIP and facsimile over Internet protocol (FAXoIP) services to the general public. By guaranteeing specific parameters, such as packet delay, packet jitter, and service interoperability, these extranets will ensure reliable network support for such applications. VoIP products and services transported via the public Internet will be niche markets that can tolerate the varying performance levels of that transport medium. Telecommunications carriers most likely will rely on the public Internet to provide telephone service between/among geographic locations that today Dept of ECE

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are high-tariff areas. It is unlikely that the public Internet's performance characteristics will improve sufficiently within the next two years to stimulate significant growth in VoIP for that medium. However, the public Internet will be able to handle voice and video services quite reliably within the next three to five years, once two critical changes take place: •

•

an increase by several orders of magnitude in backbone bandwidth and access speeds, stemming from the deployment of IP/ATM/synchronous optical network (SONET) and ISDN, cable modems, and x digital subscriber line (xDSL) technologies, respectively the tiering of the public Internet, in which users will be required to pay for the specific service levels they require

On the other hand, FAXoIP products and services via the public Internet will become economically viable more quickly than voice and video, primarily because the technical roadblocks are less challenging. Within two years, corporations will take their fax traffic off the PSTN and move it quickly to the public Internet and corporate Intranet, first through FAXoIP gateways and then via IP– capable fax machines. Standards for IP–based fax transmission will be in place by the end of this year. Throughout the remainder of this decade, videoconferencing (H.323) with data collaboration (T.120) will become the normal method of corporate communications, as network performance and interoperability increase and business organizations appreciate the economics of telecommuting. Soon, the video camera will be a standard piece of computer hardware, for full-featured multimedia systems, as well as for the less-than-$500 network-computer appliances now starting to appear in the market. The latter in particular should stimulate the residential demand and bring VoIP

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services to the mass market—including the roughly 60 percent of American households that still do not have a PC. The migration to VoIP networks is being driven by a number of factors; the key concept is that of advanced services. These services will be able to provide all the existing AIN features and add new ones based on IP services. A key aspect of the new VoIP infrastructure is that there is no need to build circuit switched connections between the devices, which reduces the cost of providing the services, and simplifies deployment. All systems, except the media gateway itself, require only IP interfaces. The IP interfaces of these devices can provide the telephony signaling as well as the media interfaces. This provides for simpler distributed signaling and processing capability, reduces the cost of components, and speeds up application development and deployment. The new IP based applications can be delivered in a variety of methods depending upon their complexity. For simple applications the Media Gateway Controller (MGC) can provide the application intelligence in a very distributed fashion. The MGC provides call control to the user via the Media Gateway with a client/server protocol called MGCP. The MGC controls all routing and call control to the devices within its’ MGCP domain. This functions very similarly to a Class-5 End Office switch and provides the same features one would expect on a standard POTS line. However it also has the ability to play tone and announcements to a caller, as well as gather digits from the caller. This provides capabilities similar to an SCP or Announcement Server for simple applications. These features are provided by the capabilities defined in MGCP and H.248. An example application and call flow is shown below. In this application a caller incorrectly dials a number and receives a message such as: “Your call cannot be completed as dialed”.

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Advanced Intelligent Network applications Today’s Advanced Intelligent Network (AIN) provides many useful features common to callers today. These features include Caller ID, Voice Mail, Call Waiting, Pre-and-post paid calling cards, 911, Call Blocking, and Auto Call-back to name a few. These features represent years of development and investment by vendors and service providers, and are delivered via a proven circuit switched infrastructure. The network architecture is shown in general format below: .

A subscriber in this network is provided primary dial tone and feature set by the Class-5 End Office switch. This system provides the basic call control features such as Dial Tone, Call Waiting, forwarding to Voice Mail, and billing. It passes calls to the Class-4 Tandem switch as needed based on the dial plan in the region. For messaging, a local mail server is usually employed for the subscriber base in the region. The calls requiring other advanced features such as 911, Local Number Portability, or 800 service are forwarded to a TCAP Service Control Point (SCP) for servicing when the proper Trigger Detection Points are met. A connection is established from the subscriber to an Intelligent Dept of ECE

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Seminar 2004

VoIP-Voice over Internet Protocol

Peripheral when media output and input analysis is required. Once the information is processed, with the help of a backend Database, the call can then be rerouted and the subscriber connected to the proper destination with the proper billing. Although this system is proven and widely deployed it is inefficient in terms of port and equipment utilization. A connection must be established between the End Office Switch and the Tandem switch as well as a connection from the Tandem to the SCP, or the Tandem to the IP. Numerous devices and paths could be required for any given call. It also mandates that the Intelligent Peripherals, Voice Mail Server, and some SCP have telephony interfaces as well as database access. This creates an expensive system, which has potential capacity constraints. VoIP Advanced Applications The migration to VoIP networks is being driven by a number of factors; the key concept is that of advanced services. These services will be able to provide all the existing AIN features and add new ones based on IP services. A key aspect of the new VoIP infrastructure is that there is no need to build circuit switched connections between the devices, which reduces the cost of providing the services, and simplifies deployment. All systems, except the media gateway itself, require only IP interfaces. The IP interfaces of these devices can provide the telephony signaling as well as the media interfaces. This provides for simpler distributed signaling and processing capability, reduces the cost of components, and speeds up application development and deployment. The new IP based applications can be delivered in a variety of methods depending upon their complexity. For simple applications the Media Gateway Controller (MGC) can provide the application intelligence in a very distributed fashion. The MGC provides call control to the user via the Media Gateway with a client/server protocol called MGCP. The MGC controls all routing Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

and call control to the devices within its’ MGCP domain. This functions very similarly to a Class-5 End Office switch and provides the same features one would expect on a standard POTS line. However it also has the ability to play tone and announcements to a caller, as well as gather digits from the caller. This provides capabilities similar to an SCP or Announcement Server for simple applications. These features are provided by the capabilities defined in MGCP and H.248. An example application and call flow is shown below. In this application a caller incorrectly dials a number and receives a message such as: “Your call cannot be completed as dialed”. EXAMPLE 1. MEDIA GATEWAY PROVIDED PROMPTS

Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

The above scenario is fairly simple and saves bandwidth on the network. Since the announcement is stored on the MG there is no bandwidth required except the MGCP signaling shown. This is effective for small networks with minimal prompting requirements. However due to the need for coordination of the file systems of the MGs and the messaging of the MGCs it can create administrative overhead. When networks grow to a certain size it becomes much easier to administer a separate Media or Announcement Server. This centralizes the configuration/maintenance for the audio files and allows all gateways in the domain to use the same source files. The calls are connected as though the Media Server was simply another endpoint, but in this case the endpoint is instructed to play a file using the same messages used in the above example. A network topology using a Media Server is shown below. Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

The approach shown above permits many MGs to share the file information stored in the Announcement Server, so their local files are not used. In general this will be made available to all MGs within the direct control of the MGC. This is called the MGC’s domain, and it controls all ports and all calls within its domain. Nonetheless this approach does require the MGC to be aware of the Announcement Server and know at what point to refer the caller to which prompt. This logic is fairly simple in the case of a misdialed number but is more difficult when the application requires more complexity and interaction with the caller. In the case of an Interactive Voice Response (IVR) server providing calling card service the logic for the interaction is much greater and is not feasibly controlled by an MGCP endpoint. In this case the Announcement Server has the ability to play announcements and retrieve DTMF tones, but due to the nature of MGCP, cannot easily control a call. This would be analogous to a POTs subscriber attempting to transfer a call on a Class-5 End Office switch, while maintaining control of the call. In the case of a Calling Card application the MGC can intercept the callers DTMF input, but the logic must be tightly coupled with the prompts of the Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

Announcement Server. In practice the coordination of this is beyond the capability of most MGCs and the function is moved onto a more specialized platform. This is usually done with a SIP Application Server.

Scaling Networks and Adding Intelligence As the VoIP networks grow they require a centralized control system for the specialized applications that become incorporated into the network. SIP provides for a lightweight and flexible protocol and architecture for this type of application server. In the case of a SIP Application Server the MGC passes the call to the appropriate server, which handles the call logic required for the call. In the case of misdialed call for instance the MGC can route the call to the server with a specific URI on the server and tells the server to play the appropriate message. All routing is handled similarly by the MGC, reducing the logic and processing requirements. Since the MGC can load level between a number of SIP Servers it is very beneficial to minimize the loading on the MGC and maximize the loading on the SIP Servers. This provides a more scalable End Office environment with a single MGC routing calls to many redundant SIP Servers. In addition the Application Server is available to take calls from any MGCP Domain in the network and can provide a centralized point for database access when used in advanced applications such as IVRs. The database activity can also be run in a distributed fashion depending upon the back end database selected. An example of the misdialed call is shown below.

Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

EXAMPLE 3. SIP APPLICATION SERVER

The application here is very simple, however in more advanced applications the requirement for full call control becomes more critical. In some applications the call must be returned to the Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

Application Server due to lack of pre-paid funds and the caller given the chance to “recharge” their calling card. This requires a well-written script application to intelligently handle the unpredictable range of responses, as well as database read/write capability. It requires the IVR to have complete call control for reestablishing calls in progress, and providing prompts based on database information. It is important to note that the RTP media stream can be originated or terminated by the Application Server. A unified messaging server for instance can receive RTP media traffic from a gateway and perform many different operations on this media. The media can be stored as speech, and it can be retrieved as speech as in normal voice mail. It could also be converted to an email message or the Application Server could even perform speech recognition on the inbound media. Designs have been put in place for an entire speech driven web application; SIP will be used to access these very intelligent voice assistants and voice browsers. The call flow for such a call is shown below but it utilizes the same fundamental and simple equipment listed in the above application. Whatever control logic is used, such as XML or CPL, the only thing that is changed is the complexity of the call flow and the application. In this way a single SIP Application Server (or Server Farm) can manage to perform many application within a network, and have those application easily and simply changed-out or upgraded.

Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

CONCLUSION 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.Moreover,VoIP is suitable for computer telephony integration and other next generation applications. A VoIP network can centralize much of its intelligence so the management of the network can also be centralized and there is no longer a need to have diverse PBXs located around the country and around the world. A service provider can centralize the network operations and billing. Enterprises can also deploy an IP telephony system next to the existing PBX integrating it into a new and expanding IP telephony network. This allows them to leverage the existing PBX and all the expansion will be on the new IP Telephony system, while still maintaining a standard dial plan and seamless voice network. Branch offices can be swapped over to the IP telephony system when the existing PBXs are ready to be removed. VoIP will one day make voice communications an integrated element of rich communication experience that includes video and data. VoIP will no longer be a service but a technology that is utilized in an application that may run on a computer, a PDA or other information and communication appliance. Voice communication will be initiated via something like an e-mail address, hyper link or initiated within an application the application itself.VOIP is growing fast. The very knowledge of the applications of this technology is enough for users and manufacturers to flock towards it. It is ideal for computer based communications and at the same time bringing down the cost of multimedia transfer.

Dept of ECE

GEC,TRICHUR

Seminar 2004

VoIP-Voice over Internet Protocol

REFERENCES ™ Internet Telephony Conference &EXPO. ™ IEC:Voice over Internet Protocol. ™ “Voice over IP issues and challenges”,by Prof Raj Jain,Ohio State University. ™ “VoIP for next generation economical telephony” INFOTECH ™

“Advanced VoIP applications”,Glen Gerhard

Dept of ECE

GEC,TRICHUR