Chapter 1

Introduction

1

Chapter 1

Introduction

1.1 Introduction

IP telephony which is known as Voice over Internet Protocol in short VoIP is

expanding rapidly in every where day by day and keeping crucial effect on modern

science. VoIP provides cheep call rate and only for this advantage though it has some

limitation people are welcoming and accepting it in rapid manner and migrating from

old Public Switch Telephone PSTN system. The quality of service (QOS) of VoIP

still not reached the standard and desired level if we compare it to the old PSTN and

mobile services. But keeping mind of those limitation people is showing huge interest

on it only for cheep call rate. It is true and definitely unacceptable that VoIP is illegal

in some countries though their government never can prove any logical unfairness of

VoIP business. The truth is if VoIP can over come some of its limitation and meet

desirable and standard quality of service then both old PSTN and Mobile companies

will lose their market. In Bangladesh some very few companies got VoIP license. In

science point of view it is expected that government of Bangladesh will give more

VoIP licenses. As VoIP is networking it is good opportunity for a student to enrich

both his networking and VoIP knowledge. Internship opportunity on VoIP research

can be considered a great opportunity for a student to learn some most important and

effective skills.

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1.2 Origin of the Report

As a compulsory part of Internship program, this particular report is being

prepared by the author on the proposed topic “Voice over Internet Protocol (VoIP) and its

configuration in Session Border Controller Switch (SBC)”. The intention was to give an

opportunity to the students to gain some real world experience by working in a practical

environment. The internship supervisor was Mobassher Hossain Chowdhury, Networking

Engineer (NOC), Alternate Access BD. Pvt. Ltd. and the faculty advisor was Mohammad

Saiful Islam Mamun, Lecturer, Computer Science and Engineering department, Stamford

University Bangladesh.

1.3 Objective

The Primary Objective of this report is:

To explore all about Voice over IP.

To exhibit how VoIP use networking concepts.

To show the networking is expanding and keeps vital and crucial effect on

modern science.

To measure the gap between theoretical and practical of Networking and Voice

over IP.

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1.4 Background of the company

Alternate Access Bd. Pvt. Ltd. is a Call Center. It is a branch office and

situated in New DOHS Mohakhali, Dhaka, Bangladesh. The main office is in New York,

USA and name of the main company is Rapid Target Services. The mother company

Rapid Target Service was founded in April, 2003. It offers telecom services to the

prepaid phone card industry as well as being a serious player in the wholesale field of

telecommunications. Headquarter of Rapid Target Services is in Manhattan, NY. RTS is

an established leader in the telecom field, specializing in direct termination. With its

proficient infrastructure, RTS has the ability to offer its clients fast and efficient service

regardless where the client is located.

Since Rapid Target Services entered the telecom market in 2003, it has

become known worldwide as one of the most aggressive and quality-driven Direct

Terminators in the market. RTS routes millions of minutes through its own switching and

network infrastructure and allows other carriers to take advantage of its competitive rates

and global buying power. What sets RTS aside from other carriers is the fact that they

build their own termination networks. Unlike the vast majority of companies in the VoIP

industry, RTS doesn't simply re-sell capacity from other carriers, it primarily sells its

own. Consequently, RTS can offer truly aggressive pricing and network flexibility.

RTS services a diverse customer base of over 50 carriers, including calling

card vendors,

Retail operators, major wholesalers and a number of PTTs. RTS focuses on providing

capacity to niche markets such as Asia, the Middle East and Africa, but also has

termination worldwide. RTS’s goal as a carrier’s carrier is to offer their customers the

most aggressive pricing in the market while maintaining the same quality and service that

the customer is accustom to receiving from their wholesale providers.

4

Chapter 2

VoIP

5

Chapter 2

VoIP

2.1 Voice over Internet Protocol (VoIP)

Voice over Internet Protocol (VoIP) is simply the transmission of voice traffic

over IP-based networks. The Internet Protocol (IP) was originally designed for data

networking. The success of IP in becoming a world standard for data networking has led

it to adapt in voice networking. VoIP is also known as Internet telephony and helps the

conversion of voice into a digital signal that can be sent over the Internet. It is the

integration of conventional telephone services with the growing number of other IP-based

applications. VoIP is the transmission of voice signals as packets of data using IP. In

VoIP the voice information is sent in digital form in discrete packets rather than by the

traditional circuit-committed architecture of PSTN. Before transmitting, voice signals are

digitized with an Analog to Digital Converter, and then transmitted over the networks as

digitized packets, finally at the destination they are transformed back to analog format

with a Digital to Analog Converter. [1]

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2.2 Advantages of VoIP

The main secret behind this rapidly emerging VoIP is for its lower cost. With this

lower cost benefit it gives some other highly noticeable advantages as follows:

2.2.1 Lower Cost

No other telephone services that have invented can give such lower cost relief like

VoIP. VoIP has become popular largely because of the cost advantages to consumers

over traditional telephone networks. The general scenario for a traditional telephone

customer is that they pay a monthly fee for service, per call charge for local telephone

calls and a per-minute charge for long-distance calls. On the other hand the general

scenario for VoIP customer is that they don’t need to pay monthly service charge or

subscription charge and what they need to pay is per-second lower charge for both local

and long calls. [2]

For example, we know there is a good Bangladeshi man-power in Saudi Arab.

Say their relative who is in Bangladesh want to communicate with them. If that relative

want to give call using land telephone or mobile phone he have to pay say 15 taka or

more in case of mobile. But if he uses VoIP call he will be charge for only 3 taka per

minute with using per second pulse. Yes in some cases voice quality may be poor but if

people can talk only 3 taka rather than 15 taka then they will not consider that poor voice

quality.

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2.2.2 More efficient use of bandwidth

VoIP compresses data packets during transmission, and this causes more data to

be handled over the carrier. As a result, more calls can be handled on one access line.

It is known that about 50 % of a voice conversation is silence. VoIP fills the

‘empty’ silence spaces with data so that bandwidth in data communication channels is not

wasted. In other words, a user is not given bandwidth when he is not talking, and this

bandwidth is used efficiently for other bandwidth consumers. Moreover, compression and

the ability to remove redundancy in some speech patterns add up to the efficiency.

Traditional phone calls work by allocating an entire phone line to each call. With

VoIP, voice data is compressed and transmitted over a computer network. This means

VoIP uses substantially less bandwidth than a traditional telephone call and is

consequently more cost effective.

2.2.3 Flexibility

VoIP telephony enables the user to integrate computer applications like email, fax

and web conferencing, with the telephone. It gives the flexibility of using the phone,

while accessing all the other programs, and surfing the Internet at the same time. This

way, you'll save more money and energy as it combines all of the services into one basic

application.

It is also possible for the users to take the VoIP adapters anywhere, and use their

number at any place which offers an Internet connection. This feature is particularly

helpful for those with active lifestyles and whose jobs require lot of travel. [3]

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2.2.4 Cheap user hardware and software

If you are an Internet user wishing to use VoIP for voice communication, the only

additional hardware you require besides your computer and Internet connection are a

sound card, speakers and a microphone. These are quite cheap. There exist several

software packages downloadable from the Internet, which you can install and use for the

purpose. Examples of such applications are the well-known Skype and Net2Phone. You

do not actually need a telephone set, which can be quite expensive, along with the

underlying equipment, especially when you have a phone network.

The installation process for VoIP phones is very simple, and once done, high

mobility of the system is an added advantage. The hassles of separate cabling for

telephone systems can also be avoided by using this technology. The infrastructure of the

whole system is very scalable and new components can be added easily without much

difficulty. As the transfer of voice, which is converted into signal, is based on software,

rather than hardware, it is easier to alter and maintain the whole system. All these

attributes makes VoIP more popular, as one does not have to be very good at computers.

2.2.5 Cheap, Simple and Scalable Infrastructure

VoIP has 3 sites. All sites need router to route calls. There are many VoIP

supported router of different companies. Cost for set up VoIP services depend upon

router. Though approximately 6 to 10 lakhs taka in Bangladeshi currency is god enough

to set up VoIP route. Definitely it is huge cheaper than to set up all infrastructure of a

mobile or land telephone company.

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Configuration of router and all VoIP is easy and user friendly. Without CISCO

other routers are configured by using graphical user interface (GUI). So it is very much

simple to configure. You just need to have basic computer, networking and VoIP

knowledge without being expert.

You need to connect your router to Digital Subscriber Line (DSL) or high speed

broad band and to Sim server and finally you have to configure it. Al these are VoIP set

up. So it is little, simple and Scalable infrastructure and easy to maintenance.

In case of user, they just need to set up VoIP dialer software in their PC like SIP

phone, Net phone, VoIP phone etc. All these can be downloaded free from internet. [4]

2.2.6 Efficient Use of Network Resources

Internet is a packet switched or connectionless network, the individual packets of

each voice signal travel over separate network paths for reassembly in proper sequence at

their destination. These make more efficient use of network resources and provide more

reliability than the circuit switched PSTN. [5]

2.27 High Fidelity Voice Transmission

Internet has no 64 kbps bandwidth limitation, so it can provide high fidelity voice

transmissions very easily.

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2.3 Disadvantages of VoIP

If VOIP is starting to sound really good to you, make sure you understand the

following downsides as well.

2.3.1 No service during a power outage:

During a power failure a regular phone is kept in service (unless it is a cordless)

by the current supplied through the phone line from the central office. This is not possible

with IP phones, so when the power goes out, there is no VOIP phone service. In order to

use VoIP during a power outage, an uninterruptible power supply or a generator must be

installed on the premises.

2.3.2 VoIP Voice Quality:

As VoIP uses an Internet connection, its quality of services (QoS) depends upon your

broadband connection, your hardware, the service provided by your provider etc. The

measurement of quality of services (QoS) depends up on some factor. These factors are:

Latency

Jitter

Packet loss

More and more people are enjoying high quality of phone calls using VoIP, but still

many users complain of hearing noise, echo and having to wait a lot before hearing an

answer etc. These distortions, noises, echoes, lost of some conversations are accused of

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transmission errors. This is more noticeable in highly congested networks and/or where

there are long distances and/or internetworking between end points.

Some kind of stability in Internet data transfer needs to be guaranteed before VoIP could

truly replace traditional phones. [6]

2.3.3 Reliability:

Another issue associated with VoIP is having a phone system dependant on

individual PCs of varying specifications and power. A call can be affected by processor

drain. Let's say you are talking away on your softphone, and you decide to open a

program that saps your processor. Quality loss will become immediately evident. In a

worst case scenario, your system could crash in the middle of an important call. In VoIP,

all phone calls are subject to the limitations of normal computer issues. [7]

2.3.4 Emergency calls:

Another major concern with VOIP involves emergency 911 calls. Traditional

phone equipment can trace your location. Emergency calls are diverted to the nearest call

center where the operator can see your location in case you can't talk. However, because

a voice-over-IP call is essentially a transfer of data between two IP addresses, not

physical addresses, with VOIP there is currently no way to determine where your VOIP

phone call is originating from. [8]

2.3.5 More dial:

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In VoIP sometimes you may need to dial more than usual. Say you are calling Sri

Lanka from America. The router which acts as a gateway in Sri Lanka can’t access more

than 16 calls at a time. That means there is no available channel for Sri Lanka gateway to

take any call. In that real time if you try to call, your call will be rejected by the gateway.

This case will be continued before any channel is available. So user may have to dial

more than usual and this is another drawback.

2.3.6 .Security:

This one is the last in this list, but it is not the least! Security is a main concern

with VoIP, as it is with other Internet technologies. The most prominent security issues

over VoIP are identity and service theft, viruses and malware, denial of service,

spamming, call tampering and phishing attacks.

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Chapter 3

Development Challenges of VoIP

14

Chapter 3

Development Challenges of VoIP

3.1 Development Challenges of VoIP

The goal of VoIP developers is to add telephone-calling capabilities to IP based

networks and interconnect these to the public telephone network and to private voice

networks, in such a way as to maintain current voice standards and preserve the features

everyone expects from the telephone. VoIP development needs to take place in five

specific areas. [9]

Voice quality should be comparable to PSTN, even over networks having variable

levels of QoS.

IP networks must meet strict performance requirements and criteria including

minimizing call refusals, network latency, packet loss and disconnects. This is

required even when there is heavy congestion in the network or when resources

have to be shared among multiple users.

Call control and signaling should be transparent; the users should be unaware of

what technology is actually implementing the service.

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PSTN / VoIP service internetworking and equipment interoperability between the

voice and data network environments should not affect the QoS and reliability.

System management, security, addressing and accounting must be provided,

preferably consolidated with the PSTN operation support systems.

3.2 Quality of Service (QoS) issues in VoIP networks

The advantages of reduced cost and bandwidth savings of carrying voice over

data networks are associated with some QoS issues unique to packet networks.

Delivering quality voice signals from one point to another cannot be considered

successful unless the quality of the delivered signal satisfies the recipient. Providing a

level of quality that at least equals the PSTN is viewed as a basic requirement. Although

QoS usually refers to the fidelity of the transmitted voice and facsimile document it can

also be applied to network availability, telephone feature availability and scalability etc.

[10] Many factors have been identified those play big roles in determining the quality of

service of VoIP. They are as follows:

3.2.1 Delay

VoIP delay or latency is characterized as the amount of time it takes for speech to exit

the speaker’s mouth and reach the listener’s ear. The following are the source of delay in

an end-to-end VoIP call.

Accumulation delay, is also called algorithmic delay, is caused by the time need

to collect a frame of voice samples to be processed by the voice coder. This

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depends on the type of voice coder used and varies from a single sample time,

which is .125 ms to many ms. [11]

Processing delay is caused by the actual process of encoding and collecting the

encoded samples into a packet for transmitting over the IP network. The encoding

delay is a function of both the processor execution time and the type of algorithm

used. To reduce this delay often multiple voice coder frames are collected in a

single packet. [12]

Network delay is caused by the physical medium and protocol used to transmit

the voice and data, and by the buffers used to remove packet jitter on the received

side. This delay is a function of the capacity of the links in the network as well as

the processing that happens as the packets pass through the network. [13]

Two problems that result from high end-to-end delay in a voice network is echo and

talk over. Echo is caused by signal reflections of the speaker’s voice from the far end

telephone equipment back into the speaker’s ear. Talkers’ overlap is the problem of one

caller stepping on the other talker’s speech. This becomes significant if the one-way

delay becomes greater than 150 ms.

3.2.2 Jitter

Jitter is the variation of packet inter-arrival time. In a packet voice environment

the sender is expected to reliably transmit voice packets at a regular interval. These voice

packets can be delayed through out the packets network and may not arrive at the same

regular interval at the receiving station. The jitter buffers used to remove the packet delay

variation that each packet experiences as it transmit the packet also adds delay. This

delay can be a significant part of the over all delay, as packet delay variations can be a

high as 70 ms to 100 ms in IP networks. [14] Removing jitter requires collecting packets

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and holding them long enough to allow the slowest packets to arrive in the time to be

played in the correct sequence which in turn causes additional delay. The conflicting

goals of minimizing delay and removing jitter has led to the development of various

schemes to adapt the jitter buffer size to match the time varying requirements of network

jitter removal. Well-engineered dynamic jitter buffer is the best mechanism to use for

packet based voice network. Static jitter buffers forces the jitter buffers to be either too

large or too small, thereby causing the audio quality to suffer, due to either lost packets or

excessive delay. The problem is that the packet loss rate differs even given the same

delay jitter conditions according to the implementation of the jitter buffers of terminals.

3.2.3 Echo

One of the main problems of a very big end-to-end delay is the problem of

echoes. Echo is the reflected copy of the voice heard some time later and a delayed

version of the original. This happens whenever the round-trip delay exceeds 50 ms. [16]

Round-trip delay time is the time required for a signal pulse or packet to travel from a

specific source to a specific destination and back again.

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Figure 3.1: Sources of Echo in PSTN

There are mainly two types of echoes, hybrid echo and acoustic echo.

3.2.3.1 Acoustic echo:

An acoustic echo is one of the simplest acoustic modeling problems. Echoes

occur when a sound arrives via more than one acoustic propagation path, as shown in Fig.

We may hear a discrete echo, for example, if we clap our hands standing in front of a

large flat wall outdoors, such as the side of a building. To be perceived as an echo,

however, the reflection must arrive well after the direct signal (or previous echo). [17]

Figure 3.2: Geometry of an acoustic echo caused by multi path

propagation. A direct signal and a floor bounce are received from

the source S at the listening point L.

Acoustic echo arises when sound from a loudspeaker for example; the earpiece of a

telephone handset is picked up by the microphone in the same room. The problem exists

in any communications scenario where there is a speaker and a microphone. Examples of

acoustic echo are found in everyday surroundings such as:

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Hands-free car phone systems

A standard telephone or cell phone in speakerphone or hands-free mode

Dedicated standalone "conference phones"

3.2.3.2 Hybrid echo:

Hybrid echo is generated by the public switched telephone network (PSTN) through

the reflection of electrical energy by a device called a hybrid (hence the term hybrid

echoes). Most telephone local loops are two-wire circuits while transmission facilities are

four-wire circuits. Each hybrid produces echoes in both directions, though the far end

echo is usually a greater problem for voice band.

Echo in a telephone network is acceptable because the round-trip delays through the

network are smaller than 50 ms. Echoes are a problem in VoIP as the round-trip delays

are almost always greater than 50 ms. [18]

3.2.4 Packet Loss

Packet loss in data networks is both common and expected. Like data network

VoIP network cannot provide a guarantee that packets will be delivered at all. Packet will

be dropped under peak loads and during periods of network congestion. But due to the

time sensitivity of voice transmission, the normal TCP based transmission scheme are not

suitable. When putting voice on data networks it is important to build a network that can

successfully transport voice on a reliable and timely manner. When putting critical traffic

on data networks, it is important to control the amount of packet loss in that network.

VoIP network packet loss introduces audio distortions that cause VoIP quality to

decrease as the rate of packet loss increases. In a particular connection this general effect

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can be modeled by the distribution of the lost packets and by the packet loss concealment

algorithm. [19]

In voice communications, packet loss shows up in the form of gaps or periods of

silence in the conversation, thus leading to a "clipped speech" effect that is unsatisfactory

for most users and unacceptable in business communications.

3.2.5 Bandwidth Availability

Bandwidth is the portion of the network that is available to an application to

transfer information on the network. In case of VoIP the level of reliability and sound

quality that is acceptable among users has not yet been reached and this is primarily

because of bandwidth limitations and this also leads to packet loss.

Numbers of VoIP subscribers are rapidly increasing if new calls are accepted

without limit in the VoIP network the total bandwidth requirement may exceed the

network capacity. In that case QoS (packet loss ate, delay etc) for calls in process may be

worse. Therefore a mechanism called call admission control is necessary to reject a new

call when enough network spare capacity is not available. The necessity of admission

control in the connectionless IP networks is similar to the circuit switched networks,

although the function and the implementation may be different.[20]

3.2.6 Voice Activity Detection (VAD)

In normal voice conversations, some one speaks and someone listen, at least 50

percent of total bandwidth is wasted. VoIP networks utilize the wasted bandwidth by

enabling VAD. VAD experiences some inherent problems in determining when speech

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ends and begins, and in distinguishing speech from background noise. So it is expected to

develop VAD protocol more reliable way to proper use of bandwidth. [21]

3.2.7 Tandem Coding

Typically where networks join, the speech traffic is passed as 64 kbps PCM. If the

originating system has coded the speech in another format, a decodec is required. This

results in a transcoding or tandeming of speech codecs. Generally the quality of the

combination cannot be better than the poorest link, and may be noticeable worse if two or

more low bit rate codecs are included. The order is also important, because these systems

distort speech in a nonlinear way. G.729 followed by G.711 will not produce exactly the

same quality as G.711 followed by G.729. Delay also increases significantly with tandem

coding. These problems can be avoided by tandem free operation; where the systems

negotiate a common codec which is used end to end. [22]

3.3 Solutions for the Major Problems of VoIP

In VoIP the term QoS mainly depends upon the bandwidth capacity in network. The

more free bandwidth ensure more better quality of voice though its difficult to get such

expected bandwidth today because of huge internet user. So there should some way some

technique some protocol to ensure and improve quality of service of VoIP in this limited

bandwidth network. The techniques and solutions to the major problems of VoIP are as

follows.

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A way to improve speech quality is to implement some kind of echo cancellation

mechanism. Echo cancellation is not required on short haul PSTN connections.

But if a VoIP system connects to a local PSTN, echo cancellation is probably

needed to cancel the local hybrid reflections. If the system does not connect to a

local PSTN, echo cancellation should still be included to remove any acoustic

echo.

The task of solving the problem of jitter in VoIP networks has two conflicting

goals. These are minimizing delay and removing jitter. This has led to the

development of various schemes to adapt the jitter buffer size to match the time

varying requirements of network jitter removal. These schemes have the explicit

goal of minimizing the size and delay of the jitter buffer while at the same time

preventing buffer underflow caused by jitter. One approach, which is used in IP

networks to adapt the jitter buffer, 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 in turn used to adjust the jitter buffer to

target a predetermined allowable late packet ratio. [23]

Schemes called lost packet compensation schemes used by VoIP to overcome the

problem of lost packets. Interpolate for lost speech packets by replaying the last

packet received during the interval when the last packet was supposed to be

played out. This works well when the incidence of lost frames is infrequent. It

does not work very well for continuously series loss of packets. Another way is to

send redundant information at the expense of bandwidth utilization. The 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 exactly correct

for the lost packet. However, this approach uses more bandwidth and creates

greater delay. An alternative approach is to develop an algorithm in the digital

23

signal processor that detects missing packets, and then replays the last

successfully received packet at a decreased volume in order to fill the gaps. [24]

Allowing limited users in the networks minimizes problem of bandwidth

availability. Protocols are also being developed to overcome many QoS issues of

VoIP networks.

Chapter 4

Internal Architecture of VoIP

24

Chapter 4

Internal Architecture of VoIP

4.1 Internal Architecture of VoIP

The internal architecture of VoIP defines how analog voice will be converted into

digital signal, how voice will be tag into packet and what are the protocols used by VoIP

to make a call and terminate the call. To setup a VoIP communication we need:

1. First the ADC to convert analog voice to digital signals (bits)

2. Now the bits have to be compressed in a good format for transmission: there are a

number of protocols we'll see after.

3. Here we have to insert our voice packets in data packets using a real-time protocol

(typically RTP over UDP over IP)

4. We need a signaling protocol to call users: H323 and SIP both do that.

5. At RX we have to disassemble packets, extract data, then convert them to analog

voice signals and send them to sound card (or phone)

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6. All that must be done in a real time fashion because we cannot waiting for too

long for a vocal answer.

Voice)) ADC à Compression Algorithm à Assembling RTP in UDP/IP à

----> |

<---- |

Voice((DAC ß Decompression Algorithm ß Disassembling RTP from UDP/IP ß

Figure 4.1: Basic internal architecture of VoIP

4.2 Analog to Digital Conversion

This is made by hardware, typically by card integrated ADC. Today every

sound card allows you convert with 16 bit a band of 22050 Hz (for sampling it you need a

freq of 44100 Hz for Nyquist Principle) obtaining a throughput of 2 bytes * 44100

(samples per second) = 88200 Bytes/s, 176.4 kBytes/s for stereo stream. [25]

For VoIP we needn't such a throughput (176kBytes/s) to send voice packet: next we'll see

other coding used for it.

4.3 Codecs used by VoIP

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Voice over Internet Protocol (VoIP) is a group of communication technologies

used to transfer and deliver voice data through IP networks such as the internet. It

converts analog audio signals into digital form for transmission and back again into an

audio signal for repetition. Internet telephony is a service that utilizes VoIP. When

making a call using Internet telephony, the analog voice signal is altered to digital format,

which is then compressed into IP packets for transmission via the internet. The entire

process is inverted at the recipients end. Internet telephony covers audio, video, fax and

text.

The VoIP codec is the actual algorithm that is used to convert voice data into

digital data, compression of the digital data into IP packets to save bandwidth, and

decompression of data once it has reached its intended recipient or target location.

Codecs perform the conversion by replicating the audio data a few thousand times a

second. For example, a G.722 codec samples audio data 16000 times a second. This is

known as the sampling rate. Sampling rates differ from codec to codec. Bandwidth usage

increases with higher rates of sampling. The most commonly used codec in VoIP is the

G.729 which has a sampling rate of 8000 times per second. [26]

The table below lists the available VoIP Codecs.

Codec Bandwidth/kbps Comments

G.711 64 Delivers precise speech transmission. Very low processor

requirements. Needs at least 128 kbps for two-way.

G.722 48/56/64 Adapts to varying compressions and bandwidth is

conserved with network congestion.

G.723.1 5.3/6.3 High compression with high quality audio. Can use with

dial-up. Lot of processor power.

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G.726 16/24/32/40 An improved version of G.721 and G.723 (different from

G.723.1)

G.729 8 Excellent bandwidth utilization. Error tolerant. License

required.

GSM 13 High compression ratio. Free and available in many

hardware and software platforms. Same encoding is used

in GSM cell phones (improved versions are often used

nowadays).

Table 4.1: VoIP Codecs

4.4 Compression Algorithms

Now we have the raw data and we want to encapsulate it into TCP/IP stack. We

follow the structure:

VoIP data packets

RTP

UDP

IP

VoIP data packets live in RTP (Real-Time Transport Protocol) packets which

are inside UDP-IP packets.

Firstly, VoIP doesn't use TCP because it is too heavy for real time applications,

so instead a UDP (datagram) is used.

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Secondly, UDP has no control over the order in which packets arrive at the

destination or how long it takes them to get there (datagram concept). Both of these are

very important to overall voice quality and conversation quality. RTP solves the problem

enabling the receiver to put the packets back into the correct order and not wait too long

for packets that have either lost their way or are taking too long to arrive (we don't need

every single voice packet, but we need a continuous flow of many of them and ordered).

4.5 RTP Real Time Transport Protocol

RTP is designed for end-to-end, real-time, transfer of multimedia data. The

protocol provides facility for jitter compensation and detection of out of sequence arrival

in data that are common during transmissions on an IP network. RTP supports data

transfer to multiple destinations through multicast. RTP is regarded as the primary

standard for audio/video transport in IP networks and is used with an associated profile

and payload format.

Multimedia applications need timely delivery and can tolerate some loss in

packets. For example, loss of a packet in audio application may result in loss of a fraction

of a second of audio data, which can be made unnoticeable with suitable error

concealment algorithms. Multimedia applications require timeliness over reliability. The

Transmission Control Protocol (TCP), although standardized for RTP use (RFC 4571), is

not often used by RTP because of inherent latency introduced by connection

establishment and error correction, instead the majority of the RTP implementations are

built on the User Datagram Protocol (UDP). [27]

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Protocol components

The RTP specification describes two sub-protocols:

The data transfer protocol, which deals with the transfer of real-time multimedia

data. Information provided by this protocol includes timestamps (for

synchronization), sequence numbers (for packet loss detection) and the payload

format which indicates the encoded format of the data.

The Real Time Control Protocol (RTCP) is used to specify Quality of Service

(QOS) feedback and synchronization between the media streams. The bandwidth

of RTCP traffic compared to RTP is small, typically around 5%. [28]

4.5.1 RTP Packet header

The RTP header has a minimum size of 12 bytes. After the header, optional header

extensions may be present. This is followed by the RTP payload, the format of which is

determined by the particular class of application. The fields in the header are as follows:

Figure4.2: RTP Packet Header

bit

offset

0-1 2 3 4-7 8 9-15 16-31

0 Ver. P X CC M PT Sequence Number

32 Timestamp

64 SSRC identifier

96 CSRC identifiers (optional)

30

Ver.: (2 bits) Indicates the version of the protocol. Current version is 2.

P (Padding): (1 bit) Used to indicate if there are extra padding bytes at the end of

the RTP packet. Padding might be used to fill up the block of certain size, for

example as required by an encryption algorithm.

X (Extension): (1 bit) Indicates presence of an Extension header between

standard header and payload data. This is application or profile specific.

CC (CSRC Count): (4 bits) Contains the number of CSRC identifiers (defined

below) that follow the fixed header.

M (Marker): (1 bit) Used at the application level and defined by a profile. If it is

set, it means that the current data has some special relevance for the application.

PT (Payload Type): (7 bits) Indicates the format of the payload and determines

its interpretation by the application. This is specified by an RTP profile.

Sequence Number (16 bits): The sequence number is incremented by one for

each RTP data packet sent and is to be used by the receiver to detect packet loss

and to restore packet sequence. The RTP does not take any action when it sees a

packet loss, but it is left to the application to take the desired action. For example,

video applications may play the last known frame in place of the missing frame.

Timestamp (32 bits): Used to enable the receiver to play back the received

samples at appropriate intervals. When several media streams are present, the

timestamps are independent in each stream, and may not be relied upon for media

synchronization. The granularity of the timing is application specific. For

example, an audio application that samples data once every 125 µs could use that

value as its clock resolution. The clock granularity is one of the details that are

specified in the RTP profile or payload format for an application. [29]

SSRC (32 bits): Synchronization source identifier uniquely identifies the source

of a stream. The synchronization sources within the same RTP session will be

unique.

CSRC: Contributing source IDs enumerate contributing sources to a stream

which has been generated from multiple sources.

31

Extension header: (optional) The first 32-bit word contains a profile-specific

identifier (16 bits) and a length specifier (16 bits) that indicates the length of the

extension (EHL=extension header length) in 32-bit units, excluding the 32 bits of

the extension header.

4.6 H323 Protocols

H.323 is a standard that specifies the components, protocols and procedures that

provide multimedia communication services (real-time audio, video, and data

communications) over packet networks, including Internet protocol (IP) based networks.

H.323 is part of a family of ITU-T recommendations called H.32x that provides

multimedia communication services over a variety of networks.

The H.323 standard is a cornerstone technology for the transmission of real-time audio,

video, and data communications over packet-based networks. It specifies the

components, protocols, and procedures providing multimedia communication over

packet-based networks. Packet-based networks include Internet Protocol (IP) based

(including the Internet) or Internet packet exchange (IPX) based local-area networks

(LANs), enterprise networks (ENs), metropolitan-area networks (MANs), and wide area

networks (WANs). H.323 can be applied in a variety of mechanisms: audio only (IP

telephony); audio and video (videotelephony); audio and data; and audio, video and data.

H.323 can also be applied to multipoint-multimedia communications. H.323 provides

myriad services and, therefore, can be applied in a wide variety of areas: consumer,

business, and entertainment applications. [30]

32

Figure4.3: H.323 Terminals on a Packet Network

4.7 Internetworking with Other Multimedia Networks

The H.323 standard specifies four kinds of components, which, when

networked together, provide the point-to-point and point-to-multipoint multimedia-

communication services:

terminals

gateways

gatekeepers

multipoint control units (MCUs)

4.7.1 Terminals

Used for real-time bidirectional multimedia communications, an H.323 terminal

can either be a personal computer (PC) or a stand-alone device, running an H.323 and the

multimedia applications. It supports audio communications and can optionally support

video or data communications. Because the basic service provided by an H.323 terminal

is audio communications, an H.323 terminal plays a key role in IP telephony services. An

H.323 terminal can either be a PC or a stand-alone device, running an H.323 stack and

multimedia applications. The primary goal of H.323 is to interwork with other

multimedia terminals. H.323 terminals are compatible with H.324 terminals on SCN and

wireless networks, H.310 terminals on B-ISDN, H.320 terminals on ISDN, H.321

terminals on B-ISDN, and H.322 terminals on guaranteed QoS LANs. H.323 terminals

may be used in multipoint conferences. [31]

33

4.7.2 Gateways

A gateway connects two dissimilar networks. An H.323 gateway provides

connectivity between an H.323 network and a non-H.323 network. For example, a

gateway can connect and provide communication between an H.323 terminal and SCN

networks (SCN networks include all switched telephony networks, e.g., public switched

telephone network [PSTN]). This connectivity of dissimilar networks is achieved by

translating protocols for call setup and release, converting media formats between

different networks, and transferring information between the networks connected by the

gateway. A gateway is not required, however, for communication between two terminals

on an H.323 network.

4.7.3 Gatekeepers

A gatekeeper can be considered the brain of the H.323 network. It is the focal point for

all calls within the H.323 network. Although they are not required, gatekeepers provide

important services such as addressing, authorization and authentication of terminals and

gateways; bandwidth management; accounting; billing; and charging. Gatekeepers may

also provide call-routing services.

4.7.4 Multipoint Control Units

MCUs provide support for conferences of three or more H.323 terminals. All

terminals participating in the conference establish a connection with the MCU. The MCU

manages conference resources, negotiates between terminals for the purpose of

determining the audio or video coder/decoder (CODEC) to use, and may handle the

media stream. The gatekeepers, gateways, and MCUs are logically separate components

of the H.323 standard but can be implemented as a single physical device.

34

Figure 4.4: H.323 Zone

4.8 Session Initiation Protocol (SIP)

The Session Initiation Protocol (SIP) is a signaling protocol, widely used for

controlling multimedia communication sessions such as voice and video calls over

Internet Protocol. The protocol can be used for creating, modifying and terminating two-

party or multiparty sessions consisting of one or several media streams. The modification

can involve changing addresses or ports, inviting more participants, adding or deleting

media streams, etc. Other feasible application examples include video conferencing,

streaming multimedia distribution, instant messaging, presence information and online

games.

The SIP protocol is a TCP/IP based Application Layer protocol. SIP is designed to be

independent of the underlying transport layer. It can run on Transmission Control

Protocol (TCP) and User Datagram Protocol (UDP) incorporating many elements of the

35

Hypertext Transfer Protocol (HTTP) and the Simple Mail Transfer Protocol (SMTP).

[32]

4.9 Design of SIP

Protocol:

The design of SIP elements is similar to HTTP like request and response

transaction model. Each transaction consists of a client request that follows a particular

method or function on the server and at least one response. SIP uses most of the header

fields, encoding rules and status codes of HTTP and provides a readable text based

format.

SIP works in concert with several other protocols and is only involved in the signaling

portion of a communication session. SIP clients typically use TCP or UDP on port

numbers 5060 and/or 5061 to connect to SIP servers and other SIP endpoints. Port 5060

is commonly used for non-encrypted signaling traffic whereas port 5061 is typically used

for traffic encrypted with Transport Layer Security (TLS). SIP is primarily used in setting

up and tearing down voice or video calls. The voice and video stream communications in

SIP applications are carried over another application protocol, the Real-time Transport

Protocol (RTP). Parameters (port numbers, protocols, codec) for these media streams are

defined and negotiated using the Session Description Protocol (SDP) which is transported

in the SIP packet body. [33]

A motivating goal for SIP was to provide a signaling and call setup protocol for IP-based

communications that can support the call processing functions and features present in the

public switched telephone network (PSTN). SIP by itself does not define these features;

rather, its focus is call-setup and signaling. However, it was designed to enable the

construction of functionalities of network elements designated proxy servers and user

agents. These are features that permit familiar telephone-like operations: dialing a

36

number, causing a phone to ring, hearing ringback tones or a busy signal. Implementation

and terminology are different in the SIP world but to the end-user, the behavior is similar.

4.10 SIP Entities:

A SIP network is composed of four types of logical SIP entities. Each entity has

specific functions and participates in SIP communication as a client (initiates requests),

as a server (responds to requests), or as both. One physical device can have the

functionality of more than one logical SIP entity. For example, a network server working

as a Proxy server can also function as a Registrar at the same time.

Following are the four types of logical SIP entities:

User Agent: In SIP a User Agent (UA) is the endpoint entity. User Agents initiate

and terminate sessions by exchanging requests and responses. RFC 2543 defines

the User Agent as an application, which contains both a User Agent client and

User Agent server, as follows:

User Agent Client (UAC): A client application that initiates SIP requests.

User Agent Server (UAS): A server application that contacts the user when a SIP

request is received and that returns a response on behalf of the user. Some of the

devices that can have a UA function in a SIP network are: workstations, IP-

phones, telephony gateways, call agents, automated answering services.

Proxy Server: A Proxy Server is an intermediary entity that acts as both a server

and a client for the purpose of making requests on behalf of other clients.

Requests are serviced either internally or by passing them on, possibly after

translation, to other servers. A Proxy interprets, and, if necessary, rewrites a

request message before forwarding it.

37

RFC 3261 defines Proxy Server as:

A proxy server "is an intermediary entity that acts as both a server and a client

for the purpose of making requests on behalf of other clients. A proxy server

primarily plays the role of routing, which means its job is to ensure that a

request is sent to another entity "closer" to the targeted user. Proxies are also

useful for enforcing policy (for example, making sure a user is allowed to make

a call). A proxy interprets, and, if necessary, rewrites specific parts of a request

message before forwarding it." [35]

Redirect Server: A Redirect Server is a server that accepts a SIP request, maps

the SIP address of the called party into zero (if there is no known address) or more

new addresses and returns them to the client. Unlike Proxy servers, Redirect

Servers do not pass the request on to other servers.

Registrar: A Registrar is a server that accepts REGISTER requests for the

purpose of updating a location database with the contact information of the user

specified in the request. A registrar is a server that accepts register requests and

places the information it receives in those requests into the location service for the

domain it handles.

4.11 SIP Messages

Message Types: There are two types of SIP messages:

Requests—sent from the client to the server.

Responses—sent from the server to the client.

Requests:

Method Description

INVITE Initiates a call, changes call parameters (re-INVITE).

38

ACK Confirms a final response for INVITE.

BYE Terminates a call.

CANCEL Cancels searches and “ringing”.

OPTIONS Queries the capabilities of the other side.

REGISTER Registers with the Location Service.

INFO Sends mid-session information that does not modify the

session state.

Table 4.2: Request Methods

Responses:

Response messages contain numeric response codes. The SIP response code set

is partly based on HTTP response codes. There are two types of responses and six

classes:

Response Type:

Provisional (1xx class): Provisional responses are used by the server to indicate

progress, but they do not terminate SIP transactions

Final (2xx, 3xx, 4xx, 5xx, 6xx classes): Final responses terminate SIP

transactions.

39

Classes

1xx = provisional, searching, ringing, queuing etc.

2xx = success

3xx = redirection, forwarding

4xx = request failure (client mistakes)

5xx = server failures

6xx = global failure (busy, refusal, not available anywhere)

Codes Description

100 Continue

180 Ringing

200 OK

300 Multiple choices

301 Moved permanently

302 Moved temporarily

400 Bad requests

401 Unauthorized

403 Forbidden

408 Request timeout

480 Unavailable

481 Call-leg/Transaction does not exist

482 Loop detected

5xx Server error

600 Busy

603 Declines

604 Does not exist

606 Not acceptable

Table 4.3: Response Code Examples

40

Figure 4.5: SIP call flow

41

Chapter 5

Physical Architecture of VoIP

Chapter 5

Physical Architecture of VoIP

5.1 Physical Architecture of VoIP

The architecture of VoIP can be devided by 3 parts. They are:

42

1. Session Border Controller.

2. Vendor Gateway

3. Customer Gateway

Generally if you consider according to the business logic then these 3 categories are

mainly 3 different business sites. In VoIP business some companies do businesses on

Session Border Controller, some other companies do business on Vendor Gateway and

rests are on Customer Gateway. The details of these three sites are here in below.

5.2 Session Border Controller

A session border controller (SBC) is a device used in Voice over Internet

Protocol (VoIP) networks to control over the signaling and usually also the media

streams involved in setting up, conducting and tearing down telephone calls.

SBCs usually sit between two service provider networks, they are customer

network that sends calls and vendor network which is the destination of that calls. SBCs

commonly maintain full session state and offer the following functions:

Security: Protect the network and other devices from attacks such as denial of

service.

Connectivity: Allow different parts of the network to communicate by for

example supporting NAT traversal.

Quality of service: The QoS policy of a network and prioritization of flows is

usually implemented by the SBC.

Regulatory: Many times the SBC is expected to provide support for regulatory

requirements such as emergency calls and lawful interception.

Statistics: Since all sessions that pass through the edge of the network pass

through the SBC, it is a natural point to gather statistics and information on these

sessions.

43

Within the context of VoIP, the term session refers to a call. Each call consists of one or

more call signaling message exchanges that control the call, and one or more call media

streams which carry the call's audio, video, or other data along with information of call

statistics and quality. Together, these streams make up a session. It is the job of a session

border controller to exert influence over the data flows of sessions.

The term border refers to a point of demarcation between one part of a network and

another. As a simple example, at the edge of a corporate network, a firewall demarcates

the local network (inside the corporation) from the rest of the Internet (outside the

corporation). A more complex example is that of a large corporation where different

departments have security needs for each location and perhaps for each kind of data. In

this case, filtering routers or other network elements are used to control the flow of data

streams. It is the job of a session border controller to provide administrative control in

managing the flow of session data across these borders.

Figure 5.1: Session Border Controller

The term controller refers to the influence that session border controllers have

on the data streams that comprise Sessions, as they traverse borders between one part of a

Session Border Controller SBC

Inbound Outbound

Calls Calls

44

network and another. Additionally, session border controllers often provide measurement,

access control, and data conversion facilities for the calls they control.

Customers send call to the customer gateway and that gateway sends calls to

SBC. Say one customer sends call to Sri Lanka from USA. He or she has dialed

9477245036 say. Session Border Controller (SBC) has its own database. It then compares

digits according to the country code. For that number it will get 94 is the destination of

Sri Lanka. Then it will compare the break out or carrier. Then it gets 77 is for carrier

Mtn. So the destination of this call is Sri Lanka mobile Mtn. And finally SBC will send

this call in that route which provides services to Sri Lanka mobile Mtn.

Figure 5.2: Call flow in SBC

Here in Figure call 9477245036 hits in SBC’s inbound port. SBC then compare from its

country database and got its Sri Lankan number. Then it will compare from Sri Lankan

carrier database and got this is Sri Lanka mobile Mtn’s number. Then it will send

appropriate gateway which provides service for Sri Lanka Mtn numbers.

45

5.3 Vendor Gateway

The role and activities of vendor gateway is to provide calls to the local

destination number. The gateway is a router which accepts calls from Session Border

Controller and then passes it to SIM server and finally from SIM server calls enter in

local network and reach to the destination number. The SIM server tags a local number

upon a call and spoofed the call as a local call. These procedures are used on those

countries where VoIPs are illegal. The gateway passes the call to the local network via E1

line in those countries where VoIPs are legal. The difference between E1 and SIM server

is that E1 provide much more better and reliable service than SIM server.

46

Figure 5.3: Vendor site

5.4 Customer Gateway

The customer gateway sends call to the Session border controller. In VoIP this

customer side is a big organization. These organizations provide VoIP services by selling

scratch card. Customers have to buy that scratch card to get balance or credit to give call

and to get prefix or access pin number. Customers then have to give call using prefix or

access pin number before the destination number. Some organization provides facilities

offering customer to buy balance or credit and access number by their credit card. User

can send call from PC or mobile. Some organizations provide both facilities to customer

and customer can send calls using PC or cell phone whatever they want.

Customer gateway has its own database. From database they compare calls are coming

from valid user or not. If its fake user then they discards that call else they pass that call

to the Session Border Controller. Usually they have contract with that organization which

provide services of Session Border Controller.

47

Figure 5.4: Call flow in Customer side

Here in figure a customer sends call using his or her prefix 07203 and then number

9477245036. The customer gateway then compare is it valid user or not if valid then the

user has enough balance or not. If all condition success then the gateway sends this call to

the Session Border Controller.

So finally if we look total VoIP call flow in a single view it will be like following figure:

48

Figure 5.5: VoIP Call Flow

Here in figure a customer from USA calls to a Sri Lankan number 9477245037 adding

prefix before the number. This call goes to the customer gateway. Customer gateway first

check the prefix to check the coming call is from valid user or fake user. If valid then it

checks the customer has enough credit to send this call. If all conditions meet then it

forward this call by replacing prefix 07203 to 292547 to the SBC. The reason for

replacing is this is the prefix that is used in SBC to identify it.

In SBC the call will first verified by SBC whether the calls come from authorized

customer or not. If it verified successfully then it look up country database table. Then it

get back 94 is for destination Sri Lanka. Then it looks to carrier database of Sri Lanka

and get 77 is a Sri Lanka mobile Mtn number. Then after calculating its routing database

finally it sends call to appropriate route by replacing its own prefix by null. Here it will

replace 292547 by null and send the call in format 9477245037.

In Vendor side the gateway will forward the call to SIM server. SIM server contains

some sim and it sends the call to local GSM network by the following format: Source:

9477241562 and destination 9477245037. And finally call reach to the destination

mobile.

5.5 Inbound and Outbound calls

Besides this some other information are noticeable in VoIP architecture. Those

are described in following:

49

Inbound Call: Calls those are coming in inbound ports are called inbound call.

Inbound ports just accept call. In gateway there are some inbound ports. Some

organization uses 10 ports some other 500 ports depending up on their business

policy.

Outbound Call: Calls those get out from outbound ports are called outbound

call. Outbound ports just pass call according to destination. In gateway there are

some outbound ports. Some organization uses 10 ports some other 500 ports

depending up on their business policy.

Figure 5.6: Inbound and Outbound calls

5.6 FXS (Foreign Exchange Subscriber)

FXS interface (the plug on the wall) delivers POTS service from the local

phone company’s Central Office (CO) and must be connected to subscriber equipment

(telephones, modems, and fax machines). In other words an FXS interface points to the

subscriber.

50

An FXS interface provides the following primary services to a subscriber device:

Dial Tone

Battery Current

Ring Voltage

5.7 FXO (Foreign Exchange Office)

FXO interface (the plug on the phone) receives POTS service, typically from a

Central Office of the Public Switched Telephone Network (PSTN). In other words an

FXO interface points to the Telco office.

An FXO interface provides the following primary service to the Telco network device:

on-hook/off-hook indication (loop closure)

Figure 5.7: FXO and FXS

51

Chapter 6

Business Policy

52

Chapter 6

Business Policy

6.1 Business Policy

It was said before that VoIP has 3 different sites. All they have different

business policy. Those policies and the way the make profits are describing here:

6.2 Customer Site

Customer pay bill to those company who provide Session Border Controller

services. They collect bill from customer. Customer site usually sell scratch card or

collect bill from user’s credit card.

53

Figure 6.1: Sample Customer Tariff in Session Border Controller

Here in figure we can see a customer Nobel Communication which is set up in

SBC as NBL and the name of its tariff is NBL Cust. We can see here in figure customer

NBL pay SBC 0.05 cent per minute for Pakistan mobile calls. Say they collect 0.15 cent

from user for per minute calls. So their profit is (0.15-0.05) 0.10 sec. per minute. So if a

user talks 10 minute to Pakistan customer Nobel Communication will get 100 cent or 1

dollar from this user.

6.3 Vendor Site

Vendor site collect bill from Session Border Controller site. SBC pay for how

many total minutes they use that vendor’s route. Vendors don’t need to pay anyone

except taxes to their government.

54

Figure 6.2: Sample Vendor Tariff in Session Border Controller

Here in figure we can see for Pakistan mobile Session Border Company like

Rapid Target Services have to pay 0.0385 for per minute.

6.4 Session Border Controller Site

Session border controller site like ours Rapid Targetr Services have to pay bill

to the vendor and collect money from customer. If we look back again to previous two

figures then we can see Rapid Target Services use route of Avia Phone Company for

destination to Pakistan. For using that route it has to pay 0.0385 cent per minute to Aviya

Phone Company. Rapid Target Services send customer Nobel Communication’s calls to

Aviya Phone Company’s route. For this Rapid Target Services collect 0.05 cent from

customer Nobel Communication. So the profit is (0.05 – 0.0385) 0.0115 for per minute.

55

Figure 6.3: Business policy used by SBC

To make theme clear say SBC pay 1 taka per minute to vendor for call charge and collect

2 Taka per minute from customer so the profit for SBC is 1 taka per minute.

56

Figure 6.4: Profit Report in SBC

Here in figure we can see all details of billing system in SBC switch. Client NBL uses

AVPC Pakistan mobile route and talked total 4307.27 minute. So at rate per minute 0.05

cent the cost comes (0.05 cent * 4307.27 minutes) = 215.3732 dollar. So that’s the

Revenue and customer NBL will give us 215.3732 dollar for the day 25th December 2009.

Now at 0.0385 per minute rate we will give Aviya Company (AVPC) (0.0385 cent*

4307.27 minutes) = 165.8382 dollar.

57

And finally our profit is (215.3732 – 165.8382) = 49.5350 dollar. And this profit is for

the day 25th December 2009.

58

Chapter 7

Session Border Controller & VoIP

Configuration

59

Chapter 7

Session Border Controller and VoIP Configuration

7.1 Session Border Controller and VoIP configuration

Session Border Controller is a layer 3 switch. There are many companies in the

market who provide SBC switch product. Some common products are Cisco 7600,1000

and MC 3800 series product, VoIP Switch SBC, Netgear SBC, Avira SBC, Avya SBC,

Quintum SBC, Mera SBC, NexTone SBC etc. We are using VoIP switch here in Rapid

Target Service. This switch is in Callpop, USA. We access this switch by remotely

accessing. There are many ways to use remote access service. Some remotely access

software is:

1. Windows default remotely access software.

2. Radmin Viewer

3. LogMeIn

4. Team Viewer

5. Real VNC

60

We usually use Radmin Viewer to access our switch.

Figure7.1: Radmin Viewer Software

In figure we can see 3 switches. Kelvin, RTS and Subash all these are our switches.

61

Figure 7.2: RTS SBC desktop

In figure we can see the SBC desktop. Here we can see 3 windows. Top one is

menu bar, then Call window and Statistics window.

Statistics window shows the status. Here we can see Total 24275 calls were

attempted and from them 3999 calls were successful. Right now 7 calls are pending and 4

calls are connected. We can see the value of ASR and ACD. These two are used to

measure call status. ASR and ACD are described later.

62

Call window exhibits the current active calls status. Here we can see from top 5

calls, customer NBL is sending call to Pakistan. For sample the tope call is

318809230066172139, NBL (SIP). Here 31880 is the prefix of customer NBL,

9230066172139 is the calling number, NBL refers to customer name and (SIP) refers the

protocol used by this customer.

There are 5 status of a call. They are:

1. Blue with triangle sign which refers call is pending or attempted

2. Blue with rectangle sign which refers call is failed

3. Green with triangle sing which refers call is running or active call

4. Green with rectangle which refers call is ended successfully

5. Red with cross sign which refers call is unauthorized or unable to

access this call. The issue is either the client has not enough credit to

make this call or the call comes from a customer who is expired in

contact or call comes from fake customer.

63

Figure 7.3: Call status

64

We can also access our switches by browser. We can use only internet explorer

to access our switches. This service is known as webconfig. The following figure shows

our switch configuration by internet explorer.

Figure7.4: SBC in Web Config

The figure represents configuration graphical user interface GUI in web

application. Here we can see active calls section. Three are 8 active calls at the time when

it was captured. Account name refers customer names. Duration how many minutes or

hour customers are talking. Route refers the route used by SBC to send call to its

destination. Dialed numbers refer number that are dialed and here before number prefix

are added.

65

7.2 Set up Gateway Client (GW Client)

Clients are known in SBC as gateway client in short GW client. Set up order are

given from USA through mail. They just say the client name and how much credit should

assigned. I have to gather all set up information from longjump. To make this clear I need

to say something about longjump. Longjump is an online data store or database

application. In longjump all clients and vendors information are stored. After getting

order or tasks to set up vendor or client in SBC we have to go longjump and gather

information of that client or vendor. The web address of longjump is

www.longjump.com. The following figure represents how we get tasks.

Figure7.5: Tasks to create GW client

66

After getting order to set up customer Callvox in SBC I have to go in longjump.

The following figure shows the information from longjump.

Figure7.6: Customer Callvox’s information from Longjump

The highlighted red part is the required information for set up. This part

contains parameters like customer’s IP, what protocol this customer using, codecs

supported by this customer router, its prefix and its tariff name.

At the very first we have to create tariff for this customer and its name will be

CLVX Cust. We have to use prefix before country code or break out. This prefix will be

71714.

67

Figure7.7: Customer CLVX’s tariff

In figure we can see customer Calvox’s tariff which is named as CLVX tariff

here. We can see Jamaica mobile Centennial break out code with prefix 71714 and with

rate 0.12 cent for per minute.

68

Now we will create account for customer Calvox including its credit in SBC.

Figure 7.8: Customer Calvox’x account in SBC

Here in figure we can see Calvox customer’s parameters. Calvox is using

G723.1 and G729. Its primary codec is G729. It has 500 Call limits which mean it can

send 500 calls at a time. Its IP addresses are given in IP addresses box. In top we can see

how much fund it has. Here it is 997474.9893 dollar.

69

7.3 Set up Gateway in SBC

Gateway set up option is for vendor site. This set up contains what patterns of

calls vendor want to accept i.e. prefix, their IP, which codec they support, which protocol

they want to use and their call limits. The following figure represents how we get order to

set up a new gateway.

Figure7.9: Tasks to create Gateway

70

The following figure exhibits the parameters collected from Longjump.

Figure 7.10: Vendor Easycom Exchange’s information from Longjump

First we have to create tariff for Easycom Exchange with

appropriate prefix. The following figure shows the tariff EASY Vend. One thing is highly

noticeable here that the customer tariffs are marked by cust at the end of customer short

name for example CLVX cust and for vendor, vend is used at the end of vendor’s short

name.

Figure7.11: Vendor Easycom Exchange’s tariff

71

The following figure represents gateway set up for a vendor in SBC.

Figure 7.12: Easycom Vendor set up in SBC

Here we can see IP, codecs, protocols and port limits used by vendor Eaycom .

72

7.4 Dial plan

Finally the dial plan is the last task to complete all setup to route

calls into a country. Every set ups are meaningless without dial plan. Dial plan binds the

client and vendor set up. Without dial plan client can’t send any calls. Dial plans say

which route should use for the incoming calls. Let us see the figure:

Figure7.13: Dial Plan

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The figure defines the dial plan for customer Callvox. If we go

back in GW client set up then we can see Callvox’s prefix is 71714. So if Callvox send

traffic to Jamaica then call will come to the inbound as 717141876xxxxxx format

because 1876 is the international telephone code for Jamaica. You can see the incoming

call can be route through 3 routes. They are VIRT Jam mo Digi, TLPT Jam mo Digi and

TLPT jam mo CWJ. For route VIRT 71714 prefix will be replaced by null. For TLPT

71714 prefix will be replace by 105119 because the vendor TLPT want to accept call in

that format. So for TLPT if inbound call is 717141876xxxxxx then outbound call will be

1051191876xxxxxx format.

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Chapter 8

Monitoring and Quality Control

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Chapter 8

Monitoring and Quality Control

8.1 Monitoring

To monitor the status of call of routes the following two terms are used.

8.1.1 Average Seizure Ratio (ASR)

ASR stands for the percentage of calls that actually get terminated. For more

details it is the ratio of success calls. It is defined by the following formula:

ASR= (Total successful call/ Total call attempted) * 100

For example say for route A 160 calls were attempted among the 90 calls were succeed.

So ASR of route A = (90/160) * 100 = 56.25 %

8.1.2 Average call duration ACD

ACD stands for average call duration. Some telephone or VoIP companies use

ALOC instead of ACD though both refer same. It is defined by the following formula.

ACD = Total duration of calls/ Total number of calls

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For example, in route A there are 5 successful calls. Say their duration is like:

Number of Calls Dialed Number Duration in minute

1. 9477xxxxxx 10

2. 9477xxxxxx 1

3. 9477xxxxxx 0.5

4. 9477xxxxxx 1.5

5. 9477xxxxxx 1

Total 5 calls 14 min.

Table 8.1: ACD Calculating

So the ACD will be 14/5 = 2.8 minute

Figure 8.1: Route Status

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The figure exhibits the status of routes for the date 28th December 28, 2009. Here

Count is total Success calls, Count Fail is total failed calls, ASR is Average Seizure

Ratio, AvgDur is average duration or ACD and Sum Dur is total duration. In figure the

status of the bottom route AVPC Pak Mo Mobi is ASR 24% and ACD 3.22 minute. The

status for another route named AVPC Pak mo Tele, the ASR is 4 and ACD is 2.11

minutes. The status of this route is definitely bad. We can see only 19 success calls and

412 failed calls. So there must be some issue. For this issue we send Trouble Ticket (TT)

to the vendor. Now I will show here how we send TT to the vendor.

Figure8.2: Trouble Ticket

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8.2 Testing

The purpose of testing is to verify the quality of a route. Sometimes It happened

that one route gave good quality before but its performance is going down from the last 2

hours then in that case testing are done to understand what is going on in that route. This

is the initial step of troubleshooting. After finding out the problem if the issue exists in

Session border controller then its responsibility goes to the network operation center

(NOC) of SBC service provider. If the issue is on vendor side NOC department of SBC

service provider sends TT to the vendor.

Another purpose of testing is route selling and buying. When SBC want to make a

contract with a vendor to use its route then before contract they test that route. Then

according to the test result they decide its quality and decide they should buy this route or

not.

8.3 Testing Parameters

The following parameters are used in testing a route:

Post Dial Delay (PDD): The time delay between dialing and call connection is

referring to PDD. Usually 40 seconds are standard PDD. If the PDD is high then

standard then there must be some issue.

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False Answer Supervision (FAS): The one of the top most issues in VOIP

telephony is False Answer Supervision (FAS). Two types are FAS is experienced

in VOIP and they are:

When the switch starts to bill the call before the call s actually

connected or user start to talk.

When switch bills the call after 3 second or 5 second or in some cases

more time after the call actually connected.

In case of FAS the issue must be in setup of router.

Echo: When caller gets back his or her voice back this situation is called echo

problem.

Static: If caller gets noise tone then it refers to Static problem.

One way Audio: The caller can hear receive voice but receiver can’t or in reverse

receiver can hear caller voice but caller can’t hear receiver and all these are the

issue that refers to one way audio.

IE errors: IE errors refer to H323 protocol call signaling error.

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Release Reason: Release reason refers to SIP protocol call signaling error.

Rapid Target Test Form

     

Support Member: Tareq  

Test Date: 9th September 2009  

Test Start Time: 11.45.09PM  

Vendor Name: TLPT  

     

     

     

Country: Haiti  

Breakout: Mobile   

Carrier/City Digicell  

Number dialed IE Error Release Reason

50937448475 34 -1

50937533619 34 -1

50936066348    

50938079199 34  

50936073255 34  

50936073255 34  

50938690446    

50938690446 34  

50937590610    

50937590610 34 -1

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PDD Ring? Answer? Echo? Static? One way Audio?

7 Yes No Yes No No

           

           

           

6 Yes Yes No    

           

6 Yes No Yes No No

           

           

Volume? Duration of call Other Quality Notes  

    no circuit channel available

    no circuit channel available

Good 1.15    

    no circuit channel available  

    no circuit channel available  

    no circuit channel available  

  0.15 IVR  

    no circuit channel available

Good 0.51    

    no circuit channel available

       

Table 8.2: Test Report

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References

[1] www.nascio.org/publications/documents/NASCIO-VOIP.pdf

[2] www.quickstartvoip.com/voip-advantages.html

[3] www.why-switch-to-voip.com/Advantages_Disadvantages_VoIP.html

[4] www.nascio.org/publications/documents/NASCIO-VOIP.pdf

[5] www.voipadvantage.co.uk

[6] www.ciscopress.com/articles/article.asp?p=606583&seqNum=8

[7] http://www.faqs.org/docs/Linux-HOWTO/VoIP-HOWTO.html#ss4.2

[8] www.national.com/appinfo/ adc /files/ABCs_of_ ADC s. pdf

[9] www.msforum.org/techinfo/reports/MSF-TR-QoS-001-FINAL.pdf

[10] www.voip-news.com/.../voip-qos-service-quality-012207/

[11] www. voip .about.com/od/glossary/g/ delay .htm

[12] www.trilliumwoodpartners.com/id29.html

[13] www.commsdesign.com/design_corner/showArticle.jhtml?articleID=207000171

[14] www.toncar.cz/Tutorials/VoIP/VoIP_Basics_Jitter.html

[15] www.voip.about.com/od/glossary/g/echo.htm

[16] http://www.microtronix.ca/echo_problems.htm

[17] www.polycom.com/.../vortex_choose_ acoustic _ echo _canceller.pdf

[18] www.ditechcom.com/solutions/appnotes/AN_HEC_Wireline_Bidi.pdf

[19] www.cisco.com/web/.../Dispelling_the_ Packet _ Loss _Myth.pdf

[20] www.is.co.za/.../ VoIP +its+all+about+quality+of+service.htm

[21] www.en.wikipedia.org/wiki/Voice_activity_detection

[22] www.cisco.com/en/US/docs/switches/wan/mgx/.../VSM15Ch4.pdf

[23] www.patton.com/whitepapers/ voip _and_ qos . pdf

[24] www.brocade.com/forms/getFile?p.../ voip _ solutions _ga_tb... pdf

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[25] http://ww1.microchip.com/downloads/en/devicedoc/21841a.pdf

[26] http://www.voipcodec.org/

[27] http://www.javvin.com/protocolRTP.html

[28] http://icapeople.epfl.ch/thiran/CoursED/RTP.pdf

[29] www.siptutorial.net/ RTP / header .html

[30] www.iec.org/online/tutorials/acrobat/h323.pdf

[31] www.ja.net/documents/publications/factsheets/035-h.323.pdf

[32] www.ee.ucla.edu/~vandenbe/publications/sip.pdf

[33] www.sipcenter.com/sip.nsf/html/.../Ubiquity_SIP_Overview.pdf

[34] www.cs.columbia.edu/~hgs/teaching/ais/slides/2003/sip_long.pdf

[35] www.faqs.org/ rfc s/ rfc3261 .htm

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