Voice over Internet Protocol

Final Project: Voice over Internet Protocol

Presented by: Mohamad Haidar

Advisor: Dr. Rad Alrifai

TO

Computer Science and Telecommunications Department

04/12/23 Graduation Project 2

Table of Contents

• Introduction

• Packet Voice Transport

• How VoIP works?– Advantage– Requirements– Components

• Real Time Transport Protocol


(Continued)• VoIP Routing• VoIP Applications• QoS Issues and Challenges• H.323 Standard• Brief Description of OPNET• Project

– Description– Results and Analysis– Conclusion

• Future Works


Introduction• The Past

• The Present

• The Future• VoIP is a category of hardware and software that enables

people to use the internet as the transmission medium for telephone calls.

• Today, both the telephone and the personal computer are fundamentals helpers in most homes and offices.

• Traditionally, continuous voice service was carried over circuit-switched telephony networks using analog methods.

• Digital representation of audio signals is fundamentally attractive since it offers more flexibility in manipulating and processing than does analog representation.


• Packet switching integrates voice and data in a single network– It can exploit the bursty nature of data and voice to reduce the

transmission bandwidth needed to carry a particular mix of traffic over a circuit network.

– More flexible allocation of bandwidth to individual calls and interfaces.

• Adding voice to packet networks requires an understanding of how to deal with system level challenges, such as interoperability (different protocols at the same time), packet loss, delay, density (voice traffic), scalability (the ability to scale VoIP to large volumes of traffic) and reliability


Packet Voice Transport– Frame Relay:

• flexible bandwidth, widespread accessibility, support of diverse traffic mix and technological maturity.

• Based on PVC.• Connection oriented

– ATM: • Multiservice• high speed• scalable.• Connection oriented• Advantage over FR: fixed cell size.

– IP: • Connectionless protocol.• Packets follow different paths. Paths shared with

different packets.


How VoIP Works?

• VoIP digitalizes voice signals in data packets, sending them and reconverting them in voice at destination.

• Digital format can be better controlled: it can be compressed , routed, converted to a better format. Also a digital signal is more noise tolerant than the analog one

Voice(source)

A/D converter

INTERNETINTERNET

D/A converter

Voice(destination)


Advantage of Using VoIP

• PSTN: – pay for the usage time (Distance sensitive)– could not talk with more than one person at a time. – No data transfer– More expenses (phone)

• VoIP: – Talk all the time with any number of persons who are also

connected to the internet at the same time (Distance insensitive)

– Data may also be exchanged while conversation is in progress.


VoIP Requirements• High reliability• High availability of connection• High voice quality• Interoperability between vendors• Highly user friendly components / software • Network management • Efficient network signaling• Real time billing collection and account

information.


VoIP Components• IP Telephony Software: minimum system requirements

– 486DX computer with 8MB RAM – 16-bit sound card – 14.4 kbps modem (minimum)– SVGA display– Speakers – Microphone – Internet connection – Net Phone software

• Other options:• IP Telephones:IP telephones have the functionality suitable

to VoIP. At the same time, these phones should have a normal telephone’s functions for user’s comfort


Compression Algorithms• Pulse Code Modulation (PCM): Digitizes audio signals, 8000

samples/sec and 8 bits/sample 64Kbps.

• Adaptive Pulse Code Modulation (ADPCM): Lower bit rate than PCM by recording only the difference between samples and adjusting the coding scale dynamically to accommodate large and small differences. Requiring 32Kbps.

• Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP): speech frame 10ms and 80 samples/frame at 8Kbps.

• Low Delay Code Excited Linear Prediction (LD-CELP): 2.5 ms frame, samples at 16Kbps.

• ACELP: samples at 5.3 Kbps.

• Linear Predictive Coding (LPC) is one of the most powerful speech analysis techniques, and one of the most useful methods for encoding good quality speech at a low bit rate. Samples 2.5 Kbps!


Mean Opinion Square

• Mean Opinion Square (MOS):Each CODEC provides a certain quality of speech. The quality of transmitted speech is a subjective response of the listener. A common benchmark used to determine the quality of sound produced by specific CODECs is the Mean Opinion Square (MOS)

Bit Rate (Kbps)

MOS score

Voice Quality

G.711 PCM 64 4.1 A

G.729 CS-ACELP

8 3.92 A

G.726 ADPCM 32 3.85 B

G.728 LD-CELP

16 3.61 C

G.723.1 5.3 3.65 C

G.729a CS-ACELP

8 3.7 B


Routing Protocols

• Real Time Protocol (RTP): It encapsulates data into a TCP/IP stack according to the following structure.

• RTP services: Timestamp, sequence numbering, payload ID and source ID.

IP Header UDP Header RTP Header RTP Payload


Need for RTP

RTP provides functions to support:

– Loss, out of order: sequence number

– Loss, jitter: timestamp

– Source/payload identification

• RTP combines its data transport with a control protocol (RTCP), which makes it possible to monitor data delivery for large multicast networks. Monitoring allows the receiver to detect if there is any packet loss and to compensate for any delay jitter. Both protocols work independently of the underlying Transport layer and Network layer protocols.


VoIP Routing

• RSVP

– can be automatically invoked by VoIP gateways to ensure that voice traffic is able to use the best path through the network

– It reserves bandwidth for transmission

– RTP ensures delivery of packets.


Transport Protocol

• TCP is costly for real time applications.

• UDP, each packet takes different path and packets do not arrive in order. RTP will manage this.


VoIP Applications1. 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.

2. Trunking application: 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.


VoIP QoS Issues & Challenges

• QoS: is the ability of networks to guarantee and maintain certain performance levels for each application according to the specified needs of each user.

• In VoIP, QoS mainly is thought of:– Reservation of Bandwidth

– Predefined delay

– minimize Jitter or Congestion

• Challenges are:– Delay: Unacceptable in voice. Causes talker overlap


Types of Delays• Accumulation Delay: This delay is caused by the need to collect a frame of

voice samples to be processed by the voice coder. It is related to the type of voice coder used and varies from a single sample time (0.125 microseconds) to many 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.

• 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.

• Jitter: It quantifies the effects of network delays on packet arrivals at the receiver. Packets transmitted at equal intervals from the left gateway arrive at the right gateway at irregular intervals.


(Continued)• Techniques used to avoid jitter

– 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. So, we 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.

• Approaches to adapt buffer size1. 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.

2. 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.


• Other Challenges:– Congestion: the problem of congestion can be solved by the router

prioritizing the data traffic below the voice traffic priority.

– Voice Activity Detection: voice activity detection improves bandwidth savings more than compression. It suppresses silence.

• Packet Loss: packet loss occurs either in bursts or periodically due to a consistently congested network.

– Compensation:

• Interpolate for lost speech packets by replaying the last packet received during the interval when the lost packet was supposed to be played out (Disadv: 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 (more BW and 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


• Echo Compensation: The echo canceller compares the voice data received from the packet network with voice data being transmitted to the packet network. The echo from the telephone network hybrid is removed by a digital filter on the transmit path into the packet network.

• Sequence Errors: Congestion in packet switched networks can cause packets to take different routes to reach the same destination. Consequently, packets may arrive out of order resulting in garbled speech.


H.323 Standard• H.323 is an International Telecommunication Union

Telecommunication Standardization Sector (ITU-T) specification for transmitting audio, video and data across an Internet Protocol (IP) network, including the Internet

• The H.323 standard consists of the following components and protocols

• Benefits: provide base

functionality for voice,

video and data. The H.323

standard supports a wide

variety of encodings for

voice and data

communication

Feature Protocol

Call Signaling H.225

Media Control H.245

Audio Codes G.711, G.722, G.723, G.728, G.729

Video Codes H.261, H. 263

Data Sharing T.120

Media Transport RTP/RTCP


OPNET• It allows the definition of a network topology, the nodes, and

the links that go towards making up a network.

• OPNET is capable of simulating large communication networks with detailed protocol modeling and performance analysis.

• Models built with OPNET are hierarchally structured:– Process domain, lowest level, is structured as Finite State Machine

(FSM). Then FSM, with the help of geographical editor, allows the user to specify relation between single states and their transitions.

– Processes, as well as data generators and queues, can then be grouped into nodes in the node domain.

– Nodes can now be connected with each other to build up different network architectures in the network domain.


Project

• Starting the Project

• Description of the Project and Experiments– In order to understand the behavior of voice on the network, it

is necessary to understand the network itself and test for its ability to respond to voice without any problems or excessive delays.

– Summary of scenarios with the statistics collected on the following page


Scenario # Abbreviation Description Statistics Collected

1NO_VoIP_10 No VoIP implemented on first LAN,

while using 10Mbps as link

*Email Download Response Time (sec)*HTTP Object Response Time(sec)*Ethernet Delay (sec)*Server Load (bits/sec)

2 WITH_VoIP_711_10

VoIP is implemented on first LAN using G.711 Voice Encoder Scheme and 10Mbps link.

Email Download Response Time (sec)*HTTP Object Response Time(sec)*Ethernet Delay (sec)*Server Load (bits/sec)*Packet delay variation (sec)*End-to-End packet delay (sec)

3 WITH_VoIP_729_10 VoIP is implemented on first LAN using G.729 Voice Encoder Scheme and 10Mbps link.

Same as 2

4 NO_VoIP_2LANs_10 No VoIP is implemented on the two LANs using 10Mbps

Same as 1

5 WITH_VoIP_711_2LANs_10 VoIP is implemented using G.711 as Voice Encoder Scheme on the two LANs with 10Mbps link.

Same as 2

6 WITH_VoIP_729_2LANs_10 VoIP is implemented using G.729 as Voice Encoder Scheme on the two LANs with 10Mbps link

Same as 2

7 NO_VoIP_2LANs_100 No VoIP is implemented on the two LANs using 100Mbps

Same as 1


Same as 2


Same as 2


• The Network for the scenarios from 1-3 looks like:

• The network for scenarios from 4-9 looks like:


Ethernet Delay

Email Download Response time (sec)

Server Load

Results and Analysis


With_VoIP

Comparing G.711 (HTTP)Comparing G.711 and G.729 Email


Comparing G.711 Email Comparing G.711Ethernet Delay


Comparing G.729 EmailComparing G.729 HTTP


Comparing G.711 and G.729 on all LANs, packet end-to-end delay

Comparing G.729 Ethernet Delay


Comparing Packet Delay variation on All

Comparing Server Load on all


Conclusion

• In conclusion, deciding on what scheme to use depends heavily on the network designer. In a small network like the one we tested, it is not a bad idea to implement G.711 mainly because there was no huge difference between the two schemes. Even the server load and the delay on the network were not severe when using G.711, for we are using small networks.

• While, when bigger networks are implemented, G.711 might cause some loss or unrecognizing of some voice packets that might not be delivered to the destination due to the size of the voice packet. On the other hand, G.729 would be better for larger networks since it compresses the voice packet in addition to digitizing it


Future Works• The goal of VoIP developers is to add telephone calling

capabilities to IP-based networks and interconnect these to traditional public telephone networks and to private voice networks maintaining current voice quality standards and preserve the features everyone expects from the telephone.

• Future works may include designing a WAN and trying to apply the same or different parameters on the network to study the effect of VoIP on WANs. Moreover, with some more time spent on effectively studying OPNET, a future design may include creating the same machines that are unavailable in OPNET’s library and are available at Roosevelt University’s computer labs and trying to simulate one of them running VoIP, then a larger scenario which will be simulating a VoIP over all the intranet of Roosevelt University.


Conclusion• It is necessary that companies invest in IP telephony today if they

want to play a major role in the future.

• I learned about the components and the procedures of VoIP in order to install it.

• I achieved great understanding of protocols governing VoIP, such as UDP, RTP, RTCP, RIP, and H.323 and the different voice compressing techniques like G.711 and G.729.

• I stated the importance of integrating voice and data on the same network and what are the changes to take place in that case.

• I learned OPNET which is a powerful networking simulation tool.

• I came up with some conclusions that allow us to run VoIP on any fast Ethernet network with the knowledge of the assumptions taken before running the simulations.

• cost is an important issue today, the solution should also fit all economic needs in terms of budget and high performance.

Voice over Internet Protocol

Documents

voice signals

voip works

rtp rtp

future voip

voip pstn

density voice traffic

mos score voice quality

data transport