VOICE TRANSMISSION ON IP BASED NETWORK

USING WIMAX

A PROJECT REPORT

Submitted by

R. ABINAYA(42007106002)

S. INDHUMATHI(42007106035)

P. LOGESWARI(42007106046)

In partial fulfillment for the award of the degree

of

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

TAGORE ENGINEERING COLLEGE

CHENNAI

JANUARY-2011
ABSTRACT:

The new era of communication, currently employed in some parts of the world, is

Worldwide Interoperability for Microwave Access (WIMAX). The Wimax technology is

becoming the way to avert the impending crisis of rural connectivity . VoIP applications are

being widely used in today's networks challenging their capabilities to provide a good quality

of experience level to the users. The work presented here is a unique contribution assessing the

VoIP sessions quality on a real WiMAX test-bed, using H.323 and SIP (session initiation

protocol) protocols. VoIP quality is measured according the quality perceived by the end users

as well as through conventional network parameters, such as one-way delay and packet loss.

The results put in evidence a good quality for VoIP services with 60 simultaneous users in a

WiMAX link with resources pre-provisioned. A WiMAX infrastructure will let service providers

deliver VoIP to rural areas. This project explains about the purpose of implementation of VoIP

over Wimax, the study of VoIP and Wimax systems, its implications and applications and its

wireless capabilities.
VOIP OVER WIMAX

A WiMAX infrastructure will let service providers deliver VoIP to rural areas without a
backhoe.  AT&T and Covad have revealed their intention to use WiMAX technology to provide VoIP
and data services to rural areas to avoid local access charges. The announcement earlier this month
clears up the mystery surrounding the service providers' membership in the WiMAX Forum
Apparently, instead of trenching fiber everywhere, as Verizon is doing, they will deliver VoIP
services over a less expensive wireless infrastructure. 

VoIP over WiMAX is attractive for both enterprises and carriers, but Challenges
remain. First, no WiMAX - certified equipment has yet been released—though some shipments
are expected in 2005, just in time for AT&T and Covad's plans. Second, the initial WiMAX
equipment will be certified in the 2.5- and 3.5-GHz spectrum ranges. The 2.5-GHz spectrum is owned
largely by Sprint and Nextel, while the 3.5-GHz spectrum is not available for use in the United
States . This means that any near-term WiMAX equipment deployment will likely use unlicensed
spectrum, which carries the risk of interference from other wireless devices.

 Organizations seeking a quick way to provide connectivity to branch offices should consider
using high-speed wireless, even if it's pre-WiMAX. AT&T and Covad's announcement should help
confirm that wireless VoIP can be made reliable and cost-effective. 

REASON FOR CHOOSING VOIP

• Voice over Internet Protocol (VoIP) is a technology that enables one to make and receive
phone calls through the Internet instead of using the traditional analog PSTN (Public Switched
Telephone Network) lines.

• VoIP is packetisation and transport of classic public switched telephone system audio over an
IP network.

• It allows 2-way voice transmission over broadband connection.

• It is also called IP telephony, internet telephony, voice over broadband, broadband telephony.
EXISTING METHOD:

Fig.1. Voice transmission from PC to PC

Existing phone systems are driven by a very reliable but some what inefficient method for
connecting calls called circuit switching. Circuit switching is a very basic concept that has been used
by telephone networks for more than 100 years. When a call is made between two parties, the
connection is maintained for the duration of the call. Because you're connecting two points in both
directions, the connection is called a circuit. This is the foundation of the Public Switched Telephone
Network (PSTN). Then came voip over broad band which was packet Switching. Broadband
wireless has long held the promise of delivering a wide range of data and information services
to business and residential customers quickly and cost-effectively. Unfortunately, that promise has
been imperfectly met in the past because of both the immaturity of the existingtechnologies and the
relatively high cost of networking equipment
COMPARISON OF DIFFERENT STANDARDS:

FEATURES 802.11 802.11b 802.11a 802.11g 802.16

Assigned Spectrum 2.4 6GHz 2.4GHz 5.9GHz 2.4GHz 10GHz-66GHz

Maximum 2Mbps 11Mbps 54Mbps 54Mbps 70Mbps

Throughput

Transport Protocol Ethernet Ethernet Ethernet Ethernet TCP/IP,ATM

Supported

Modulation System Frequency Frequency OFDM OFDM QUAM,PSK,

Hopping Hopping Direct OFDM
Direct Sequence
Sequence
Adaptive No No No No Yes
Modulation

Network Point to PTMP PTMP PTMP PTMP,PTCM,Me

Architecture Multipoint sh
Supported PTMP

QOS No No No No Yes

Fig.2.Tabulation
PROPOSED MEHOD

R
O
U
T VOIP GATEWAY
E BOX
R
WIMAX
WIMAX CPE
BB
WIMAX

INTERNET

Fig.3. Voip over wi-max

The 802.16 standard can accommodate both continuous and bursty traffic, but it uses what is
essentially a connection-oriented protocol somewhat akin to those of ATM and frame relay.
Modulation and coding schemes may be adjusted individually for each subscriber and may be
dynamically adjusted during the course of a transmission to cope with the changing radio frequency
(RF) environment. In the higher frequencies, 16 quadrature amplitude modulation (QUAM) and 64
QUAM are automatically invoked by the protocol to match signal characteristics with network
conditions.The orthogonal frequency division multiplexing (OFDM) modulation scheme is specified
for the lower band with a single carrier option being provided as well The 802.16 protocols are highly
adaptive, and they enable subscriber terminals to signal their needs while at the same time allowing
the base station to adjust operating parameters and power levels to meet subscriber needs. Polling on
 the part of the subscriber station is generally utilized to initiate a session, avoiding the simple
contention-based network access schemes utilized for WLANs, but the network operator also has the
option of assigning permanent virtual circuits to subscribers—essentially reservations of bandwidth.
Provisions for privacy, security, and authentication of subscribers also exist. Advanced network
management capabilities extending to layer 2 and above are not included in the standard. The purpose
here is to present the WiMAX network solutions, to provide best practice guidelines that aggregate
existing standards and recommendations to WiMAX network operators on deploying VoIP.

VOIP NETWORK DEPLOYMENT SCENARIO

Several categories of VoIP Networks are to be considered based on the terminal and the
location of the VoIP platform

1. Fixed User End-User VoIP Termination– VoIP platform located on the same premises as the
WiMAX Provider’s network. as follows:
2. Mobile User – VoIP platform located on the same premises as the WiMAX Provider’s
network
3. Fixed User – VoIP platform located at a separate location from the WiMAX Provider’s
network.
4. Mobile User – VoIP platform located at a separate location from the WiMAX Provider’s
network
5. Roaming – When a WiMAX user is roaming the VoIP platform used may be located either in
the visited network or the home network.

VOIP ECOSYSTEM
An ecosystem is a unit of interdependent systems interacting as a functional whole. The
purpose of the interdependent systems is to make and receive calls. The three logical networks that
make up the VoIP Ecosystem include the following:
• PSTN Carrier Network: Made up of traditional Cell and POTS line providers interconnecting
using a circuit switched network.
 • VoIP Application Service Provider Network: The network where the VoIP equipment is located.
• WiMAX Provider Network: For the objective of this paper, this network will
include the WiMAX base stations, ASN and CSN.
GATEWAY
A Gateway is a device used to translate media streams between different technologies. With
VoIP, the gateway converts the media stream as well as the control protocol between packet-
switched and circuit-switched methods.

BASIC TECHNIQUES IN VOIP

1. Audio codec.
2. Data transport (RTP, RTCP)
RTP (Real-Time Transport Protocol)
RTCP (Real-Time Control Protocol)
3. Addressing.
4. Signaling (SIP, H.323).

AUDIO CODEC
The choice of CODEC used is important because it determines the required bandwidth per
call. G.711 and G.729 are the most widely used codecs in existing networks, however, G.729
CODEC is widely used in wireless networks . G729 has a data rate of 8 kbps and a 10-30 ms sample
period. Average bandwidth usage is ~40 kbps per call.
The G.711 codec is currently used in a wide range of applications. Its voice sampling rate is 8
kHz and each sample is encoded with 8 bits resulting in a constant 64 kbps bit rate and offers very
good voice quality. Samples can be packed into frames every 10 ms or another longer sampling rate.
The G.729 codec can also generate speech frames every 10 ms or longer sample rate .Each 10 ms
frame contains 80 voice samples (collected at a sampling rate of 8000 samples per second), or another
longer sampling rate. However, it requires a 5 ms look-ahead delay before producing any new frame.
It is developed for multimedia simultaneous voice and data applications.
 SPEECH CODEC NO. OF BYTES BIT-RATE TOTAL NO EFFECTIVE
PER FRAME OF BYTES EFFICIENCY
G.711 80 64kb/s 120 67%
G.723.1 24 6.4kb/s 64 37%
G.729 10 8kb/s 50 20%
G.729 10 8kb/s 60 33%
G,.729 10 8kb/s 70 43%
GSM-EFR 31 12.4kb/s 71 44%

Fig.4. Effective efficiency of different types of codecs

DATA TRANSPORT PROTOCOL

RTP
RTP stands for Real-Time Transport Protocol. Application layer protocol for transmitting real
time audio, video etc.The data transfer protocol, deals with the transfer of real-time multimedia data.
Information provided by this protocol include timestamps (for synchronization), sequence numbers
(for packet loss detection) and the payload format which indicates the encoded format of the data.

RTCP
RTCP stands for Real-Time Control Protocol. The RTCP is used to specify Quality of Service
(QOS) feedback and synchronization between the media streams.Responsible for periodical
transmission of control packets to all participants in the session.Supports for multi-point
communication.

ADDRESSING
Here phone no. is converted into IP address. Simple format addressing:
<user | phone no.>@<domain | hostname | IP address>
If you make a call using a phone with an adaptor, you’ll be able to dial just as you always have. If
there is no adapter we have to use the IP address of the particular phone.
 SIGNALLING
Signaling is one of the most important functions in the telecommunications infrastructure
because it enables various network components to communicate with each other to set up and
tear down calls. Significant efforts were undertaken in past decades to develop the signaling
protocols in use in today’s telephone network, also known as the public switched telephone
network (PSTN). These protocols, such as Signaling System H.323 and SIP, are defined in
large detailed specifications developed by various standardization organizations
Signaling in VoIP is needed for:
1. Locating partners.
2. Agreeing on port numbers for RTP/RTCP sessions.
3. Agreeing on coding/decoding procedures.
Types of signaling protocols:
1 .H.323.
2.SIP (Session Initiation Protocol).

OVERVIEW OF SIP
Intense work on SIP really began in earnest with the Internet Engineering Task Force
(IETF) in1999. It was one of many efforts and has been led by the IETF-SIP working group.
Their charter States that SIP is a text-based protocol, similar to HTTP and SMTP, for
initiating interactive Communication sessions between users. Such sessions include voice,
video, chat, interactive games, and virtual reality. This group has worked long and hard to help
SIP mature. What began as a series of proposed drafts and standards , including numerous
extensions, has become foundation for VoIP and unified communications.
The basic model and architecture defined for SIP sets out some specific characteristics:
• Wherever possible, SIP services and features are provided end-to-end
• Extensions and new features must be generally applicable; they cannot apply only to
some specific set of session types
• Simplicity is key
• Existing IP protocols and architectures are re-used and integrated tightly

SIP uses an addressing structure similar to email addresses. Users may log in any where
and be dynamically assigned an IP address, so there has to be a way to resolve some of the
commonconventions in the active and current IP address.

SIP is text based, so the addresses, which are SIP URLs or URIs (Uniform Resource
Locaters or Indicators), can be imbedded in email messages or Web pages. Additionally, as SIP
is a text protocol, SIP URLs and URIs are network-neutral. Thus, a URL might point to an
email-like address, using SIP, an H.323 address or even a telephone number on the PSTN.

SIP operates independently of the IP network layer. It requires only unreliable packet
delivery and provides its own reliability mechanism. Although it’s widely used in IP networks
today(usually over UDP to avoid the overhead of TCP), SIP can run over IPX, Frame Relay,
ATM, AAL5, or X.25 with no changes.

SIGNAL CONTROL FOR SIP:

1. UA wants to make a collection with Terminal. It sends an INVITE request to IWF.
2. IWF translates INVITE to SETUP and tells Terminal.
3. Terminal gets the request and arranges with IWF.
4. Instead of UA, the IWF arranges with Terminal to get a translation address, a media
format for the following dialogs.
5. Terminal and UA communicate with media channel.
6. UA informs IWF to finish the dialog.
7. IWF finishes the dialog, and informs Terminal to disconnect the link with H.425
8. Terminal stops the dialog.
 Fig.5. Signal Control – Start from SIP UA

OVERVIEW OF H.323.
In this section we describe, at a high level, the H.323 architecture by defining the main
components of the architecture: the terminal, the gatekeeper, the gateway, and the multipoint
control unit. We then define the various protocols that are part of the H.323 family and are used
by the components of the architecture for communicating with each other. We also define how
services can be implemented within the H.323 architecture.
 The H.323 standard was initially targeted to multimedia conferencing over LANs that do
not provide guaranteed QoS. A typical H.323 network is composed of a number of zones
interconnected via a WAN. Each zone consists of a single H.323 gatekeeper (GK), a number of
H.323 terminal endpoints (TEs), a number of H.323 gateways (GWs), and a number of
multipoint control units (MCUs), interconnected via a LAN. A zone can span a number of LANs
in different locations, or just a single LAN. The only requirement is that each zone contain
exactly one GK, which acts as the administrator

Fig.6.Signal Control – Start from H.323 Terminal

When a Terminal invites a SIP UA, firstly, it will send SETUP to IWF. IWF translates it
to be INVITE without media capability description. That is because the Terminal’s media
capability is not recognized by IWF. SIP media adopt Offer/Answer model. It means UA’s
media capability can only be advanced in INVITE or the Response of INVITE. So, UA tells IWF
its media capability in the Response of INVITE. IWF records it and arranges with Terminal.
The details are as follows:
1. Terminal sends a SETUP to IWF and makes a connection between Terminal and SIP UA.
2. IWF translates SETUP into INVITE and then informs SIP UA with this INVITE.
3. SIP UA accepts the request and sends response to IWF. The response which include UA’s
media capability is an OFFER.
4. IWF arranges with Terminal about transport layer address and media format.
5. Establishing the dialog. Terminal and UA can communicate each other with media channels.
6. Terminal informs IWF to release control connection with H.245 and finish the dialog.
7. IWF runs the dialog stop program which is defined in H.323 and informs UA stop with
SIP BYE.

EFFECTIVE BANDWIDTH UTILIZATION

Voice over Internet Protocol (VoIP) has been dominating the telecommunications world.
Unfortunately, its applications are injecting a huge number of small packets in the network,
which produces high overhead and therefore wastes network bandwidth. This paper proposed the
use of a novel multiplexing technique, Delta-Multiplexing, to save the wasted bandwidth. In the
Delta-Multiplexing technique, the VoIP packets destined to the same destination gateway are
aggregated in a single UDP/IP header, therefore reducing the header overhead and saving
network bandwidth. Moreover, the Delta-Multiplexing technique reduces the size of the packets
payload by transmitting the difference between the consecutive packets payloads. Accordingly,
the Delta- Multiplexing technique greatly saves bandwidth. We have simulated the Delta-
Multiplexing technique using a 14-byte LPC codec. The result showed that Delta-Multiplexing is
capable of saving between 68% and 72% as compared to conventional techniques (without
multiplexing). Moreover, the Delta-Multiplexing technique reduces the number of VoIP packets
running over the network, therefore reducing network traffic, overload, and congestion, thus
improving the overall network performance.
REFERENCE

1. M. Decina and V. Trecordi, “Voice over Internet Protocol and Human Assisted E-Commerce”, IEEE
Comm. Magazine, Sept 1999.
2. ITU-T Rec. H.323, “Packet based Multimedia Communications Systems”, v. 2, 1998.
3. J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston,J. Peterson, R. Sparks et al., SIP: Session
Initiation Protocol, RFC 3261, Internet Engineering Task Force, June 2002
4. www.voip.com
5. J Gerard , Waldron, and W. Rachel, "Voice-over-IP: The Future of Communications", April 29,2002
6.D. Geneiatakis, G. Kambourakis, A. Dagiouklas, C.Lambrinoudakis, S. Gritzalis, “Session Initiation
Protocol Security Mechanisms: A state-of-the-art review”, INC'05International Network Conference,
S. Furnell, S. K. Katsikas (Eds.), pp. 147-156, Samos, Greece, Ziti Pubs, July 2005
7.Ramakhrishnan, R.S., Kumar, P.V., “Performance Analysis of Different Codecs in
VoIP Using H.323”, http://tifac.velammal.org/CoMPC/articles/30.pdf