This document appears to be a graduation project submitted by a student named Ahmed AbdulSattar Younis to the Lebanese French University Faculty of Business Management. The project focuses on simulating Voice over Internet Protocol (VoIP) performance using the Asterisk server software and SIPp call generation tool. Key sections of the document outline the design, implementation, and analysis of the VoIP simulation.

1. Lebanese French University
Faculty of Business Management
Bachelor in Computer Business / Networking Dept.
This graduation project is submitted for the degree of Bachelor in
Computer Business
Voice over Internet Protocols
(VoIP)
By
Student name: Ahmed AbdulSattar Younis
Student Number : 37
Supervisor name: Mr. Rawad Abikhalil
Internship Company: Gulfkeystone Petroleum
Internship Tutor: Mr. Ali Husain Ibraheem
Academic year (2010-2011)

2. BACHELOR in Business Computer Studies
BA MIAGE Graduation Project (GP)
INTERNSHIP FEASIBILITY APPROVAL REQUEST
Student full Name & Number: Ahmed Abdulsattar Younis
Student signature:
Student Information: Business computer Networking Dept.
Full name: Ahmed Abdulsattar Younis
Student number: 37
Student Address: Iraq – Erbil Minara
Internship/work company information :
Company name: Gulf Keystone Petroleum International Ltd
Company Address: 3rd Floor, UB Centre, Bakhtyari, Erbil, Kurdistan Region of Iraq
Company Activities: Oil development and Exploration.
Company Representative (internship agreement ) :
Full name: Hussein Dizzaye
Tel: +964 (0) 750 445 4542
Email: hdizzaye@gulfkeystone.com
Tutor:Company
Full name: Ali Husain Ibraheem
Tel: +964 (0) 750 330 5390
Email: aibraheem@gulfkeystone.com

3. Acknowledgement
We express my sincere thanks and appreciation to my supervisor
(Mr. Rawad Abikhalil ) for his advice, support and guidance.
We would also like to thank all my family and friends for all their
support, time and encouragement, which helped us a lot all the way long.

4. Dedication
To My Family,
To My Country,
To My Supervisor,
To My Friends,
To My Department,
To My University,
To whoever interested in Persuasion,
I dedicate this Project.

5. 4.1.1 Asterisk . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
4.1.2 SIPp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20
4.1.3 CPU RAM Analysis . . . . . . . . . . . . . . . . . . . . . 21
4.1.4 Shell Script . . . . . . . . . . . . . . . . . . . . . . . . . . 21
4.2 How to Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
Contents
1 Introduction 1
1.1 General View . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
1.2 Problem and Motivation . . . . . . . . . . . . . . . . . . . . . . . 2
1.3 Simulation Solution . . . . . . . . . . . . . . . . . . . . . . . . . . 2
2 State of the Art 3
2.1 VoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.1.1 VoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1.2 IP Telephone . . . . . . . . . . . . . . . . . . . . . . . . . 4
2.1.3 VoIP Server . . . . . . . . . . . . . . . . . . . . . . . . . . 6
2.1.4 VoIP Gateway . . . . . . . . . . . . . . . . . . . . . . . . 7
2.1.5 VoIP Protocols . . . . . . . . . . . . . . . . . . . . . . . . 8
2.1.6 PSTN . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
2.1.7 GSM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
2.2 Simulation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
2.2.1 Simulation Types . . . . . . . . . . . . . . . . . . . . . . . 13
2.2.2 Metrics and Parameters to be reported . . . . . . . . . . . 14
2.3 Related Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
2.3.1 SIPstone - Benchmarking SIP Server Performance . . . . 15
3 Specification 19
3.1 Server(Asterisk) . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
3.2 Simulate(SIPp) . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
4 Design 20
4.1 Tools . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 20

6. 5 Implementation 23
5.1 Asterisk . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
5.1.1 Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 23
5.1.2 Asterisk Run Command . . . . . . . . . . . . . . . . . . . 24
5.2 SIPp . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24
5.2.1 User Agent Client(UAC) Scenario . . . . . . . . . . . . . 24
5.2.2 SIPp Run Command . . . . . . . . . . . . . . . . . . . . . 27
5.2.3 Screens . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
5.3 SAR . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
5.3.1 SAR Run Command . . . . . . . . . . . . . . . . . . . . . 30
5.3.2 Screens . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
5.3.3 CPU . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
5.3.4 RAM . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
5.4 SHELL SCRIPT . . . . . . . . . . . . . . . . . . . . . . . . . . . 30
5.5 Performance Analysis . . . . . . . . . . . . . . . . . . . . . . . . 31
6 Conclusion 35
6.1 Summary . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
6.2 Feature Work . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35
References 36

7. List of Figures
1.1 VoIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 1
2.1 VoIP Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . . 3
2.2 ATA . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.3 Linksys VoIP Router . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.4 Soft Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 5
2.5 VoIP Phone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6
2.6 VoIP Gateway . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
2.7 200 OK[3] . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11
2.8 Registration test . . . . . . . . . . . . . . . . . . . . . . . . . . . 16
2.9 Redirect test . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17
2.10 Proxy 200 test . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
4.1 VoIP Design . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22
5.1 Key1 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
5.2 Key2 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
5.3 Key3 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
5.4 User . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
5.5 System . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32
5.6 Idle . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
5.7 Memused . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33
5.8 Commit . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

8. Chapter 1
Introduction
1.1 General View
Figure 1.1VoIP
Voice over Internet Protocol (VoIP) is a way to connect the voice conversations
over the network (Internet) or through any network using Internet Protocol and
therefore can be any number of people together online to a single network using
1

9. Internet Protocol to converse by telephone using the technique. The technology
converts analog audio signals from the phone an any audio once into digital
signals. This signal is divided into packets using Internet protocol and sent in
a number of digital tracks through the same data network. The arrival of these
packets to the specified destination (future) and the re-compilation of packets
sent to be heard clearly[1].
1.2 Problem and Motivation
Problem of project voice when we want to test the performance of the VoIP
Quality of Service (QoS). Our Motivation is to provide solution this problem
by Simulation. Also, no previous project in LFU-BMU on treats VoIP.
1.3 Simulation Solution
The idea of Simulation is to simulate a real time application will a real VoIP
server and a set of connection (more). We use simulation because it is difficult
to bring 50 or more persons for test for performance of VoIP Quality of Service
(QoS).
2

10. Chapter 2
State of the Art
2.1 VoIP
Figure 2.1VoIP Operation
3

11. 2.1.1 VoIP
History
The Institute of Electrical and Electronic Engineers (IEEE) published a pa-
per titled ”A Protocol for Packet Network Interconnection” in 1974 , VocalTec
released the first ever Internet VoIP program in 1995, ITU-T begins develop-
ment of standards for the transmission and signalling of voice communications
over Internet Protocol networks with the H.323 standard in 1996[1].
Applications
• VoIP in the Home:
It became possible to use the technology of VoIP at home, a number of
companies offer VoIP calling services that can be used in the home, more
or less replacing conventional PSTN service[3]. Some of these services
work only with proprietary telephone-calling software and don’t allow to
use a hard phone. Certain providers can support the use of a special
hard phone connected to PC’s USB port and uses the PC as a gateway
mechanism for accessing the network. Others provide an ATA device so
that can use one or more analog phones to receive calls using the service.
Still others offer the ability to use IP phones. VoIP enter in the home
quickly because of the license cost and the possibility of expansion and
proliferation of services such as: conference calls and the possibility to
install and preparation and control system[3].
• VoIP in Business:
Many vendors are producing cost-effective VoIP server devices for small
Ethernet LANs. These devices connect together endpoints in a small
office like a PBX, either through the use of conventional analog and dig-
ital phones (more on this later) or new-generation IP phones. In either
case, connecting calls to the PSTN and between local phones is usually
the responsibility of these server deviceslike a PBX or KTSand there are
several ways to make that PSTN connection. Using VoIP in small busi-
ness environments is easier when there’s some network savvy around the
office. Some traditional phone vendors are now implementing VoIP sys-
tems, so the availability of third-party networking expertise is accelerating
the adoption of VoIP in small businesses[3].
2.1.2 IP Telephone
Analog Telephone Adaptors(ATA)
The analog phone adapter ATA links any normal analog phone to the network
with Voice over Internet Protocol. This adapter has a port on the phone and
4

12. Figure 2.2ATA
Ethernet network port, ATA is a VoIP Gateways. VoIP at home or in business
does not need Internet access, a simple ATA would suffice[1].
VoIP Router
Figure 2.3Linksys VoIP Router
VoIP Router is connect two or more VoIP clients. A router also does frag-
mentation and reassembly with packets. Another function of a VOIP Router,
just like the name suggests, is to route the various packets into their destina-
tions. that a VOIP router communicates with the other VOIP routers on the
internet. VoIP at home or in business without any analog device needs internet
access. In this case a router is needed[1].
Soft Phone
Figure 2.4Soft Phone
A softphone is a software that emulate regular VoIP phone. This program
does not require more than provide a sound card and speakers and a micro-
phone. It provide a lot of service such as calling, answering and forwarding
cells. Examples: Net2Phone, PC2Call[1].
5

13. Figure 2.5VoIP Phone
VoIP Phone
Equipment is specially designed for this purpose is linked with a VoIP network
directly and does not need any additional equipment, such as ATA and did not
need to download any programs. These phones operate in accordance with
Internet protocol and one or more. Example: SIP Phone.
2.1.3 VoIP Server
VoIP Server turns an ordinary computer into a voice communications server.
VoIP server is no more than an existence computer with some programs also do
not need additional resources.
PBX
A private branch exchange (PBX) is a telephone exchange that serves a par-
ticular business or office. PBXs make connections among the internal telephones
of business company and also connect them to the public switched telephone
network (PSTN) via trunk lines.
Asterisk
Asterisk is the world’s most popular open source telephony project. Under
development since 1999, Asterisk is free, open source software that turns an
ordinary computer into a feature-rich voice communications server.Asterisk
makes it simple to create and deploy a wide range of telephony applications and
services. Code for Asterisk, originally written by Mark Spencer of Digium, Inc.,
has been contributed from open source software engineers around the world.
Currently boasting over two million users, Asterisk supports a wide range of
telephony protocols. It includes rich support for the handling and transmission
of voice over traditional telephony interfaces including analog lines, ISDN-BRI
lines and digital T1/E1 trunks. Asterisk also features support for a wide range
of VoIP protocols including SIP, IAX and H.323 among others. It supports
U.S. and European standard signaling types used in business phone systems.It
runs on Linux, Mac OS X, OpenBSD, FreeBSD and Sun Solaris. Asterisk can
be configured as the core of an IP or hybrid PBX (Private Branch Exchange),
switching calls, managing routes, enabling features, and connecting callers with
the outside world over IP[6].
6

14. Asterisk is like an erector set or a box of Legos for people who want to create
communications applications. That’s why we refer to it as a ”tool-kit” or ”de-
velopment platform”. Asterisk includes all the building blocks needed to create
a PBX system or virtually any other kind of communications solution. Some of
the blocks in the kit include:
• Drivers for various VoIP protocols.
• Drivers for PSTN interface cards and devices.
• Routing and call handling for incoming calls.
• Outbound call generation and routing.
• Protocol conversion (conversion from one protocol to another).
• Database integration for accessing information on relational databases.
• Web services integration for accessing data using standard internet proto-
cols.
This combination of components allows an integrator or developer to quickly
create voice-enabled applications. The open nature of Asterisk means that there
is no fixed limit on what it can be made to do. Asterisk integrators have built
everything from very small IP PBX systems to massive carrier media servers.
2.1.4 VoIP Gateway
Figure 2.6VoIP Gateway
A VoIP Gateway is a network device which convert voice calls between an IP
network and Public Switched Telephone Network (PSTN). There are two ways
VoIP gateways are used. First way is to convert incoming PSTN/telephone lines
to VoIP/VoIP telephone.the second way is to connect private branch exchange
(PBX)/Phone system to the IP network. The main functions of VoIP gateways
include voice compression or decompression, control signaling and call routing.
VoIP gateways cans also with additional features such as interfaces to external
controllers like Gatekeepers or Soft-switches and network management systems.
7

15. 2.1.5 VoIP Protocols
A signalling protocol is a common language spoken by telephones and call-
management servers, the PSTN, and legacy PBX systems as they communicate
to set up, monitor, and tear down calls[2].
H.323 Protocol
H.323, currently in Version 2, is the ITU-T’s recommendation for PBX-styled
signaling that supports a packet-based network. H.323 was originally proposed
as a solution for video conferencing using LANs. Later revisions saw it morph
into a full-fledged PBX-replacement plan. H.323 is implemented in Microsoft’s
NetMeeting conferencing software, as well as in Avaya’s MultiVantage and Nor-
tel’s Meridian IP-PBX systems. Most integrators who choose this standard
tend to choose a single vendor’s implementation across their entire network.
Incompatibility of H.323 implementations is a problem only when trying to link
opposing vendors’ H.323 systems together[3].
H.323 message packets are compact, and H.323 signaling is very fast, es-
pecially compared to SIP, which uses comparatively long and wordy message
packets. The design of H.323 is very much rooted in the design philosophy of
the PSTN: brevity and availability are striven. H.323 signals are short. The net-
work is used as little as possible to carry call signaling and as much as possible
to carry sound. The gatekeeper is the host on the network that provides cen-
tralized call monitoring and signaling capabilities for H.323 endpoints (called
terminals). It is common to refer to an H.323 gatekeeper as a softswitch or
softPBX[3].
IAX Protocol
The Inter-Asterisk Exchange Protocol (IAX), currently in its second revision.
The chief difference between IAX and the other signaling families is that IAX
doesn’t implement RTP as the packetizing mechanism. Instead, IAX has its
own way of packaging encoded voice.IAX is also (Network Address Translation)
NAT-proof, hundreds of simultaneous calls from behind a masquerading firewall
will function correctly, just as HTTP does[3].
IAX is implemented in a far-simpler and less application-exhaustive manner
than SIP and H.323. It is really intended just for telephony applications, while
H.323 and especially SIP, include far more extensibility. IAX is therefore much
more compact; complete implementations have been done with as little as 64
kb of object code. Unlike SIP and H.323, IAX is not a standards recommenda-
tion, but rather an independent protocol created by Mark Spencer, founder of
Digium[3].
8

16. SIP Protocol
The Session Initiation Protocol (SIP) was developed by the Internet Engineer-
ing Task Force (IETF) as a way of signaling multiuser distributed telephony and
messaging applications on an IP network.SIP has garnered much praise from IT
professionals, while suffering some criticism from traditional telecommunica-
tions people. The main reason for its less-than-perfect repute with telecom pros
is its origin outside the telecom world. But many telecom guys have had to for-
give this, because they’re learning that SIP has almost no shortcomings when
compared to its ITU-inspired cousin[3].
The essential duties and formulaic pieces of SIP are the same as H.323. There
are VoIP endpoints of varying capabilities, and there are servers that participate
in the signaling process and establish policy for the voice network. Unlike H.323,
however, SIP is far more extensible. It is more than just a set of voice and
video telephony protocols. Rather, it’s a packaging framework for all types of
message-based applications, from intercom calling to instant messaging and AV
services[3].
Avaya, Cisco, Siemens, Alcatel, and the major telephony hardware vendors
have indicated a strong support attitude about SIP, while some have even backed
away from H.323 investments. This bodes well for SIP’s future, and there are
already more SIP IP phones installed worldwide than there are H.323 ones. SIP
uses readable headers like SMTP or HTTP. SIP is currently in Version 2.0. Its
definition is found in RFCs 3261 through 3265[3].
• SIP Nodes
SIP endpoints and servers are called nodes. A SIP phone is a node. SIP
phones can communicate directly with each other in order to establish
media sessions, just as H.323 terminals can establish direct channels. But
more often than not, especially in an enterprise setting, SIP is used with
a SIP server. SIP phones normally report to a dedicated SIP server node
called a registrar upon boot-up[3].
• SIP Registrar
The SIP registrar is a database server that communicates with SIP
nodes in order to collect, store, and disperse information about the where-
abouts of SIP users. When a SIP node registers with a registrar, it tells
that registrar how to get hold of the user, specifically what IP address
and port to use for future SIP communication. You could think of the
registrar as a router, because its main purpose is to give advice on how to
reach SIP users, just as a TCP/IP router’s purpose is to give advice on
how to reach other networks[3].
• URIs
9

17. SIP endpoints can be referenced using Uniform Resource Indicators, but
so can SIP users. Consider this URI:
sip:lerxt@sip.bytor.com
This convention indicates both the user to be contacted and the server
that is expected to know the address of that user’s SIP endpoint. In this
case, the user is lerxt and the server is sip.bytor.com[3].
• SIP Methods and Responses
SIP signals fall into 10 categories called methods. Each method accom-
plishes a different function for SIP:
– INVITE: This method is used to start sessions and advertise endpoint
capabilities.
– ACK: This method is used to acknowledge to the called SIP peer
that an INVITE has succeeded.
– BYE: This method occurs when the call is completed, that is, one
user at a minimum wishes to end the call.
– CANCEL: This method is used during attempts to override a prior
request that hasn’t yet been completed.
– OPTIONS: This method is used to query a SIP peer for its capabil-
ities information, without actually establishing a media channel.
– REGISTER: This method notifies the SIP server at which endpoints
a particular user can be reached.
– INFO: This method is used to transmit telephony application sig-
nals through the SIP signaling path; these signals can include dialed
digits.
– PRACK: This method (Provisional ACK) is used to notify an end-
point of intent to set up a complex call without actually providing
an ACK. PRACK is the SIP equivalent of ”all is well.”
– SUBSCRIBE: This method provides a way of establishing event han-
dlers within SIP telephony applicationsi.e., ”Tell me when Bob misses
a call” or ”Tell Bob when I am registered with the server.”
– NOTIFY: This method delivers messages between endpoints as events
occuri.e., ”Bob missed a call.”
When a call must be started, ended, or altered, a SIP method is employed.
The SIP methods in the preceding list are similar in concept to the HTTP
methods GET and POST, and like HTTP, SIP expects response codes
when it sends a method.
In Figure 2.7, you can see that a (highly simplified) call from an Internet
host (A), in the form of a SIP INVITE, to 5150@oreilly.com, would ordi-
narily result in a 200 OK response, clearing the way for the call to begin.
• SIP Proxies
10

18. Figure 2.7200 OK[3]
A SIP proxy is a server that routes or redirects SIP INVITE methods
on behalf of one or more domains, just as a web server provides responses
to HTTP methods for certain domains. So, when an incoming call from
a foreign network is recognized, the SIP proxy’s job is to connect it to
the called user’s endpoint, if possible. Outbound SIP proxies serve the
task of connecting calls, but on behalf of a local network of SIP users.
Many users may share the same SIP proxy because they work in the same
office or perhaps because they subscribe to the same SIP dial-tone ser-
vice provider. Outbound SIP proxies are often used to overcome network
communications problems posed by NAT firewalls. Some local calls may
benefit from being routed through a SIP proxy, too. Forcing even local SIP
endpoints to use a SIP proxy allows for easy enforcement of a dial-plan,
greater administrative control over the voice network, and the ability to
do centralized telephony applications such as call recording[3].
• SIP User Agent Elements
All SIP proxies and endpoints are comprised of two key software ele-
ments, the user agent client and user agent server (UAC and UAS).
All SIP devicesbe they softphones, hardphones, voice mail servers, or
full- blown PBX servers must be able to speak the SIP protocol, and the
UAC and UAS elements are their mouthpieces. The UAC sends methods
and receives responses, so its logical equivalent in HTTP is the web
browser. The UAS receives SIP methods, processes them, and returns
responses, so it’s more like a web server. In varying degrees of
completeness, all SIP endpoints and servers have both a UAC and a
UAS[3].
• SIP redirect
11

19. When a SIP server responds to a calling endpoint’s INVITE method
with a 3xx response, that SIP server is redirecting the calling endpoint
to a different SIP server. The calling endpoint should then contact that
server with an INVITE method for further assistance in connecting the
media stream. This feature is not implemented on all systems that support
SIP. In fact, where complex, signaling-neutral dial-plan programming is
available (like in Asterisk), SIP redirection isn’t always necessary. The use
of SIP redirects is more common in large, SIP-only networks with multiple
servers, such as those that span the Internet[3].
• Real-Time Streaming Protocol
Unlike H.323, the SIP protocol family provides a built-in recommen-
dation for streaming prerecorded audio and video, in this case the RTSP
protocol. This is the same protocol used by RealOne and similar media
player applications. It’s defined by RFC 2326[3].
H.323 vs SIP vs IAX
H.323 SIP IAX
Standards body ITU-T IETF Digium, Inc.
Architecture Distributed Distributed, Peer-to-Peer Distributed
Built-in legacy
device support
recommended
Yes, via gateways No recommended No
Signalling Transport TCP/UDP TCP/UDP UDP
Multimedia Telephony, video Telephony, instant messaging, video Telephony
SoftPBX call path is called a... Gatekeeper-routed Proxy SoftPBX
Messaging approach ISDN-like HTTP-like Proprietary
Table 2.1Comparison between H.323 and SIP and IAX[2][3]
Conclusion
H.323 is older than SIP, more considerate of legacy infrastructure, and con-
tains many applications, but H.323 is more complex reverse SIP application
and be easy to understand and not complicated. One big advantage of SIP
is its comfort in non-telephony applications, such as instant messaging, video,
and white-boarding. Apple’s iChat and Microsoft’s MSN Messenger use SIP
for these purposes. Earthlink’s SIPShare application allows peer-to-peer media
sharing using SIP. Extensibility is an appealing characteristic of SIP. Lack of
extensibility is the drawback of H.323. Our project depends on the SIP protocol
to communicate through the Internet Protocol.
12

20. 2.1.6 PSTN
The public switched telephone network (PSTN) is the network of the world’s
public circuit-switched telephone networks. The network that connects all these
phones together wraps around the globe. The job of the PSTN (Public Switched
Telephone Network) is to reliably facilitate telephone conversations at any time
of day, year-round. The PSTN combines analog, digital, and electromechanical
data links. There are also private networks run by large companies which are
linked to the PSTN only through limited gateways, like a large private branch
exchange (PBX)[5].
2.1.7 GSM
Global System for Mobile communications (GSM) is the most popular stan-
dard for mobile phones in the world. GSM is used by over 3 billion people
across more than 212 countries. GSM is considered a second generation (2G)
mobile phone system, GSM EDGE is a 3G version of the protocol. Today’s
GSM platform is living, growing and evolving and already offers an expanded
and feature-rich ’family’ of voice and multimedia services[2].
2.2 Simulation
VoIP simulation try to provide a mirror image of what might be a VoIP
architecture in the real world. The objective of the simulation is the study of
VoIP and build models of VoIP architecture. It also tests the quality of service
offered to people so there is no flaws in the use of VoIP, and study the Latency,
Jitter, Errors Rate and Availability. For example of VoIP Simulation: SIPstone
is a benchmark for SIP (Session Initiation Protocol) proxy and redirect Servers
(SIP Server).
2.2.1 Simulation Types
The Real Test
Real test of VoIP be several or more people are communicating to the server
in order to test the performance of communication through IP. This is costly
and also physically impossible to bring many people.
Pure Simulation
Pure Simulation is also called Network Simulation is based on mathematical
equations or events that occur in the network, and can also edit the different
environments. Simulation makes the client and the server and the result of their
work. There is one drawback is that simulation is not simulate the real world.
Examples of notable network simulation software are:
ns2/ns3 (Network Simulator), OPNET (Optimized Network Engineering Tools).
13

21. Semi-Simulation
Semi - Simulation of VoIP is Depends on a server with a specific program,
such as Asterisk, this program is making many clients are communicating to the
server. Simulations here are making real clients are communicating through the
Internet protocol to the server for the quality of the performance of VoIP. Our
project will be based on this simulation for making large numbers of clients.
2.2.2 Metrics and Parameters to be reported
When simulation an architecture, the following data has to be provided.
• Description of the clustering configuration, including the number of hosts
and the load balancing mechanism used.
• All aspects of the server hardware.
• The server operating system and version and any non-standard tunings or
settings.
• The server software configuration.
• The number and type of load generators that participate in the run.
• The number and type of network segments connecting the load generators
to the SIP server.
• The arrangement of load generators on the network segments.
• The workload parameters except the load level.
To remove one variable from the measurement results, the server and
load generators should be connected by a switched 100Mb/s Ethernet that
only carries measurement-related traffic and no other LAN traffic.
• The number of connections requested by the clients and accepted by the
Server Under Test SUT per second;
• CPU and memory utilization of server at various loads;
• A curve plotting the Transaction Response Time (TRT) as a function of
request arrival rates;
• Calls per second (CPS) and Registrations per second (RPS);
14

22. 2.3 Related Work
Projects of VoIP Simulation are few, we will also talk about one project, it is
SIPstone. It is also regarded as standard. We will talk about how it was started
and the methods it used for getting the results and environments that use the
VoIP. Our simulation will use it to get advanced results in the VoIP parameters
to improve the performance of VoIP.
2.3.1 SIPstone - Benchmarking SIP Server Performance
SIPstone is designed by Henning Schulzrinne, Sankaran Narayanan, Jonathan
Lennox and Michael Doyle[4]. This section proposes an initial simple set of met-
rics for evaluating and benchmarking the performance of SIP proxy, redirect and
registrar servers. SIPstone is a benchmark for SIP (Session Initiation Protocol)
proxy and redirect Servers (SIP Server). SIPstone is designed to be a bench-
mark with a single standardized implementation and workload. The implemen-
tation performs a series of tests that generate the pre-configured workload. This
workload is performed in a controlled IP telephony test bed that simulates the
activities of multiple users initiating SIP calls. The benchmark currently is lim-
ited to evaluating the sustainable rate of what we believe to be a representative
workload, consisting of registrations, successful forwarding and unsuccessful call
attempts[4].
SIP supports user mobility by allowing addresses in the form of user@domain
to be used when making Internet calls. For instance, alice@home.com can call
bob@office.com no matter what communication device, IP address or phone
number bob is using currently. The locations where users could be reached are
maintained by the SIP location or registration servers. The performance of a
SIP server depends not only on the implementation but also the size of the
user population serviced, the number of DNS lookups required, the transport
protocol used (UDP or TCP), and the type and statistical arrival process of
requests submitted[4].
SIPstone describes a workload for SIP requests in proxies using a set of tests
that exercise various components of typical SIP servers.
Architecture
The ”server under test” (SUT) is a SIP proxy, redirect or registrar server
whose performance is to be estimated. The benchmark consists of a set of
SIPstone load generators that create the SIP request load, a call handler that
simulates a user agent server (UAS) and a central benchmark manager (”coor-
dinator”) that coordinates the execution of the benchmark, and the SUT. The
call handlers may run along with the load generators or on different systems[4].
15

23. Description of Tests
The benchmark currently consists of five tests. Each test is performed using
both UDP and TCP, with results reported separately for each transport proto-
col. Thus, a total of ten results is to be reported for servers that support both
transport protocols[4]. The individual tests are:
• Registration
The transaction delay is measured from sending the first REGISTER re-
quest to receiving the final 200 response, including the 401 ”Unauthorized”
message.
Figure 2.8Registration test
• Outbound proxy
The load generator sends an INVITE request to the SUT acting as an
outbound proxy. The message flow is the same as for the ”Proxy 200”
test below[4].
• Redirect
The load generator sends an INVITE request to the SUT acting as a
redirect server. It is assumed that all call destinations have registered
with that server. The transaction delay is measured from sending the
INVITE request to receiving the 3xx response[4].
• Proxy 480
The load generator sends an INVITE request to the SUT acting as a
redirect server. The call destinations are known to the server, but have
16

24. Figure 2.9Redirect test
not registered, so that the server returns a 480 (Temporarily unavailable)
response. The request is not authenticated.
• Proxy 200
The load generator alternates between INVITE and BYE transactions to
the SUT acting as a non-forking proxy server. The BYE transaction is
sent immediately after the INVITE transaction completes. The requests
are not authenticated. The server should insert a Record-Route to reflect
real operational conditions, however, the operation of the system does not
depend on this since the caller load generator always sends the request to
the SUT.
17

25. Figure 2.10Proxy 200 test
18

26. Chapter 3
Specification
3.1 Server(Asterisk)
In our project, we need a server for receiving a large number of callers each
one can listen to the answering machine for a certain period. For that we will
need to Asterisk server, which can provide the Interactive Voice Response(IVR)
system can receive large numbers of callers.
3.2 Simulate(SIPp)
The real tests in VoIP, to meet a large number of callers each one with a
computer and then call to the server at the same time in order to improve the
performance of communication and knowledge of its shortcomings. And this
is very expensive and also impossible to collect large numbers of callers. We
will need to SIPp which is doing the same test simulation of the real and also
generates a large number of distributors such as: 100.100000,..etc. And also we
can control the connection process, such as: increasing the number of callers
every 5 seconds and other complex scenarios.
19

27. Chapter 4
Design
In VoIP Simulation project design is on two stages , the first stage is on tools
there are available for do this project, the anther stage is how work this project
means what happen in inside this project.
4.1 Tools
We must have 4 tools for design the project:
4.1.1 Asterisk
The first stage in design for VoIP Simulation is in Asterisk server, so we start
in build Asterisk server. Two ways in build Asterisk server. First way we can
Installing Asterisk by go to Synaptic Package Manager then write asterisk and
install packages. Second way is write command: apt-get install (name package),
in terminal Linux. The Asterisk version is 1.6. Finally we can run asterisk by
[asterisk -R].
4.1.2 SIPp
The second stage in design is clients which clients call to server. SIPp can
generation clients and can call to Asterisk server, so in build SIPp is two ways
are the same as build in Asterisk server. Pre-requisites to compile SIPp are:
• gcc-c++: C++ Compiler.
• ncurses-devel: curses or ncurses library.
20

28. • openssl-devel and libssl-dev: For authentication and TLS support: OpenSSL
= 0.9.8.
• libpcap 0.8 and libnet: For pcap play support.
For install SIPp go to Synaptic Package Manager then write sip-tester(SIPp)
and install it. Now we can start SIPp and call to Asterisk server for VoIP
Simulation.
4.1.3 CPU RAM Analysis
The third stage is use System Activity Reporter(SAR) tool can measure CPU
and RAM state when clients call to server. In the build stage is the same as
tools in the above. the important SAR packages:
• debconf (0.5): Debian configuration management system.
• libc6 (2.3.4-1) : GNU C Library: Shared libraries and Timezone data also
a virtual package provided by libc6-udeb.
• dep: lsb-base (3.0-6) : Linux Standard Base 3.1 init script functionality.
• dep: ucf (2.003) : Update Configuration File: preserves user changes to
config files.
Now we can start in measure CPU and RAM state and wait for results.
4.1.4 Shell Script
The final stage is how method to run project in a particular case. We use
’sh’ shell script because we can run the tools in the same time, then extract
the results then put them in file for we can study the results and know how to
improve them.
4.2 How to Work
Here we will describe or explain what happen in the project, so we design the
project by Flow Chart even be a simple way to understand the project.
21

29. Figure 4.1VoIP Design
22

30. Chapter 5
Implementation
The idea of the project is first run asterisk for receive calls from users ,then
run SAR for CPU and RAM analysis, then run SIPp for generate the users,
then put the results from analysis to file for use in curve.
5.1 Asterisk
5.1.1 Configuration
The Asterisk configuration files for a standard install in the /etc/asterisk
directory:
sip.conf
In the /etc/asterisk/sip.conf file we can configure everything related with the
SIP protocol, add new sip users or define sip providers.
[ sip p ] type=f
r i en d con tex
t=s ip p host=d
ynamic p or t
=6000 u s er=si
p p
c a n r e i n v i t e=no
d is a l l o w=a l l a ll
ow=ulaw
The [sipp] section defines parameters for the sipp SIP extension. The parame-
ter type=friend simply means that this SIP extension can both accept and make
calls[11]. We are calling the context sipp in /etc/asterisk/extensions.conf; When
we look at that file in the next section, the application of contexts should become
clearer[11]. The term host=dynamic tells Asterisk that it doesn’t matter if the
23

31. IP address of the SIP extension sipp changes[11]. The port for SIP connections
is 6000. Then user=sipp: can only accept calls from sipp. Then Disallow all
codecs (global configuration). Finally allow codecs in order of preference.
Interactive Voice Response (IVR)
An interactive voice response system lets computer systems interact with tele-
phone callers, who provide input to the system either by pressing the keypad
on their telephone set or by saying something (natural language speech recog-
nition). Most IVR systems provide selection menus for routing calls without
requiring operator intervention, but modern IVR systems can also be very com-
plex applications that handle information or control equipment[10].
extensions.conf
The /etc/asterisk/extensions.conf – known as the (dialplan) – is the heart of
every Asterisk configuration. In a sense, we can equate them with a switch-
board used in early telephone systems. The dialplan determines which phones
can make calls to other phones, and how[11].
[ sip p ]
exten => 2005 ,1 , Answer
ex ten => 2 005 ,2 , SetMusicOnHold ( d ef a u l t )
exten => 2 0 05 ,3 , WaitMusicOnHold ( 20) ex te
n => 2005 ,4 , Hangup
The first section of the configuration is the context [sipp]. The caller dials ex-
tension 2005, Asterisk answers and plays default music then wait for 20 seconds
then close the call.
5.1.2 Asterisk Run Command
asterisk -R : This is show the Asterisk CLI (command line interface) which
lets us interactively control Asterisk, and also is important for run Asterisk
server. So we will start this command and after that we will stop this command
by core stop now.
asterisk -vvvvvc : This command run Asterisk server and display communi-
cation situations. asterisk : This command run Asterisk server in background.
5.2 SIPp
5.2.1 User Agent Client(UAC) Scenario
24

32. A s ce n a r io wi l l always s t a r t with :
<?xml v er s i o n =”1.0” en cod in g=”ISO−8859−1”?>
<!DOCTYPE s c e n a r io SYSTEM ” sip p . dtd”>
<s c e n a r io name=”uac3”>
And end with:
</s cen a r io >
<send r etr a n s =”500”>
I n i t i a t e tim er to 500 m i ll is e c o n d s when th e message i s s en t .
And end with :
</send >
Inside the ”send” command, you have to enclose your SIP message between
the ”[CDATA” and the ”]]” tags. Everything between those tags is going to be
sent toward the remote system. Those keywords in the SIP message are used
to indicate to SIPp that it has to do something with it.[7]
INVITE sip : [ s e r v i c e ]@[ r emote ip ] : [ r emote p or t ] SIP /2. 0
Here the INVITE message is sent service name, IP address and port.
Via : SIP /2 . 0 / [ tr a n sp or t ] [ l o c a l i p ] : [ l o c a l p o r t ] ; br anch =[ branch ]
The message is sent through protocol SIP, transport UDP, host IP address, port
chosen by the system and Provide a branch value is a concatenation of cookie
+ call number + message index.
From : sip p <s ip : sipp@ [ l o c a l i p ] : [ l o c a l p o r t ] >; ta g =[ ca ll n u m b er ]
The message is sent from caller number.
To : su t <s ip : [ s e r v i c e ]@[ r emo te ip ] : [ r emote p or t ]>
The message is sent to service.
C a ll−ID : [ c a l l i d ]
Caller ID.
CSeq: 1 INVITE Generates automatically the CSeq number.
25

33. Contact : s ip : sipp@ [ l o c a l i p ] : [ l o c a l p o r t ]
Here contact host IP address with port.
Content-Length: [len]
Computed length of the SIP body.
v=0
o=user 1 53655765 2353687637 IN IP [ l o c a l i p t y p e ] [ l o c a l i p ]
s=−
c=IN IP [ m ed ia ip typ e ] [ m ed ia ip ]
t=0 0
m=aud io [ m ed ia p or t ] RTP/AVP 0
a=rtpm ap :0 PCMU/8000
Here is owner name, service name, system name, type IP address, local IP
address for RTP echo. Depending on the values of parameters.
Now that the INVITE message is sent, SIPp can wait for an answer by using
the ”recv” command.
<r ecv r esp on s e=”100”> op tion a l=”tr u e ”
</r ecv>
<r ecv r esp on s e=”180”> op tion a l=”tr u e ”
</r ecv>
The 100 and 180 SIP messages can be received without being considered as
”unexpected”.
<r ecv r esp on se =”200” r td=”tr u e”>
</r ecv>
Record Route Set. if this attribute is set to ”true”, then the ”Record-Route:”
header of the message received is stored and can be recalled using the[routes]
keyword.[7]
100 and 180 messages are optional, and 200 is mandatory. In a ”recv” se-
quence, there must be one mandatory message.[7]
Then send the ACK, in send the ACK is almost the same as INVITE.
26

34. <send >
<![CDATA[
ACK s ip : [ s e r v i c e ]@[ r emote ip ] : [ r emote p or t ] SIP /2. 0
.
.
]] >
</send >
Then the scenario will wait for time is write in parameter -d.
<pause/>
And finish the call by sending a BYE and expecting the 200 OK:
<send r etr a n s =”500”>
<![CDATA[
BYE sip : [ s e r v i c e ]@[ r emote ip ] : [ r emote p or t ] SIP /2. 0
.
.
]] >
</send >
<r ecv r esp on se =”200” c r l f =”tr u e”>
</r ecv>
Displays an empty line after the arrow for the message in main SIPp screen.
<Resp on seT im eR ep a r tition va lu e =”10, 20 , 30 , 40 , 50 , 100 , 150 , 200”/>
response time values are distributed between 0 and 10ms, 10 and 20ms, 20 and
30ms,...etc.
<C a llLen gth R ep a r tition va lu e =”10, 50 , 100 , 500 , 1000 , 5000 , 10000”/>
call length values are distributed between 0 and 10ms, 10 and 20ms, 20 and
30ms,...etc.
5.2.2 SIPp Run Command
sip p −s f uac3 . xml − s 2005 1 2 7 . 0 . 0 . 1 −d 5000 −m 300 −l 300 −tr a ce m sg
−tr a c e s c r ee n −tr a c e s t a t −t r a c e e r r
27

35. • -sf: Loads an alternate xml scenario uac3.xml.
• -s : service name 2005.
• 127.0.0.1 : IP address.
• -d 5000 : Pause traffic, the scenario will wait for 5 second[12].
• -m 300 : Stop the test and exit when ’300’ calls are processed[12].
• -l 300 : Set the maximum number of simultaneous calls. Once this limit is
reached, traffic is decreased until the number of open call goes down[12].
• -trace msg : Displays sent and received SIP messages in scenario file name
ppid messages.log[12].
• -trace screen : Dump statistic screens in the scenario name ppid screens.log
file when quitting SIPp[12].
• -trace stat : Dumps all statistics in scenario name ppid.csv[12].
• -trace err : Trace all unexpected messages in scenario file name ppid er-
rors.log[12].
5.2.3 Screens
Several screens are available to monitor SIP traffic. You can change the screen
view by pressing 1 to 9 keys on the keyboard. In the project is three screens is
available:[7]
Key ’1’: Scenario screen. It displays a call flow of the scenario as well as some
important informations.[7]
Figure 5.1Key1
28

36. Key ’2’: Statistics screen. It displays the main statistics counters. The ”Cu-
mulative” column gather all statistics, since SIPp has been launched. The ”Pe-
riodic” column gives the statistic value for the period considered (specified by
-f frequency command line parameter).[7]
Figure 5.2Key2
Key ’3’: Repartition screen. It displays the distribution of response time and
call length, as specified in the scenario.[7]
Figure 5.3Key3
5.3 SAR
SAR is the System Activity Reporter. By interpreting the reports that SAR
produces, you can locate system bottlenecks and suggest some possible solutions
to those annoying performance problems. The Linux kernel maintains internal
counters that keep track of requests, completion times, I/O block counts, etc.
From this and other information, SAR calculates rates and ratios that give
insight into where the bottlenecks are[8]. And anther thing is System activity
29

37. data formatter(sadf ) extracts data in a format that can be easily ingested by a
relational database.
5.3.1 SAR Run Command
sa r −u −o s tr e s s 1 . sa r 1 184
−u : F or CPU sta te , −o : s or ted r e s u l t s on form b in a r y in f i l e ’ s tr e s s 1 . sa r ’
, 1 : D isp la y CPU s ta te in each 1 second , 184 : D isp la y 184 times of CPU s ta te .
sa r −r −o s tr e s s 2 . sa r 1 184
− r : F or RAM s ta te .
5.3.2 Screens
5.3.3 CPU
The %user and %system columns simply specify the amount of time the
CPU spends in user and system mode. The %iowait and %idle columns are of
interest to us when doing performance analysis. The %iowait column specifies
the amount of time the CPU spends waiting for I/O requests to complete. The
%idle column tells us how much useful work the CPU is doing. A %idle time
near zero indicates a CPU bottleneck, while a high %iowait value indicates
unsatisfactory disk performance.[8]
5.3.4 RAM
fields show the typical free and used memory statistics, with the percentage
of memory used displayed in the %memused field. The kbbuffers and kbcached
fields show how many kilobytes of memory are allocated to buffers and the
system-wide data cache. For swap utilization, the kbswpfree and kbswpused
fields show the amount of free and used swap space, in kilobytes, with the
%swpused field showing the swap space used as a percentage. commit is the
amount of physical memory (RAM) and virtual memory that is allocated to
all running programs, or applications, and the operating system itself. The
commit will increase when applications are opened and used and decrease when
applications are closed.[9]
5.4 SHELL SCRIPT
I used shell script for I can run several command in the same time and the
simulation and the analysis are stopped in the same time. The file name is
’script2.sh’.
30

38. #!/b in /sh
gnome−t er m in a l −e ’ sa r −u −o s tr e s s 1 . sa r 1 184 ’
gnome−t er m in a l −e ’ sa r −r −o s t r e s s 2 . sa r 1 184 ’
Firstly We run tow terminal are SAR command: first is for CPU performance
-u and anther is for RAM performance. The analysis is in each 1s and number
of analysis is 184 lines. then the results is stress1.sar by -o.
a=10
wh ile [ $a −l e 300 ]
do
sip p −s f uac3 . xml − s 2005 1 2 7 . 0 . 0 . 1 −d 5000 − r $a −m $a −l $a
−tr a ce m sg −tr a c e s cr e en −t r a c e s t a t −t r a c e e r r
a=$ ( ( a +10))
done
Secondly we use loop for in each call increase by 10 callers then stop the
communication at 300 callers. we use SIPp command for generation users then
they calls to name server in Asterisk 2005 in address 127.0.0.1, and pause or
duration is 5s, and -r is number of callers, and -m is the final call, and -l is the
limit. All -r, -m and -l are the same variable a.
sa d f −d s t r e s s 1 . sa r −− −u > s tr e s s 1 . csv sa
d f −d s tr e s s 2 . sa r −− −r > s t r e s s 2 . csv sed
’ s /;/ /g ’ s tr e s s 1 . csv > s tr e s s 1 1 . csv sed
’ s /;/ /g ’ s tr e s s 2 . csv > s tr e s s 2 2 . csv
Finally sadf is doing conversion binary file to file we can read the results for
CPU and RAM. Then sed is doing conversion ” ”tospacef orwecanuseinchart.
5.5 Performance Analysis
Here is the result from all the project work, we take informations from CPU
and RAM utilization by SAR and is on form Chart.
31

39. Figure 5.4User
Figure 5.5System
32

40. Figure 5.6Idle
Figure 5.7Memused
33

41. Figure 5.8Commit
34

42. Chapter 6
Conclusion
In conclusion, I learned a lot from VoIP Simulation through information,
tools and applications. And also learned how to complete the work by writing
reports, and implementation of the project several times. I also faced difficulties
during the writing and implementation of the project, and I also want to make
the project, particularly through the IVR system to be where a large database
and listen to more than one voice. In general, VoIP technology revolution has
become a large used everywhere because of the benefits of more of the benefits
of regular communications, and also most of the applications and tests on VoIP.
6.1 Summary
First, we talked about the general problem of VoIP, and also we talked about
Motivations and solutions. Then we talked about the state of the art of VoIP
describe each part of VoIP operation, also about types of simulation and Pa-
rameters to be reported. Then we talked about related work and simulation
that is we chosen, that is SIPstone. Then we talked about specification and
design for VoIP Simulation. Then finally we talked about implementation and
performance analysis of VoIP Simulation.
6.2 Feature Work
In feature works of VoIP Simulation are simulation and test are in the big
system specially in IVR system contain on database and more of sound file for
is the test very good, and also use Free PBX can call to server from any place.
35

43. References
1 http://en.wikipedia.org/wiki/Voice over Internet Protocol.
2 Implementation of VoIP System for Pilgrims in Al-Haj, Author: Zeyad A.
Al-Shaikh.
3 Switching to VoIP, Author: Theodore Wallingford, Publisher: O’Reilly.
4 SIPstone - Benchmarking SIP Server Performance, Authors: Henning
Schulzrinne, Sankaran Narayanan, Jonathan Lennox, Michael Doyle.
5 http://en.wikipedia.org/wiki/BSTN.
6 http://www.asterisk.org/asterisk.
7 http://sipp.sourceforge.net/doc3.0/reference.html
8 http://pagesperso-orange.fr/sebastien.godard/tutorial.html
9 http://www.redhat.com/docs/manuals/linux/RHL-9-Manual/admin-primer/s1-
memory-rhlspec.html
10 http://www.the-asterisk-book.com/unstable/kapitel-ivr.html
11 http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html
12 http://sipx-wiki.calivia.com/index.php/Using SIPp to run performance tests
36