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I. INTRODUCTION

In modern times when there is financial turmoil, the industries

are looking for ways to grade down the costs in view of

sustaining their profits. One way to cut down their cost is

cheap or rather cost less alternatives for hard line telephones.

Hence Soft phones has gained enormous popularity. Using

Soft Phones the industries can set up telephony systems that

will help them curb down their call costs. Also in recent times

with the advent of free open source SIP platforms like

Asterisk, Kamailo, and Freeswitch; the world of VOIP and SIP

phones have become a major player in the world of free

telephony.

Now as SIP phone are gaining popularity more and more,

the technicians are discovering new problems associated with

this technology. While using SIP phone services whether for

home use or business purpose in a one to one or conference

mode one often comes across problems like lost calls, bad call

quality, other line seems to be engaged while it is actually not

& also jump calls(calls to wrong telephone number) . There

have been multiple softwares produced for deciphering such

problems in SIP telephony. These softwares mainly deal with

analyzing the incoming and outgoing calls through the router

that are using the SIP protocol or the RTP or UDP protocols.

After the analyses of these protocols are done the problem is

pinpointed and can be dealt with effectively.

Two of the most acknowledged software in this field are

TCP Dump and Wireshark. While TCP Dump can only run on

UNIX platforms, Wireshark can be run on any platform. In

this project we are using Wireshark on Ubuntu Karmic Koala

platform for analyzing SIP protocol for various soft phones

used by industries as well as individuals in modern times.

II. SCHEME OF THE PROJECT

In this project we will be analyzing the different SIP/VOIP

solutions using Wireshark that are available in the market and

then give a detailed report on the QoS (quality of service) as a

comparison for all these solutions. The solutions that we will

test are:

1. Empathy: This is an instant messaging client which

supports text, voice, video, file transfers, and inter-

application communication over various IM

protocols.

Empathy also provides a collection of re-usable

Graphical User Interface widgets for developing

instant messaging clients for the GNOME desktop.

It is written as extension to the Telepathy

framework, for connecting to different instant

messaging networks with a unified user interface.

Empathy has been included in the GNOME

desktop since version

2. Ekiga : This was formerly called gnome is a VoIP

and video conferencing application for GNOME

and Windows . It is distributed as free software

under the terms of the GNU General Public

License. se" Ekiga supports both the SIP and

H.323 (based on OpenH323 protocols )and is fully

interoperable with any other SIP compliant

application and with Microsoft NetMeeting . It

supports many high-quality audio and video

codecs.

3. BigBlueButton = The BigBlueButton is a versatile

open source project that is built over fourteen open source components to create an integrated web conferencing system that runs on mac, unix or pc computers. some of the features of this softphone are web cam management, presentation in which any user can upload PDF presentation, office document and keep everyone in sync with their current page, zoom, pan, and see the presenters mouse pointer. BigBlueButton voice conferencing supports voice over IP (VOIP) conferencing out-of-the-box.

Performance analysis of open source solutions using

Wireshark

Koolwal Jai ,Pal Sumalya

Abstract— The goal of this project is to form a detailed analysis of

Multi point video, audio, text and collaboration softwares like the

Ekiga , Empathy and BigBlueButton ; and will be completing a

survey, comparing them on issues like robustness in call quality,

video quality in one to one as well as conference mode , for this

purpose we shall deploy the WIRESHARK (packet sniffer for

Linux) and building on the data collected we shall have a better

comparative understanding about the performance of these

softwares in different open source environments like UBUNTU &

FEDORA.

KeyWords-wireshark,bigbluebutton,ekiga,empathy, packet,

networking, linux, alpine linux, fedora, ubuntu,Voip, web

conference, data, Multi point video, audio, text

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III. WIRESHARK

Wireshark is the world's foremost network protocol analyzer,

and is the de facto standard across many industries and

educational institutions. The features of this tool are:

Live capture of packets and their offline analysis.

Standard three pane packet browser.

Runs on multiple platforms:

Windows, Linux, Solaris, NetBSD,

FreeBSD.

Captured network data can be

browsed via a GUI, or via the TTY-

mode TShark utility

The most powerful display filters in

the industry

Rich VOIP/SIP protocol analysis.

Deep inspection of hundreds of

protocols added every day.

IV. PROTOCOLS ANALYZED

V. THE PROCESS

First we start the wireshark protocol analyzer and start the packet

capture mechanism as shown in Fig 1.

Fig 1. The wireshark capture window

Then we start the Soft phone and dialed a toll free

number (we dialed 001-800-457-7777, the toll free

number of Toshiba service center). As soon as the

number is dialed we could see wireshark capturing

the RTP (Real time protocol) packets.

After about three to four minutes, we stopped the

capturing process and started analyzing the packets

in offline mode.

We did this for two SIP phones. First we

analyzed Ekiga and then Empathy.

VI. RESULTS

Fig. 2 Table showing the details of the SIP transaction

The above table gives us the detailed account of the SIP call

made from Empathy. We can see that the jitter accounted for

is very stable and is quite acceptable.

Next we will fetch another table like this one for Ekiga

and will be able to analyze in a comparable manner between

these two SIP services.

Fig 3. Graph showing the Forward Jitter and the Reverse

jitter in Empathy.

The above graph shows the forward jitter in the call. Note

that no green spikes can be seen in the graph. Green spikes

represent the reverse jitter. Since there was no answer from our

side , there were no reverse jitter experienced. Hence there are

Packet

Sequenc

e

Time

stamp

Delta

(ms)

Jitter(ms) Skew

(ms)

IP BW

(Kbps)

Mark

er

Status

1 623 4477 44860

0

0 0 0 1.6 SET [OK]

2 625 4478 44876

0

19.8

3

0.01 0.17 3.2 [OK]

3 627 4479 44892

0

19.9

6

0.01 0.21 4.8 [OK]

4 629 4480 44908

0

20 0.01 0.21 6.4 [OK]

5 632 4481 44924

0

20.5 0.04 -0.28 8 [OK]

6 633 4482 44940

0

19.7

9

0.05 -0.08 9.6 [OK]

7 636 4483 44956

0

20 0.05 -0.08 11.2 [OK]

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no green spikes.

Fig 3. The analyzed call spikes in Empathy.

The above figure represents the Call graph. Here only one

channel seems to have the spikes. This is because only the

operator on the other side of the phone talked.

VII. THE MATHS INVOLVED

. Wireshark calculates jitter according to RFC3550 (RTP):

If Si is the RTP timestamp from packet i, and Ri is the time of

arrival in RTP timestamp units for packet i, then for two

packets

i and j, D may be expressed as

D(i,j) = (Rj - Ri) - (Sj - Si) = (Rj - Sj) - (Ri - Si)

The interarrival jitter SHOULD be calculated continuously as

difference D for that packet and the previous packet i-1 in

order

of arrival (not necessarily in sequence), according to the

formula

J(i) = J(i-1) + (|D(i-1,i)| - J(i-1))/16

RTP timestamp: RTP timestamp is based on the

sampling frequency

of the codec, 8000 in most audio codecs and 90000 in

most video codecs

As the sampling frequency must be known to correctly

calculate jitter

it is problematic to do jitter calculations for dynamic

payload types

as the codec and it's sampling frequency must be known

which implies

that the setup information for the session must be in the

trace and the codec

used must be known to the program(with the current

implementation).

If suppose we have the following sample data –

R0 = frame 624: frame.time = Jul 4, 2005 11:56:25.348411000

S0 = frame 624: rtp.timestamp = 1240

R1 = frame 625: frame.time = Jul 4, 2005 11:56:25.418358000

S1 = frame 625: rtp.timestamp = 1400

R2 = frame 626: frame.time = Jul 4, 2005 11:56:25.421891000

S2 = frame 626: rtp.timestamp = 1560

we also have rtp.p_type = ITU-T G.711 PCMA (8) and thus we know sampling clock is 8000Hz and thus the unit of rtp.timestamp is 1/8000 sec = 0.000125 sec .

Then this is how we shall calculate the JITTER

frame 624:

J(0) = 0

frame 625:

D(0,1) = (R1 - R0) - (S1 - S0)

= [in seconds] (.418358000 sec - .348411000 sec) - (1400 * 0.000125 sec - 1240 * 0.000125 sec) = 0.049947

J(1) = J(0) + (|D(0,1)| - J(0))/16

= [in seconds] 0 + (|0.049947| - 0)/16 = 0.0031216875

frame 626:

D(1,2) = (R2 - R1) - (S2 - S1)

= [in seconds] (.421891000 sec - .418358000 sec) - (1560 * 0.000125 sec - 1400 * 0.000125 sec) = -0.016467

J(2) = J(1) + (|D(1,2)| - J(1))/16

= [in seconds] 0.0031216875 + (|-0.016467| - 0.0031216875)/16 = 0.00395576953125

How bandwidth (BW) is calculated

The BW column in RTP Streams and RTP Statistics dialogs shows the bandwidth at IP level for the given RTP stream. It is the sum of all octets, including IP and UDP headers (20+8 bytes), from all the packets of the given RTP stream over the last second.

VIII. FUTURE OF OUR PROJECT

We intend to perform the packet analysis on EKIGA,

EMPATHY on both FEDORA and UBUNTU with a

permutation and combination ensuring we are able to compare

each soft phone on each OS & hence get a better insight into

both the soft phones and the robustness of the OS.

REFERENCES

[1] A. Leon-Garcia, I. Widjaja, Communication Networks: Fundamental

Concepts and Key Architectures, 2nd ed., New York: McGraw-Hill,

2004, pp. 706-756.

[2] “QoS Assessment of Video Over IP.” [Online]. Available:

http://encyclopedia.jrank.org/articles/pages/6873/Qos-Assessment-of-

Video-Over-IP.html

[3] “Ethernet Capture Setup.” [Online]. Available:

http://wiki.wireshark.org/CaptureSetup/Ethernet

[4] WIRESHARK WEBSITE www.wireshark.org