Computer Communication Project Interim Report 3

8/9/2019 Computer Communication Project Interim Report 3

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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: Thisis 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-usableGraphical User Interface widgets for developinginstant messaging clients for the GNOME desktop.It is written as extension to the Telepathy

framework, for connecting to different instantmessaging networks with a unified user interface.Empathy has been included in the GNOME

desktop since version

2. Ekiga : This wasformerly 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 versatileopen source project that is built over fourteen opensource components to create an integrated webconferencing system that runs on mac, unix or pccomputers. some of the features of this softphoneare web cam management, presentation in whichany user can upload PDF presentation, officedocument and keep everyone in sync with theircurrent page, zoom, pan, and see the presentersmouse pointer. BigBlueButton voice conferencingsupports voice over IP (VOIP) conferencing out-of-the-box.

Performance analysis of open source solutions using

Wireshark

Koolwal Jai ,Pal Sumalya

Electrical and Computer Science Engineering department.Universit O Florida

AbstractThe goal of this project is to form a detailed analysis of

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

Ekiga , Empathy, Twinkle & 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

9.10J AUNTY BUI LD Karmic Koala & Open Suse 11.2.

KeyWords-wireshark,bigbluebutton,ekiga,empathy, twinkle, CentOS, packet, networking, linux, alpine linux, fedora, ubuntu,Voip,

eb conference, data, Multi oint video, audio, text


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4. Twinkle - Twinkle is a free software open sourcesoftware application forVoice over Internet

Protocol (VoIP) voice communications in IP networks,

such as theInternet.It is designed

forGNU/Linux operating systems and uses the Qt toolkit

for its graphical user interface. For call signaling it

employs the Session Initiation Protocol. It also features

direct IP to IP calls. Media streams are transmitted viathe Real-time Transport Protocol which may be encrypted

with the Secure Real-time Transport Protocol (SRTP) and

the ZRTP security protocolsTwinkle supports the

following audio codecs.

G.711 A-law (64 kbps payload, 8 kHz sampling rate)

G.711 -law (64 kbps payload, 8 kHz sampling rate)

GSM (13 kbps payload, 8 kHz sampling rate)

Speex narrow band (15.2 kbps payload, 8 kHz

sampling rate)

Speex wide band (28 kbps payload, 16 kHz sampling

rate)

Speex ultra wide band (36 kbps payload, 32 kHz

sampling rate)

iLBC (13.3 or 15.2 kbps payload, 8 kHz samplingrate)

G.726 (16, 24, 32 or 40 kbps payload, 8 kHz sampling rate

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 ofprotocolsadded every day.

IV. THE PROCESS

First we start the wireshark protocol analyzer

and start the packet capture mechanism asshown 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 followed this process for two SIP phones.

First we analyzed Empathy and then Ekiga.

V.RESULTS

i)EMPATHY

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

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

Fig. 2 Table showing the details of the SIP transaction

Packet Sequenc

e

Time

stamp

Delta

(ms)

Jitter(ms) Skew

(ms)

IP BW

(Kbps)

Mark

er

S

1 623 4477 44860

0

0 0 0 1.6 SET [

2 625 4478 44876

0

19.8

3

0.01 0.17 3.2 [

3 627 4479 44892

0

19.9

6

0.01 0.21 4.8 [

4 629 4480 44908

0

20 0.01 0.21 6.4 [

5 632 4481 44924

0

20.5 0.04 -0.28 8 [

6 633 4482 44940

0

19.7

9

0.05 -0.08 9.6 [

7 636 4483 44956

20 0.05 -0.08 11.2 [
http://en.wikipedia.org/wiki/Free_softwarehttp://en.wikipedia.org/wiki/Open_source_softwarehttp://en.wikipedia.org/wiki/Open_source_softwarehttp://en.wikipedia.org/wiki/Voice_over_Internet_Protocolhttp://en.wikipedia.org/wiki/Voice_over_Internet_Protocolhttp://en.wikipedia.org/wiki/Internethttp://en.wikipedia.org/wiki/GNU/Linuxhttp://en.wikipedia.org/wiki/Qt_(toolkit)http://en.wikipedia.org/wiki/Session_Initiation_Protocolhttp://en.wikipedia.org/wiki/Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/ZRTPhttp://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Real-time_Transport_Protocolhttp://en.wikipedia.org/wiki/Session_Initiation_Protocolhttp://en.wikipedia.org/wiki/Qt_(toolkit)http://en.wikipedia.org/wiki/GNU/Linuxhttp://en.wikipedia.org/wiki/Internethttp://en.wikipedia.org/wiki/Voice_over_Internet_Protocolhttp://en.wikipedia.org/wiki/Voice_over_Internet_Protocolhttp://en.wikipedia.org/wiki/Voice_over_Internet_Protocolhttp://en.wikipedia.org/wiki/Open_source_softwarehttp://en.wikipedia.org/wiki/Open_source_softwarehttp://en.wikipedia.org/wiki/Open_source_softwarehttp://en.wikipedia.org/wiki/Free_software


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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

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.

ii) TWINKLE

We executed Twinkle SIP phone on the Open Suse 11.2

platform. The overall performance of the product was

satisfactory. However the forward and the reverse jitter

encountered were more than what we expected. The call clarity

was perfect.

The below figure shows the CALL FLOW of the entire

process.

Fig 4: The flow graph of the process

Packet Sequen

ce

Time

stamp

Delta

(ms)

Jitter(m

s)

Skew

(ms)

IP BW

(Kbps)

33 52496 3752872

392

0 0 0 0.58

36 52497 3752855

2

28.17 0.51 -

8.17

1.17

39 52498 3752871

2

48.48 1.82 -

29.6

5

1.75

40 52499 3752887

2

8.33 2.44 -

17.9

8

2.34

43 52500 3752873

032

42.13 3.67 -

40.1

1

2.92

44 52501 3752873

192

0.17 4.68 -

20.2

8

3.5

46 52502 3752873

352

21.43 4.48 -

21.7

1

4.09

Fig 5: The datas collected

The above figure shows the datas related with the entire

operation of the product. The relative average Jitter

encountered was 11.08 ms. This was quite high as compared to


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other opensource soft phones like that of X-Lite. However in

totality this product functioned satisfactorily.

iii) EKIGA

Ekiga was installed on UBUNTU 9.10JAUNTY BUILD

and just like in the previous cases we started wireshark prior to

the initiation of the VOIP call. Once the call was in progress

we collected the following data

Fig 6Showing the wireshark packet capture window.

Fig 7 Shows the table with packet jitter , BW as well as

SKEW.

I/O Graph showing packets/unit time v/s time

X axisTicker/Time (in sec)

Y axisPackets per unit Ticker(or Time)

Fig 8I/O Graph during the INITIATION of the VOIP CALL

Since the VOIP call was being established, hence we see very

few packets in the beginning causing a dip to almost zero

packets per unit time, the reason being that ARP was mappingthe IP address 192.168.113.2 to the MAC address of the

machine 00:50:56:f9:15:47 once this was completed it

gradually settled down to a constant rate with the UDP

successfully resolving source and destination ports. Also we

see a spike at exactly 3.169s because this is when DNS comes

into play and is trying to locate the Ekiga server using

STANDARD QUERY RESOLUTION.


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Fig 9I/O graph well after ESTABLISHMENT of the VOIP

CALL

Figure 9 shows the exceptionally stable performance of

EKIGA with a constant rate of packets along the time

axis.However we do observe a major spike in between which

must be because of a sudden surge in the available BW at 613s

, the jump being from mid 159.04 kbps to 165.85 kbps

allowing flow of more packets.

Fig 10 - I/O Graph during the TERMINATION of the VOIP

CALL

Finally we see the packets per unit time dipping to zero in

figure 10 since this is when we terminated the call.


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Packet Sequen

ce

Time stamp Delta

(ms)

Jitter(m

s)

Skew

(ms)

IP BW

(Kbps)

607 25716 41369328 33.1

9

893.5

1

438

931.

75

153.0

9

608 25717 41453412 36.5

1

893.7

8

439

829.

5

159.5

609 25718 41537496 38.0

7

893.9

3

440

725.

69

159.7

4

610 25719 41621580 45.3

3

893.6

2

441

614.

62

159.0

4

611 25720 41705664 37.2

4

893.8

3

442

511.

66

159.8

7

612 25721 41789748 47.0

9

893.4

1

443

398.

81

158.9

9

613 25722 41873832 35.9

7

893.7

2

444

297.

12

165.8

4

Fig 11Table with JITTER, BW, SKEW and TIMESTAMP

of given packets

The obviously visible fact about the above table is the almost

constant jitter. Since we know that Jitter is a variation of delay

and hence a constant jitter will not hamper the performance of

a voip call and here we see that EKIGA has almost constant

jitter throughout the duration of the call , which is just one ofthe reasons why the call quality was excellent.

VI. 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





we also have rtp.p_type = ITU-T G.711 PCMA (8) and thus we knowsampling clock is 8000Hz and thus the unit of rtp.timestamp is 1/8000sec = 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 thebandwidth 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 thegiven RTP stream over the last second.

VII. PROBLEMS THAT CROPPED UP AND HOW WE

RESOLVED THEM

Fedora Issue: we in the beginning thought of using Fedora as

a test benchmark for this project alongside Ubuntu 9.11, since

it is one of the oldest LINUX versions in the market. But as we

proceeded with our project we found out that Fedora has some

generic problem with the ITU-T G.711 PCMA audio codec.

There were a lot of glitches in the call quality which was not

the fault of the VOIP solutions that we were benchmarking.

Hence to get a better understanding of the sip phones under


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scrutiny we replaced Fedora with Open Suse 11.2 operating

system.

Cent OS issue: Next we planned to build a BigBlueButton

server on a Cent OS (Community Enterprise Operating

System) and subsequently benchmark the open source

solution. But as we proceeded with our project we found out

that the CentOS has several issues with the video cards. It

seems CentOS is not compatible with Voodoo 3 range videocards. Hence there was a problem when installing the OS.

Even after we installed it with in text mode (Linux text mode),

the GUI could not recognize mouse or legacy keyboard.

Reassigning of the UDP Ports

While performing the test we sometimes were running two sip

phones in parallel on the same machine, and by default a VOIP

call routes through the UDP PORT 5060, however when

running two phones at the same time one of the phones

occupies the port rendering it inaccessible for the second sip

phone. Hence we must make sure that we assign a different

port ranging between 5061 to 5069 for the second VOIP call.

This solved the issue faced and allowed mooth running ofbothe the SIP clients.

Initial trouble with EKIGA and how we came around that

problem

When we started out, we used a sip account to make calls.

Although UBUNTU 9.10 JAUNTY BUILD, our SIP

account was displayed as registered and our calls were also

going through, when calling each other, we could never hear

any voice from either side. This issue was resolved by

changing the settings of going into EKIGA

PREFERENCES>AUDIO>DEVICES and in the directory we

change the ringing device from SILENT (Ekiga/Ekiga) toDefault (PTLIB/ALSA).

VII. FUTURE OF OUR PROJECT

In future we will be installing BigBlueButton in UBUNTU

9.10 JAUNTY BUILD environment. We will also try and

setup a BigBlueButton server and examine its performance in

UBUNTU 9.10 JAUNTY BUILD Karmic koala

environment. We intend to perform the packet analysis on

EKIGA, EMPATHY on both Open Suse 11.2 and UBUNTU

9.10 JAUNTY BUILD 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.

IX.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 WEBSITEwww.wireshark.org

[5] How VOIP works http://www.fcc.gov/cgb/consumerfacts/voip.pdf

[6] Open Source Voip Traffic Monitoring ,Luca Deri

[7] Understanding VOIP Over Internet Protocol,Mathew Desantis US Cert

[8] Security Considerations For Voice Over IP Systems -NIST
http://wiki.wireshark.org/CaptureSetup/Ethernethttp://wiki.wireshark.org/CaptureSetup/Ethernethttp://www.wireshark.org/http://www.wireshark.org/http://www.wireshark.org/http://www.fcc.gov/cgb/consumerfacts/voip.pdfhttp://www.fcc.gov/cgb/consumerfacts/voip.pdfhttp://www.fcc.gov/cgb/consumerfacts/voip.pdfhttp://www.fcc.gov/cgb/consumerfacts/voip.pdfhttp://www.wireshark.org/http://wiki.wireshark.org/CaptureSetup/Ethernet

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Computer Communication Project Interim Report 3