Reaction paper 01

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REACTION PAPER 01
TEL 500
SAMEER DESHMUKH
9/12/13
REACTION PAPER 01
Session Initiation Protocol Improvement Using Inter-Asterisk
eXchange
Introduction:
Within the VoIP network environment, H323, SIP and IAX are three protocols that
solve the problem of voice packet signaling. In this paper, SIP and IAX are compared
based on use of bandwidth as a critical factor. Basic requirements for an audio
application conversational, Possible to use audio codec’s, VoIP conversations overhead
caused by communication protocols involved in, Analysis of the format of SIP and IAX
frames with its own package are also taken into account by applying them for theoretical
calculations of bandwidth consumption.
VoIP Protocols:
There are two types of VoIP protocols. Centralized and distributed. The above
mentioned protocols are distributed type which allow network intelligence. It is
distributed between the call control devices, VoIP gateways, IP phones, media servers
or any device that can initiate and terminate a VoIP communication.
VoIP Architecture Models:
Application
Presentation
Session
Transport
Network
Data link
Physical
Asterisk
G.729/G711/GSM/Speex
H323/SIP/MGCP/IAX
UDP/RTP/SRTP
IP/CBWFQ/WRED/IP
Precedence/Diffserv
Frame-Relay/ATM/PPP/Ethernet
Ethernet/V.35/RS-232/Xdsl
Architectural model of VoIP protocols within the OSI model
SIP (Session Initiation Protocol) Protocol:
It is a session layer protocol developed and standardized by the IETF (Internet
Engineering Task Force). It is based on the mechanism of “request and response” to
initiate a communication session, allowing sessions and video data between two
endpoints. After establishing a session, average flow of transfer is handled by RTP
protocol.
IAX (Inter Asterisk eXchange) Protocol:
The two main objectives of the project IAX derived from experience with VoIP
protocols like SIP and MGCP for control and RTP media stream such as minimizing the
use of bandwidth with specific emphasis on voice conversations, provide NAT
transparency and structure, signaling and media flow transfer to IAX.
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SAMEER DESHMUKH
9/12/13
REACTION PAPER 01
IAX uses UDP port 4569 to communicate all the packages. Trunked IAX protocol
allows the use a single header for the passage of several calls. However, so far IAX
trunked mode can only be enabled between two Asterisk servers.
Parameters Required For The Analysis Of The Use Of The Bandwidth:
Codec Sample Interval (CSI) (ms)
Codec Sample Size (CSS) (Bytes)
Codec Bit Rate (CBR) (kbps): is calculated by following expression:
CBR =
Voice Payload Size (VPS)
Packets per Second (PPS): is calculated by the following expression:
PPS =
Total Packet Size (TPS) (bytes): is given by the following expression:
TPS = L2 + IP + UDP +L5 + VPS
Where,
L2 = size of the protocol header data link layer
L5 = size of the session protocol header
Bandwidth (BW) (Kbps): the bandwidth required for n conversations full duplex
BW n = BW * n * 2
BW Calculation Protocol SIP – codec G.711
Input data: CSI = 10 ms,
BW30calls = 87.2 Kbps x 30 x2 = 5232
CSS = 80 bytes, VPS = 160 bytes
Kbps (more than 5 Mbps)
BW Calculation Protocol SIP – GSM codec
Input data: CSI = 10 ms,
BW30calls = 36.4 Kbps x 30 x 2 =2184
CSS = 80 bytes, VPS = 160 bytes
Kbps (near 2 Mbps)
IAX Protocol – G.711 codec
Input data: CSI = ms,
BW30calls = 84 Kbps + 65.6 Kbps x 29
CSS = 80 bytes, VPS = 160 bytes,
=1986.4 Kbps (near 2 Mbps)
CBR = 64 Kbps, VPS (ms) = 20 ms,
PPS = 50
H.IAX Protocol – codec GSM
Input data: CSI = 20 ms,
BW30calls = 33.2 Kbps + 14.8 Kbps x 29
CSS = 33 bytes, VPS = 33 bytes,
= 462.4 Kbps (Less than 512 kbps)
CBR = 13.2 Kbps, VPS (ms) = 20 ms,
PPS = 50
Results according applied protocol
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SAMEER DESHMUKH
9/12/13
REACTION PAPER 01
Traffic Generation: Call Simulator:
The simplest way to make calls is through configuration files. For this, you need
to configure a file extension *.call by call. Thus, 30 files must be configured to simulate
the same number of words.
Channel, context, extension and priority are the basic parameters. The files must
be located in the /var/spool/asterisk/outgoing, but must be created in a different location.
As soon as they are moved to /var/spool/asterisk/outgoing, the center recognizes and,
according to the parameters described in them, the call is made.
General Procedure:
For this analysis we perform the following steps:
 Vyatta system installation on computers that act as routers.
 Configuring interfaces and static routes on the Vyatta.
 Installing CentOS 5.2 operating system on servers Asterisk
 Asterisk 1 and 2.
 Installing Asterisk PBX on the servers 1 and 2.
 Installing Wireshark and Unsniff sniffer on the machine.
 SIP settings 30 extensions (101-130) and 30 IAX extensions (201-230) in the
Asterisk server 2 (Editing sip.conf files and iax.conf, respectively)
 Dial plan configuration on the Asterisk server 2, so that when receiving a call to
run a default recording that will simulate the flow of media (Edit the
extensions.conf file)
 60 files were created with.
 Call in the location / var / spool / asterisk / tmp on the Asterisk server 1, 30 for
SIP (101.call - 130.call) and 30 for IAX (201.call - 230.call) with which to generate
calls from the server 1 to 2.
 Dial plan configuration on server 1, creating a context, extension and symbolic
priority from where they will launch the Asterisk server calls 2.
 Asterisk software to run on both PBXs.
 SIP Analysis with Wireshark
 It starts a capture in Wireshark by checking the appropriate choice of the network
interface through which data must be captured.
 Should move, not copy-files. Call (101.call - 130.call) to / var / spool / asterisk /
outgoing, must keep a backup of those files for later use.
 With the command "show channels" on the Asterisk console can be displayed in
real time, 30 channels.
 When the call is finished, stops the capture of Wireshark and saved.
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SAMEER DESHMUKH
9/12/13
REACTION PAPER 01
Conclusions:
Protocol
SIP
SIP
IAX
IAX
Codec
G.711
GSM
G.711
GSM
BW Theor
5232 Kbps
2184 Kbps
1986.4 Kbps
462.4 Kbps
BW Real
5131 Kbps
2087 Kbps
2166 Kbps
552 Kbps
Theoretical results for the use of bw
The telephone system configured with SIP and G.711 codec has the highest
consumption of bandwidth; this is due in large part to the size of the SIP header and the
low rate compression codec G.711. This configuration, however, presents a very good
quality voice, making it ideal for PBX with a relatively low level of traffic.
The telephone system configured with IAX protocol and GSM codec has the
lowest consumption of bandwidth, due to reuse of the headwaters of the network and
transport layer and the high compression rate over G.711 GSM. This configuration
presents an acceptable voice quality, but at times of high traffic may present distortions.
The telephone system configured with G.711 codec IAX and is ideal for power or
compatible with Asterisk IAX whose traffic level is relatively high, since it has a good
voice quality but requires a bandwidth as high.
The telephone system configured with SIP and GSM codec is ideal for plants that
do not support IAX, presenting the same advantages that the previous configuration.
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