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VoIP in IRIS IVDX

Coral Telecom Ltd.
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What is Voip
VoIP is the ability to make telephone calls and send
faxes over IP-based data networks instead of
traditional PSTN, with a suitable quality of service
(QoS) and superior cost/benefit
Voip Technology involves compressing and digitizing
our analog voice signals and sending them as
individual packets across an IP Network to the
Recipient Voip terminal .The Recipient Voip terminal
then decompress the voice packets and makes it
audible to the end use.
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VoIP Standards
There are two types of Voip protocol standard

H.323
The circuit switch based protocol

SIP Session Initiation Protocol
SIP is a text based protocol similar to http

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Elements of the H.323 Technology
H.323 Terminal
H.323 Gateway
H.323 Gatekeeper


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H.323 Terminal
Used for real-time bi-directional multimedia
communications, an H.323 terminal can either
be a personal computer (PC) or a stand-alone
device, running an H.323 and the multimedia
applications. It supports audio communications
and can optionally support video or data
communications


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H.323 Gateway
A gateway connects two dissimilar networks.
An H.323 gateway provides connectivity
between an H.323 network and a nonH.323
network. For example, a gateway can
connect and provide communication between
an H.323 terminal and a PSTN network

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H.323 Gatekeeper
A gatekeeper can be considered the brain of
the H.323 network.. Although they are not
required, gatekeepers provide important
services such as addressing, authorization and
authentication of terminals and gateways;
Gatekeepers also take care of bandwidth
management, accounting, billing and charging.


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H.323 call flow (Gateway to gateway)
Iris gateway far end gateway
Setup
Call proceeding
Alerting
Connect
Release complete
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SIP (Session Initiation Protocol)
The Session Initiation Protocol (SIP) is an
application layer control protocol that can
establish and terminate multimedia session
or calls.
SIP components -The main components of a
SIP environment fall into two primary
categories, SIP servers and SIP user agents.

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SIP Servers

There are three types of SIP servers: proxy, registrar,
and redirect. Each type of server performs a different
function
Proxy Server -
Acts as an intermediary between a SIP user agent client
and a SIP user agent server. The proxy server performs
the functions of either a SIP user agent client or a SIP user
agent server, depending upon the direction of the
communication between client and server.Request are
serviced internally or by passing them on,possibly after
translation,to other servers.a proxy interprets,and if
necessary rewrites a request message before forwarding
it.

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SIP Servers
Registrar server
Receives REGISTER requests. This allows the registrar
server to keep track of the location of user agents from
which the registrar server has received REGISTER
requests.A registrar is typically co-located with a proxy or
redirect server and may offer location services.
Redirect server
Accepts initiation, in the form of a SIP INVITE request, of a
SIP session from the calling user agent, obtains the correct
SIP address of the called user agent, and replies to the
calling user agent with the correct SIP address. The calling
user agent then uses the correct SIP address to directly
initiate a SIP session with the called user agent.unlike a
proxy server it does not initiate its own SIP request.unlike a
user agent server,it does not accept calls.

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SIP User Agents

There are two types of SIP user agents . Each
user agent is associated with a SIP address.
SIP User Agent Function
User agent client
Initiates SIP requests
User agent server Receives SIP requests
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SIP Call Flow
The call flow for SIP sessions depends upon whether the SIP session
is established directly between SIP user agents or whether a SIP
server (proxy, registrar, or redirect) is located between SIP user
agents.Figure shows the typical call flow between two user agents,
with each step noted in parentheses. First, user agent A sends out an
INVITE request to initiate a call. User agent B then replies with the
Trying response code (100), indicating that the call request is being
processed. User agent B then replies with the OK response code (200),
indicating that that user agent has accepted the call. User agent A
then replies to user agent B with an acknowledgement (ACK) request,
indicating that user agent A received the final response code from
user agent B. The real-time data is then encapsulated in RTP packets
and sent between user agent A and user agent B. Either user agent A
or user agent B can then send a BYE request, indicating the that the
user agent wants to terminate the session. User agent B then sends an
OK response code (200) to user agent A to indicate that the request
has succeeded.

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SIP Call Flow
User(Iris) Remote User
Invite (1)
100 trying (2)
200 OK (4)
Bye (6)

ACK (5)
183 Alert (3)
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SIP Call Flow
Iris gateway Proxy Server Remote User

Registration
OK (200)
Invite
Trying (100)
Trying (100)
Invite
Alert (183)
Alert (183)
OK (200)
OK (200)
Bye
Bye
Ack
Ack
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SIP Call Flow
Iris gateway Redirect Server Remote User

Registration
OK (200)
Invite
Moved (302)
Trying (100)
Invite
Alert (183)
OK (200)
Bye
Ack
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Critical parameters in VoIP
Echo Cancellation -Echo becomes a problem when
the round-trip delay is more than 50 milliseconds.
Packet Delay -IP networks cannot provide a
guarantee that packets will be delivered at all, much
less in order. Packets will be dropped under peak
loads and during periods of congestion.
Jitter Compensation -Jitter is the variation in inter-
packet arrival time as introduced by the variable
transmission delay over the network.
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Front Panel Features
LED Indicators -The VoIP card has five LED indicators for
indicating the status of the physical link (Ethernet), as shown in
the figure. LED1 & LED3 is in constant green color, when Ethernet
cable is connected. LED2 & LED3 blink, when receive or send
Ethernet packets; LED4 & LED5 always is in off position.
Ethernet Port - This port is utilized for connecting VoIP card
with the network & downloading new software in card,
DIP Switch - There are 8 DIP Switches provided on the front
panel of the MCC. The settings for the same are given as under:


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Specifications of VoIP Card:

Microprocessor
Voice Microprocessor
Memory Capacity
Program memory
System Memory
MPC 860 TZ
RL56CSMV/6

KERNELROM 512Mb
FLASH - 32 Mb
SDRAM 256 Mb




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Configuration detail of VoIP card
Iris needs the following things
Ethernet connection with static IP
Default Gateway IP
Gatekeeper or remote Gateway IP (for H.323
signaling protocol)
SIP proxy or remote gateway IP(for SIP signaling)
User name & password for registration (SIP
signaling)

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Configuration detail of VoIP card
Default Gateway is the place where the static
IP is originated from for the VoIP card.
Remote Gateway is the gateway where the
call is supposed to land.
Gatekeeper is a body,which has the
registration of various remote gateways.It must
be noted that if gatekeeper is configured,Iris
will route all the VoIP calls through only.
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Configuration detail of VoIP card
SIP Proxy has same functionality as
gatekeeper does with H.323.
Iris VoIP card has 12 ports.
Any number of VoIP card can be installed in
one system.
Iris VoIP card can maximum 256 users
registration with SIP signaling.
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Configuration detail of VoIP card
Iris VoIP card has one serial port for
configuring the data base.The baud rate of the
serial port is 19200.
Iris VoIP card supports SIP & H.323 signaling
with the same hardware ,only software is
difference.
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Audio CODECS supported by IRIS IVDX
G.711 (PCM) 64Kbps
G.728 (LD-CELP) 16 Kbps
G.729, G.729a (CS-ACELP) 8Kbps
G.723.1 (CS-ACELP) 5.3Kbps , 6.3Kbps

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Internet
IRIS IVDX
IRIS IVDX
IRIS Connected Through Internet
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Implementation Of VoIP Technology
IN IRIS IVDX
Gatekeeper
MCC
VOIP
FLC

IRIS IVDX
Internet
ISP
MODEM
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How to make call net meeting to Iris
VoIP & Iris VoIP to net meeting
Configure the Net Meeting in your PC
Now open the Net Meeting
Select the Tools & select the Options
Open the Options & select the Advance Calling
Select the Gateway setting & fill the Gateway IP of the
IRIS VoIP card
Now click ok & exit from tools
Now configure the IP of your PC in IRIS VoIP card
Now you can dial any number of IRIS subscriber from
Net meeting

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Questions