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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.323The circuit switch based protocol

SIP –Session Initiation ProtocolSIP 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 non–H.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)InviteMoved (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

KERNEL–ROM– 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

MCCVOIPFLC

IRIS IVDX

InternetISP

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