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De-mystifying Unified Communications

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PCQ Bureau
New Update

How many times has it happened that you've received a nagging call on your

cellphone by someone trying to sell you a credit card just when you're in the

middle of a crucial meeting, or maybe enjoying an afternoon nap on a holiday? In

another case, how many times have you wished that a person had called you

instead of seeking clarification about something via email? While you were

available on the phone, the sender didn't know and out of the fear of

disturbing you, just shot you a mail instead. Let's take the reverse case. How

many times did you try to reach a colleague for some urgent information, but hit

a stone wall? You first called on the office extension number, but the person

wasn't there on the seat. You then called on the person's mobile, which went

to voice mail. You then sent a sms, but didn't get a reply back. Finally, as a

last resort, you just sent an email asking the person to contact you when free.

Later, the person read your mail and tried to contact you, but faced similar

obstacles. Finally, after wasting so much time, effort, and money on phone

calls, you both just ended up discussing things over email. All these situations

are not a new phenomenon. 

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Concepts, building

blocks & implementations

Chances are that you face them everyday. In fact, we all face them everyday,

which is quite sad because despite having so many different modes of

communication, we're not able to reach the people we want to when we really

need to. On an average, every corporate employee today uses at least three to

four modes of communication-landline, mobile, email, and chat. And yet,

reaching people is becoming increasingly difficult. Not being able to reach a

person is not only frustrating, but also counter productive and expensive with

so many modes of communication. It can also lead to delays in key decisions

being taken, which can lead to other problems. What's needed is an integrated

solution that will tell you whether a person is available and the best way of

reaching him/her. That's essentially what unified communications proposes to

do.



A successful unified communication deployment is one where you're always
contacted through the mode you prefer, and by the people you want to be

contacted.

While all this may sound like a distant dream, it actually isn't. In fact,

the technologies have already arrived, and the products and solutions are also

available. It's just a matter of time before we all start taking it for

granted, like email and mobile phones. In this story, we'll tell you how close

you really are to unified communication, and even do some sample deployments.

Though not full-fledged, the deployments will give you an idea of the

possibilities. The rest is up to you to explore.

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Basic building blocks



A typical UC solution is a well-knit integration of various elements like
e-mail, chat, telephones (PBX or GSM), collaboration tools, audio and video-that

form the basic building blocks of such a system. Of course, you cannot forget

IP, which forms the backbone of this integrated system. This is because you need

a standard medium so that all these building blocks speak the same language and,

therefore, communicate with each other. As a result, PBX becomes IP PBX, audio

goes to become VoIP and enables audio conferencing over IP, e-mail takes up more

messaging and collaboration features, there is web conferencing and unified

messaging.

Note that while IMs provide almost all the functionalities from mails, voice

chat, PC-to-PC calling and connectivity over SMS-all this is currently

happening in the public domain. So all you need now is to implement these

functionalities at the corporate level. Now let us see the role of each of these

blocks separately to make an efficient UC system.

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IP PBX- An IP based PBX system allows an organization to converge

both voice and data networks, over a LAN or a WAN. It uses standard

packet-switch protocols to carry voice across a data network. There are

state-of-the-art IP-based solutions that allow you to integrate incoming as well

as outgoing voice applications with applications that run on the Net like

real-time chat, Web collaboration, and e-mail. As a result, you can have

multiple interactions through a single agent simultaneously, no matter which

communication channel is being used by the customer.

VoIP- As you're using the same network for voice and data

communication, you save on infrastructure costs. Plus, of course you can use

your existing data links between your various branch offices for voice as well,

thereby saving on STD bills. There are various kinds of soft and hard phones

that enable VoIP as a preferred means of communication.

Presence server- is an important element that sits in the UC network

and keeps a record of whether the receiver is present or not. It also records

the information about the mode of communication that is being used by the user,

such as whether he is available on e-mail, IM, fixed line, cellphone, etc. This

information is collated and used by other call/phone manager systems or

communicators that accordingly send the message to the receiver over the

communication device being used by the user at that time.

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E-mail-is one of the most preferred methods of written

communications in organizations. But over a period of time when it has been in

use, it is now time to see what more can you do with it. Today solutions such as

Exchange 2007 IBM Domino have enabled user to do much more than just sending or

receiving e-mail. You can do messaging and conferencing, call (telephone, fax)

management, document sharing and much more on these.

Unified messaging-This is not the same as unified communication as

it is very often misunderstood. Unified messaging is a subset of the larger

umbrella of solutions that unified communication provides. UM keeps your

communication limited to the extent of messaging and collaboration, and lets you

do voice mail, e-mail, text to speech, and view fax messages from one common

interface. For example, you can do all this with Outlook 2007. This is far less

than the capabilities of a UC system that takes you beyond UM to add more

features like collaboration/multimedia conferencing.

Audio conferencing-While this is the most commonly used mode of

conferencing even today, it has added a lot of features in it to become an

important element of UC. There are many solutions available in this space. Skype

and public IMs are in popular, there are many commercial solutions like those

from Polycom and Avaya.

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Web conferencing-This is also an important component of a UC

solution. There are both hosted solutions like Webex and that from Citrix, and

you can also have one set up on your own over the Internet. Since everything is

going over the Internet while making a UC system work, Web conferencing becomes

the hot favorite as it can let you do all forms of data, voice and video

conferencing.

Voice mail-This refers to sending a recorded message. Though this

has been in use since long time, the real use of voice mail



beyond telephone's answering machines is being made by the UC systems. This
happens when voice mail too shifts to IP, to help deliver voice mail on any

network (GSM or PSTN) you are in. With a good UC solution in place, you can call

into your mail server from either device to have your voice messages read out.

UC Client-Now this is pretty obvious. Unless there is one client

that brings all the above mentioned components together into one console, there

can be no unified communication.

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There are some such products/solutions already available in the market like

the one's powering MS Outlook and Office Communication Server 2007, that help

one element talk to the other and respond in the manner most suitable to the

user.

Integrating the blocks



To achieve the desired scenario described in the introduction, our typical UC
user connects to the UC system via one of different available means: over the IP

network (that can be his LAN or through the Internet), over a PSTN telephone

link or from a PDA or mobile phone using a GSM or CDMA network. Therefore, the

first point of integration is at the network level, where each of methods of

transmissions must be brought to a common IP layer that our UC system works

with. This will require an IPPBX gateway to bring the PBX (telephony) to an IP

platform. Additionally to support mobile networks you will need a device that

can link GSM/CDMA to IP. The next hand off can be done with software, where you

can setup 'preference registries', that contain information about how

different users prefer to handle their calls depending on their status. For

instance, if a user sets his status to 'In a meeting', all incoming calls

except from the user's boss can be sent as emails or be directed to a

colleague. After this comes all the different hardware and software that let you

actually do e-mail, chat, collaboration, VoIP and Video on IP. Even here, you

will easily find solutions that integrate mail, chat and collaboration. To add

voice and video conferencing, you may need to add one more step to the

checklist.

Vendors offer unified communications in the form of different 'editions'

or 'packages' that parcel different sets of components. For instance, one

edition of a UC solution can have IP telephony, messaging (email and chat) and

basic conferencing. A step up from there can include video conferencing and

voice-based access to messages. Solutions are also available that perform the

role of integrating all modes of communication but leave out their

implementation to other solutions---for instance, it can let users connect to it

to access their email, voice mail, faxes, phone calls and so on, but the actual

IPPBX part with call routing will need a dedicated call management solution.

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Cisco Demos

Unified Communications
The PCQuest Team visited Cisco at their Delhi

office and got to experience a demonstration of Unified Communications in

action.

The Cisco team demonstrated both IP phones as well as

softphones in action, with the use of a Presence Server. The Presence

Server was used to maintain and manage availability status of different

users who were logged on from the IP and softphone terminals. Then, calls

were made between all combinations of IP phones and softphones (viz. IP

phone to IP phone, softphone to softphone, IP phone to softphone and so

on). After that, we were also shown how the IP phone call will always

follow the user, regardless of which terminal he is logged on from and

even if he switches from hardware to software phones. We also saw the

Cisco Call Manager in action, which is used to create rules for routing

different kinds of calls and monitor them in action.

Protocols



Although TCP/IP is the basic Internet protocol there are a few more necessary
for an IP-based communication. Let's take a look.

SIP: Session Initiation Protocol is an ASCII-based standard specified

in IETF RFC 2543, used to initiate, modify and terminate interactive user

sessions involving multimedia. You can invite users to conferencing with

multicast sessions. Since this protocol supports mapping and redirection

services, you can communicate from any location. The requests from clients

identified by SIP URLs, can be sent through transport protocols such as UDP,

SCTP or TCP. SIP determines the end system, the communication media and partners

and the called party's desire to engage in communication. Once all this is

confirmed, the protocol fixes call parameters at either end with call transfer

and termination.

H.323: is a binary ITU standard which ensures compatibility in

videoconferencing, including call control and management, termination for both

point-to-point and multipoint conferences, and gateway administration of media

traffic, bandwidth and user participation. Both H.323 and SIP are 'intelligent

endpoint protocols,' that locates remote endpoints to establish media streams

between local and remote devices.

MGCP: Media Gateway Control Protocol describes the method of

communication between a media gateway, which converts data from the

circuit-switched network format to that required for a packet-switched network

and the media gateway controller. It keeps track of the number of carriers and

other IP telephony users.

Megaco or H.248: Listed by IETF as Megaco (RFC 3015) and by ITU-T as

Recommendation H.248, Megaco/ H.248 is a call-control protocol between a gateway

controller and a gateway. While MGCP makes it possible for the controller to

determine the location of each communication end-point and its media

capabilities, for choosing a type of service, Megaco/H.248 enhances this

functionality by supporting more ports per gateway as well as multiple gateways.

Other than H.323/SIP and H.248/MGCP, there are proprietary protocols from

various companies such as Skype that have been quite popular. 'Skinny' or

Skinny Client Control Protocol (SCCP) is a proprietary protocol used by Cisco

for communications between its Call Managers and VoIP phones.

Sreedhar

Sambukumar




(Ex-GM, Software Development)


Siemens Public Communication


Networks

In my opinion, Unified Communication has not taken off in a big way in India. One of the challenges of a UC solution is that it doesn't necessarily adapt very well to already invested infrastructure. Many a UC solution expects a green field kind of setup for optimal benefit. Eg, if you have a classical TDM based PBX, most solutions expect digital/VoIP based standards.In India, the merger of enterprise networks with PSTN is limited due to Government regulations.

RTP and RTCP: are two of the more common speech transmission

protocols. The Real-Time Transport Protocol or RTP defines the way for programs

to manage real-time transmission of multimedia data over multicast network

services. It is designed to support video conferences with multiple,

geographically dispersed participants. RTP combines data transport with a

control protocol (RTCP), making it possible to monitor data delivery for large

multicast networks. This allows the receiver to detect if there is any packet

loss and to compensate for any delay jitter. Both the SIP and H.323 use RTP. To

ensure security, RTP was improved to Secure RTP (SRTP), defined in RFC 3711.

SRTP provides for encryption, authentication, and integrity of the audio and

video packets transmitted between devices.

H.264 video codec : This video codec is used as a low bit rate compressed

encoding solution for moving pictures in video conferencing. It can be used with

both H.323 and SIP based solutions. It provides an umbrella for low bandwidth

conferencing over ISDN and IP. H.264 enables videoconferences to connect at half

the bandwidth and retain the same quality. This codec supports 60 fields per

second and an advanced method of reducing the pixelation when there are a lot of

motion or scene changes. H.264 eliminates the tiling seen on videoconferences,

whenever there is a 'scene' change or a lot of motion.

Quality of Service



On the Internet, you need the desired bandwidth for your call else you face
problems such as dropped calls, jitter, latency and poor quality reception. QoS

policies ensure that transmission rates, error rates, and other characteristics

can be measured and improved upon to avoid such irksome issues. They guarantee a

certain level of performance depending on application request. These guarantees

are important, especially in case of multimedia streaming like video

conferencing, which is delay sensitive. QoS is controlled at the router level,

so you need a router that lets you configure QoS for your WAN. Protocols such as

Resource Reservation Protocol (RSVP) provide differing level of service for

packets passing through a gateway host based on policy and reservation criteria.

Data, audio and video all have different requirements on timeliness and quality.

Using ATM, which lets you pre-select a level of quality in terms of service, QoS

can be measured and guaranteed in terms of the average delay at a gateway, the

variation in delay, and the transmission error rate.

Exchange 2007

Exchange 2007's predecessor, Exchange 2003, even with the separate but

related Live Communications Server 2005 added still does not qualify to be

called a true 'Unified Communications' platform. The reason is, even that

combination does not handle the part of telephony. That support now exists in

Exchange 2007. We are going to setup Exchange 2007 and look at the the options

it has to offer towards 'unified communications'.



Exchange does not implement a very high level of call and presence management,
but provides just the basic elements for it. For instance, you can create as

many dial-plans (rules for what number patterns a user can dial), but you can

associate only one of them at a time with a particular user.

Once UM role is

configured, users can use Outlook (mail client, Web Access or Voice

Access) to connect to Exchange and access mail, voice mail and fax

messages

Getting started



Before you go ahead to install Exchange 2007, you need to have quite a few
things handy. For instance, it will work only on a 64-bit system, with the

64-bit edition of Windows 2003. This also means you need 64-bit drivers for your

system for optimal performance. Also make sure the server has at least 1 GB of

RAM. For VoIP gateways, you should have something like Asterisk (described

elsewhere in this story) or Cisco Call Manager installed that tracks and manages

VoIP components on your network. PBX networks also be managed by VoIP management

software or have their own management software --- like ones from Avaya, Siemens

and NEC --- and these can work with Exchange 2007 as well.

We installed a fresh system in our Labs and even after installing Windows

Server 2003 R2, quite a few updates were needed before the Exchange installation

would start. This includes adding IIS, .NET Framework 2.0 x64 with a hot-fix,

installing the Windows PowerShell (for running the new Exchange Management

Shell), updates to TCPIP.SYS and MSDAPS.DLL, MS Core XML Services 6.0 and

Windows Media Encoder 10 x64 with its Voice Codec. The Media Encoder component

is required for Exchange 2007's TTS engine that's used by the Unified

Messaging Role. You need to also raise the forest and domain functionality level

to Native (2003) mode.

The installer will automatically check for these pre-requisites and warn you

about needing to install them before you can go ahead. The list can change

depending on what roles you select for installation. The default installation

works in a 30-day trial mode until you enter the product key (can be done only

post installation), that will promote it to either Standard or Enterprise

edition of the product.

Some UC Solutions
Cisco Unity: Call manager, presence

management, voice, video and web conferencing, designer to create

convergence applications.

Microsoft Exchange 2007: Messaging and

conferencing, call management, Task lists, Document sharing.

NEC Voice Mail and Unified Messaging: Messaging,

voice mail, call management (for hospitality industry).

Siemens HiPath Xpressions: Messaging, call

management,



message playback.

The UM Role



You need to select the 'Unified Messaging' role from the installation screen
to get unified messaging. Once installed, you must create Dial plans, UM mailbox

policies, IP gateways and Auto attendants. Dial plans let you configure what

kinds of numbers a user can call (for example: local extensions, outside

numbers, international numbers and so on). Mailbox policies (for UC) let you

setup dialing restrictions on where users can call-within the dial plan, to

local extensions-the kind of national and international number patterns they

can dial. UM IP gateways route communications between our Exchange box and an IP

PBX. Auto attendants will automatically pick up calls that come to specific

(configured) extensions, process them and maybe forward them to a different line

or mailbox.

Client access



Users can call in to the Exchange 2007 server using Outlook 2003 or 2007.
Messages from their voice and fax accounts will be organized into separate (read

only) folders labeled 'Voice Mail' and 'Fax' respectively, within the

'Search Folders' node in the folder tree. These are also accessible from the

web front-end. The web front end is slightly revamped from the 2003 with a more

detailed 'Options' screen that lets users manage their mobile devices (for

use with ActiveSync). Users can also view the calendars of other users with

free/busy schedules when they search for them from the global address book.

UC with Open Source

A nice looking and easy-to-use interface, seamless integration, lots of

features and good documentation in an enterprise-class solution-this is all

that we promise through this writeup about a collaboration and Telephony

servers, WildFire Jabber and Asterisk. These two work hand in hand by using a

plug-in called Asterisk-IM and give you a truly integrated Unified

Communications platform.

Let us see what these solutions are. WildFire Jabber is a modified version of

Jabber Messaging and Collaboration Server, which can run on any platform

including Linux, Mac and Windows. You can manage it from an easy-to-use Web

interface. Asterisk, on the other hand, is the most renowned open source

IP-Telephony Solution available. Not only many companies use it, there are also

some vendors who manufacture IP-PBX boxes with Asterisk embedded inside.

Asterisk-IM, which we will use, is a Java based plug-in for WildFire that adds

integration functionality for Asterisk into it.

When adding credentials, make sure to use the right port and Context for Asterix Server. Values for a default installation are as above

But Asterisk has one drawback--natively it has only a CLI based management

and configuration interface. We solve this problem by using FreePBX, a graphical

interface for Asterisk that makes management and configuration a breeze.

To make things simpler, we also have a distribution called TrixBox Linux.

This CentOS based distro has all the required components for deploying a

production class IP-telephony solution. It integrates Asterisk, FreePBX,

SugarCRM, Maine, A4astersik (an billing and monitoring package for asterisk) in

one CD. All you, then, have to do is to install this distro and your full

fledged IP-PBX should be up and running in minutes.

The list of features in this Asterisk and WildFire combination easily

competes with any commercial UC solution, and it will be difficult for us to

list all of them in this story. So, we limit the scope of this article to

integrating WildFire and Asterisk Servers to exchange information and user

presence.

Pre-requisites



To start with, you will need TrixBox Linux running on a machine which has a
valid IP address (say 192.168.1.1), WildFire Jabber Server running on a Windows

Machine (say, with IP 192.168.1.2), at least two clients running any OS that are

either connected through a Hard IP Phone or with any Soft Phone installed.

Additionally, you will need Spark, a WildFire client that supports Asterisk-IM

plug-in, installed on both the clients. You will also require the Asterisk-IM

plug-in.

We assume that your WildFire and TrixBox setup is running individually and

you are able to place calls and chat between both client machines. If you face

problems in configuring TrixBox, visit http://dumbme.voipeye.com.au/trixbox/trixbox_without_tears.pdf

for help. Installing and configuring WildFire is simple too. You can get all the

above-mentioned applications on this month's PCQ Xtreme DVD. The Asterisk-IM

plug-in available on the website of WildFire Jabber doesn't work with the

latest WildFire version. So, while configuring use only the Astersik-IM.jar file

provided on this month's PCQ Xtreme DVD.

Once done, your Spark client will automatically change the status of the user to 'On Phone', whenever the user dials out or receives a call

Configuration



The integration of WildFire and Asterisk/Trixbox is very straightforward. First
open up the folder 'C:\Program Files\ Wildfire\plugins' or wherever you have

installed WildFire in your machine and copy the Asterisk-IM.jar file from this

month's DVD to the folder. Now shutdown the WildFire Server and restart it by

clicking on the WildFire Server Agent. Now you will notice two new things. One,

a folder called Asterisk-IM is created on the plug-in folder and two, there is a

new Tab available on WildFire's management interface. Now, go to the

management interface and provide credential for the Asterisk Server. For the

values of the required fields for a default Asterisk installation, please refer

to the following screenshot. The only thing that might vary is the IP address of

the Asterisk server.

Adeesh Sharma, Anil Chopra, Anindya Roy, Rinku Tyagi and Sujay V Sarma

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