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| What
is SIP? |
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What is SIP?
SIP, or Session Initiation Protocol,
is designed primarily to set-up, modify, and tear
down interactive communication sessions. SIP is agnostic
- it can support any type of communication session
whether it is voice, video, or instant messaging.
Although SIP seems "new," it is actually
based on many protocols widely used today across the
Internet and in many enterprise applications. If you
use web browsers, then you already depend on a protocol
very similar to
SIP, called HTTP (Hyper-Text Transport Protocol).
SIP is modeled after HTTP, and in fact uses much of
its syntax and semantics. Both are text-encoded protocols
that help promote interoperability and integration
within an internet-centric architecture. In effect,
SIP is to converged communications what HTTP is to
information exchange for the World
Wide Web (WWW) - it makes the communications infrastructure
transparent to end users and enables ready access
to many modes of communication.
SIP uses URIs (Uniform Resource Indicators) for user
addressing, in the
same basic form as e-mail addresses: user@domain.
A user's SIP address
can in turn be mapped into one or more contacts, each
of which can
represent a communication device or service at which
the user may be reached. For SIP communications this
can apply to any communication device, some examples
of which are:
• A phone: sip:408-555-1212@company.com;user=phone
• A fax: sip:408-555-1214@company.com;user=fax
• An IM user: sip:johndoe@company.com
These contact addresses can be numerical or internet-based,
and therefore bridge the telephone network and the
Internet seamlessly. With SIP, users can potentially
reach any telephone number or Internet address from
their existing devices (Multi-modal Communications).
They don't have to get a new gizmo for it to work.
Through SIMPLE (SIP based Instant Messaging Presence
Leveraging
and Extensions), SIP provides key functions for presence
and instant messaging. Presence lets users give a
visual indication on the devices of peers of their
status, their availability, and how they can be contacted
- before a communication session even begins. When
integrated with telephony, a user now has a powerful
efficiency enhancing feature that provides control
over how others reach them by exploiting the capabilities
of multiple devices such as IP Phones, cell phones,
softphones, pagers,
and wireless or Bluetooth devices.
Presence is not limited to a single user; it can also
apply to a group of
users (i.e. "Finance Group") or a device
(i.e. "Phone Status = Off-Hook"). Presence
information can also be leveraged across any form
of communications. Presence information can be accessed
by both users and applications, providing the opportunity
to create next-generation
converged communication applications that deliver
new capabilities such
as "polite calling," and intelligent work
agent applications that automate
user schedule and communication tasks based on customized
policy
rules.
Take Advantage of SIP
So how does a company make the move
to SIP? With Avaya there is a
logical migration path that enables you to take advantage
of SIP at your
own pace, while fully leveraging your existing communication
assets.
Step One begins with Avaya Communication Manager Release
3.0. For Avaya customers this may involve a simple
upgrade to their system, after which they can begin
their migration with the Avaya Converged Communications
Server (CCS). CCS SIP Enablement Services creates
a communication services layer that mediates between
Avaya MultiVantage Communications Applications and
a wide range of standards-based user agents, web-based
applications, and communication devices. These
services combine the standard functions of a SIP proxy/registrar
server
with SIP trunk support and available duplicated server
features to create a highly scalable, highly reliable
SIP communications network that supports telephony,
instant messaging, conferencing, and collaboration
solutions.
Once the Converged Communications Server is in place,
the enterprise
can begin leveraging the potential cost-savings and
efficiencies of SIP trunks. In addition, a gradual
step-by-step migration path is now in place, allowing
the enterprise to maximize the benefits of SIP while
fully
preserving compatibility and investment protection
with their existing
H.323, digital and analog endpoints and infrastructure.
Step Two introduces secure SIP-based enterprise Instant
Messaging
(IM) and user presence through CCS. These new capabilities
are
integrated with IP telephony, using the Avaya IP Softphone
for business users and Avaya IP Agent for call center
agents. These applications
support SIP for Instant Messaging and Presence, but
H.323 for IP
Telephony, allowing enterprises to extend the benefits
of SIP to all users without the need to make extensive
changes to their existing voice infrastructure.
Step Three in the migration process is the rollout
of new SIP-based
services such as audio and web conferencing through
Avaya Meeting Exchange. These new applications become
feature servers off of the SIP network, accessible
to both SIP endpoints as well as existing devices
that are able to use the new capabilities.
Taking Step Four towards pure SIP Telephony is easy
for enterprises
that have already deployed H.323 IP Telephony. The
migration process
can be initiated at any time and transitioned at whatever
pace is desired.
Once registered and licensed on CCS, existing Avaya
4602SW, 4610SW, 4620SW, and 4621SW IP phones can convert
their operation from H.323
to SIP through a simple and free firmware upgrade.
IP Softphone users
have a similar migration path to SIP telephony through
the Avaya SIP Softphone, which supports SIP signaling
for both IM and telephony communications.
The benefits of SIP can leveraged even with a small
initial deployment.
For example, through Avaya Handle-Based Dialing, a
plug-in to CCS, SIP telephony users can type in a
name (i.e. john doe) or handle-address (i.e. johndoe@company.com)
to reach any other user on the network, even
those on analog, H.323, or digital phones. At any
point in the migration process existing endpoints
can continue to be used, as SIP supports both numerical
(telephony) and alphanumeric addressing providing
a critical bridge for communications between PSTN
and Internet networks. This
allows users on either network to reach any other
user without giving up existing devices or the advantages
of each. |
| Phone
- Tech Tips |
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How to Add Feature
Buttons to a Phone.
You can decide which features you want to assign
to each button on a phone. The process is slightly
different if you are adding buttons to a new phone
(using the Add User wizard) or to an existing phone
(using GEDI).
Adding Feature Buttons to a New Phone
To add feature buttons to a new phone, start the
Add User wizard. Then complete the following steps.
1- In the Add User Wizard
Station Information screen, click Buttons.
Avaya Site Administration displays a picture of the
phone, based on the model number you entered in the
Set Type field of the wizard.
2 - On the picture of the phone, click the button
you want to
assign a feature to.
3 - Click the down arrow and select the feature you
want to assign
to that button.
4 - Complete any feature-related fields that appear
on the Button
Properties screen.
5 - Repeat Steps 2-4 until you are finished assigning
buttons.
6 - Click OK.
Then complete the remainder of the wizard as you
would normally.
If you need help with a wizard screen, click Help
on that screen. If you schedule the job, remember
that open Terminal Emulation windows can
delay the job or cause it to fail.
Adding Feature Buttons to an Existing Phone
Once you have added a phone,
you must use GEDI to make any
additions or changes to the feature buttons.
1 - Open a GEDI window for the voice
system you want to administer.
2 - Type change station nnnn and press the ENTER key,
where nnnn
is the extension for the phone you want to modify.
3 - Click the tabs on the screen until you locate
the Feature Button Assignment
fields.
4 - Right-click the field that corresponds to the
button you want to
assign the feature
to.
5 - Select the feature you want to assign.
6 - Complete any feature-related fields that appear
next to the button field.
7 - Repeat Steps 4-6 until you are finished assigning
buttons.
8 - When you are finished, do one of the following:
- To submit the change immediately,
click Enter (F3) on the GEDI
window.
- To schedule the change, click Schedule
(F9) on the GEDI
window.
If you schedule the job, remember
that open Terminal Emulation
windows can delay the job or cause it to fail.
|
| Magix
Phone Extinction |
|
Avaya recently announced
that as of March 31, 2006, they will no
longer be producing or distributing the Merlin Magix
phone system
and instead shift their focus to the IP Office.To help
customers with
Magix systems go through this end-of-product-life cycle
Nicom Technologies stands ready to help customers in
a variety of ways.
Magix
to IP Office Migration
The IP Office is the next
generation, VOIP-driven, communications
platform designed by Avaya to replace the Magix system.
With advanced capabilities
like a built-in conferencing bridge, remote
office capabilities, remote access to e-mail and voicemail,
one number reachability and even call center functionality
that allows your employees
to better handle incoming customer calls.
Click
here for more information on the IP Office and our
migration services.
Merlin
Magix Maintenance Optimization
With the Magix system being discontinued
by Avaya it will be crucial to maximize your maintenance
contracts to ensure that your system is fully covered
and backed by Avaya for as long it will be supported.
Click
here to learn how Nicom can help you review, extend
and maximize your Magix phone system maintenance plan.
Refurbished
Magix Equipment
After Avaya ends production of the Magix
system Nicom Technologies
will continue providing high quality refurbished products,
parts and components to help support your business
organization.
Click
here to learn more about Nicom's Merlin Magix products |
| Avaya
Mobility with Nokia |
| |
Avaya in a joint partnership with Nokia has announced
the global availability of the first phase of its
enterprise fixed mobile convergence (FMC) applications.
This extends the reach of enterprise IP telephony
by integrating mobile communications with the reliability
and features of enterprise telephony services.
These downloadable applications transform Nokia Series
60 mobile devices into virtual desktop phones by enabling
mobile workers to access the features and functions
of their Avaya Communication Manager office desk phone
through an easy-to-use onscreen interface. This is
the first tangible result of a strategic collaboration
between Avaya and Nokia to deliver enterprise FMC
solutions, announced earlier this year, and delivers
on
their promise to enable mobile workers to productively
manage their voice communications with clients and
colleagues while reducing enterprise telephony costs.
The Series 60 Platform is the world's most widely
adopted smartphone platform, with millions of devices
already in use.
To Learn More About Avaya
Mobility, click
here. To learn more About
the Avaya/Nokia Series 60 mobility, click
here.
If you are interested in learning more about what
a mobile workforce can
do for your organization click
here to get a free PDF copy of Mobile
Workforce for Dummies, a helpful book that
explains the basics behind Avaya's mobile solutions
and how they can help your organization.
|
| Go
Wireless with Spectralink |
|
Spectralink’s
sophisticated radio technology to provide the best
voice quality consistently throughout any size and
type of facility.
The Link WTS is designed for minimal training, maintenance,
and administration. The wireless phones, weighing
only six ounces are
extremely simple to use and are durable enough to
withstand the rigors
of workplace usage.
The Link WTS uses a micro-cellular
design consisting of three
components: a Master Control Unit, Base Stations,
and Wireless Telephones. The MCU interfaces directly
with the PBX, Centrex, or Key.Hybrid system through
digital or analog extensions to provide the calling
features and functionality of the host telephone system.
The
Base Stations, which are linked to the MCU, are small
radio transceivers located throughout the facility
that relay calls between the wireless telephones and
system. Calls are handed off from one base station
to another as users move throughout the facility. |
Change Your System's Clock |
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Need help changing your Avaya phone system's internal
clock? Click on the link below that corresponds to
your Avaya system for more
information.
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