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Since
its introduction in 1995 VoIP technology has come a
long way. Enterprises in India and the rest of the world
have successfully used the technology and derived cost
savings and other benefits from it. A look at the technology
in its current evolved state can help you plan ways
to use it optimally in your network. by Soutiman
Das Gupta
 VoIP
was introduced way back in 1995 and since then has come
a long way. It has now been adopted universally as a
reliable technology alternative by which voice is transmitted
over a data network instead of a traditional voice network.
Companies who have successfully deployed the technology
have realized numerous benefits.
The biggest benefit has been savings in communication
cost. By using IP Phones to make long distance calls
between branch offices and zonal offices, enterprises
effectively spend a fraction of the amount it would
spend if it went the traditional telecom network way.
The next big benefit has been optimum utilization of
network resources. Many enterprises have built robust
networks with updated routers, switches, and other devices,
and put high bandwidth links in place. In many cases,
a lot of bandwidth and other resources were available
for use even after critical enterprise data traffic
was transmitted. Transmitting voice through this unused
facility makes better use of the network resources.
Since
our government relaxed certain restrictions on the use
of VoIP from April 2002, many Indian enterprises have
successfully leveraged the technology to their benefit.
Even before this date, many enterprises implemented
VoIP in a closed user group, as was permitted earlier.
The effective use of QoS (Quality of Service) in the
network devices has helped overcome limitations like
delay, echo, and jitter. Here's a look at the technology,
the protocols, and the QoS factors.
| Understanding
QoS |
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Generally
there are three techniques that can be used (separately
or in combination) to improve network QoS.
Controlling the networking environment:
You have to provide a controlled networking environment
in which the capacity can be pre-planned and adequate
performance can be assumed.
Using management tools: You can use management
tools to configure the network nodes, monitor
performance, and manage capacity. Traffic can
be prioritized by location, by protocol, or by
application type. This allows real-time traffic
to be given precedence over non-critical traffic.
Adding control protocols and mechanisms:
You can add control protocols and mechanisms that
help avoid or alleviate the problems inherent
in IP networks. Protocols like RTP (Real Time
Protocol) and RSVP (Resources Reservation Protocol)
can provide greater assurances of controlled QoS
within the network. And mechanisms like admission
controls and traffic shaping may also be used
to avoid overloading a network.
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| Network
warm-up exercises |
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While
the higher-ups in the management are yet to pass
your VoIP procurement budget, you can do a few
things to prepare your existing network infrastructure
for the inclusion of new technology. The priority
is to make your LAN as efficient as possible.
This can be accomplished in several ways.
The most important way is to get rid of unnecessary
protocols. NetWare supports IP now, Windows 2000
no longer requires NetBIOS for file and print
traffic, and most mainframes now support IP. Although
it may require considerable effort to remove IPX,
SNA, NetBIOS over TCP, and other protocols it
will result in more available bandwidth on the
network, more available memory, and faster response
time in the clients and servers.
Once you have removed all these protocols, you
may find that the level of broadcasts on your
network doesn't justify such small subnets anymore.
Moving to larger, 'flat' networks will allow you
to remove routers that are no longer needed. These
routers are not only points of failure, but are
frequent bottlenecks, and usually a major source
of delay. Now your VoIP traffic will be more reliable
and of high quality.
QoS is a major requirement for VoIP. So why wait
until the last minute to QoS-enable your network?
Since the configuration of QoS and VoIP are both
relatively complex tasks, you don't want to be
doing them at the same time if you don't have
to. By working the bugs out of your QoS configuration
now, your VoIP project will be less likely to
encounter glitches.
And last, it's never too early to begin planning.
Among the things you can do before you even pick
a vendor
are:
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Identify the types of traffic on your network
and prioritize them. Voice may actually not
be the most important.
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Determine existing call-traffic statistics and
predict future statistics, including cost, average
simultaneous calls, average duration, and source/destination
pairs.
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Prepare your network management system for VoIP,
including upgrading your RMON probe and protocol
analyzers to recognize and decode VoIP.
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Packet
switching
A traditional voice call through a telecom network opens
a dedicated channel of around 64 Kbps in each direction
between the two parties. The bandwidth used in the transmission
is around 128 Kbps and the switch ports at all the carriers
switches are blocked for the entire duration of the
call.
Data networks use packet switching by which a connection
is open long enough to send a small packet of data from
one system to the other. VoIP converts voice into data
packets which are dynamically routed on different paths
over the network depending on availability and reconverted
to voice at the other end.
Standards soup
Like any other technology VoIP is defined by standards.
There are various standards out in the VoIP arena like
H.323, SIP (Session Initiation Protocol), and MGCP (Media
Gateway Control Protocol). Of this H.323 is the most
widely used protocol.
Widely-used H.323
H.323 is a suite of protocols developed for specific
applications like synchronization, control, and compression
codecs. A codec, which stands for coder-decoder, converts
an audio signal into a compressed digital form for transmission
and back into an uncompressed audio signal for replay.
H.323 allows customer's products to interoperate with
other H.323-compliant products and provides standards
for interoperability between LANs and other networks.
Network managers can restrict the amount of network
bandwidth available for conferencing.
A little SIP
SIP (Session Initiation Protocol) is used to establish
real-time calls and conferences over IP networks. It
is independent of the packet layer, an open standard,
and scalable. It has been designed to be a general-purpose
protocol. However, extensions to SIP are needed to make
the protocol truly functional in terms of interoperability.
Master/slave MGCP
MGCP (Media Gateway Control Protocol) is a master/slave
protocol that provides a tight coupling between the
MG (Media Gateway) which is the endpoint and the MGC
(Media Gateway Controller) server. MGCP-based VoIP solutions
separate call control (signaling) intelligence and media
handling. In MGCP architecture, the MGC server or 'call
agent' is mandatory and manages the calls and conferences
and supports the services provided.
Benefit of a converged
network
You can create a simple and scalable VoIP architecture
with the inclusion of hardware and software elements
like VoIP routers, VoIP gateways, switches, and IP phones.
In cases where a router or switch does not support VoIP,
all one needs to do is install a small chip or just
upgrade the software to VoIP-enable these devices. Most
vendors offer these upgrades for their equipment.
In case of converged networks, the initial investment
may turn out to be slightly higher, but one can observe
the ROI on this technology within a very short time
span.
VoIPs core benefit is its ability to make next generation
converged networks a reality. In a converged network
environment, telephony and data signals are transmitted
as packets over the data network. A typical office has
a separate network for data transmission and voice (telephone).
Now, a converged network enables you to transmit voice
over the existing data network. This maximizes the efficiency
of your network. The traditional voice circuits can
be used as backup or even eliminated.
It also simplifies your network architecture. A single
infrastructure is capable of carrying both data and
telephony traffic. You don't need to pull separate cables
for services. Network deployments and reconfigurations
are simplified, and service can be extended to remote
sites and home offices over cost-effective IP links.
QoS is the key
The idea of carrying voice traffic over the data network
is great. But there are a few issues that need to be
considered. Voice is converted to regular data packets
and moves in the network just like other data packets.
If there are transmission errors the packets need to
be re-transmitted, and in case of bottlenecks the traffic
is temporarily blocked. The voice packets are thus delayed
and conversation is hindered. There are also problems
like echo and jitter. Here's where QoS comes in and
plays a key role to ensure acceptable call quality.
IP Phones and current generation routers usually have
QoS support built into them. A QoS bit is attached to
all voice packets as it hits the network. The network
devices recognize this extra bit and offer it priority
over regular data packets. This way, voice moves first
over the network and thus does not cause unnecessary
delay. In case of transmission errors which require
re-transmission, there is no drop in call quality. Certain
audio codecs can compress the header of VoIP packets
to reduce its size.
Migration
issues
Migrating from an Ethernet LAN gives rise to a few delay
issues. Ethernet frames are variable in length, and
Ethernet has no mechanism for prioritizing one frame
over another. Therefore, as network traffic increases,
small frames carrying a voice payload may often have
to wait in switch buffer queues behind large frames
carrying data. With voice having small delay tolerance,
the lack of prioritization across a switched Ethernet
network may degrade the quality of voice communications.
The most promising solution is to handle the problem
at layer 3 via the RSVP (Resource ReSerVation Protocol).
RSVP operates by reserving bandwidth and router/switch
buffer space for certain high priority IP packets like
those carrying voice traffic. RSVP is still only able
to set up paths for high priority traffic on a 'best
effort' basis, and thus it cannot guarantee the delay
characteristics of the network.
Fast Ethernet and Gigabit Ethernet presents a clearer
migration path than ATM. Migrating from an ATM network
to a converged VoIP infrastructure may be principally
simple because ATM was designed specifically to support
both voice and data traffic over a common infrastructure.
It also provides multiple QoS levels.
Future
VoIP
A future VoIP network will include iPBXs (IP-based PBXs),
which will emulate the functions of a traditional PBX.
These will allow both standard telephones and multimedia
PCs to connect to either the PSTN or the Internet, providing
a seamless migration path to VoIP. An iPBX can also
combine the features of today's switches and routers
and could become the gateway for variety of value-added
services like directories, message stores, firewalls
and other network-based servers.
Such a VoIP system would also combine real-time and
non real-time communications.
Soutiman Das Gupta can be reached
at soutimand@networkmagazineindia.com
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