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QoS The Key To Accelerate Deployment Of IP Telephony?
for voice transmission over IP networks is imperative for
an experience equivalent to that offered by PSTN
prophecy of Internet Protocol (IP) becoming the universal
carrier for all communication traffic is facing rough weather
despite its promising potential. It is not just about picking
up the phone and making a call. The concept of a voice carrier
across locations having no constrains of geographies whatsoever,
is the prediction of the new generation of communication.
When the technology behind IP Telephony matured, the immediate
challenge was that of the infrastructure. It is now evident
is that critical data transmission across this medium will
be a larger concern? Migrating onto the IPSTN bandwagon
for voice transmission over the existing data infrastructure
is a Herculean task in itself.
The mission then for IP telephony is to ensure delivery
of the same quality of reliable voice conversations that
users have come to expect after a century of experience
over the primary public switched telephone network (PSTN)
lines. A greater effort is for organizations to view IP
telephony as delivering voice service over an IP network
such that users have virtually the same experience as when
voice services were transported over a circuit-switched
As end users demand more quality and reliability, providers
have shown more concern towards Internet protocol (IP) testing
for end-to-end quality of service (QoS). QoS mechanisms
provide a set of tools that can be used by the network administrator
to manage network resources in a controlled and efficient
manner. These will have the effect of offering improved
service to mission critical applications and users, while
simultaneously stemming the rate at which capacity must
In other words, QoS can help improve service to the network
users while reducing cost of providing these services. To
support IP telephony, a network must guarantee certain QoS
metrics, such as an end-to-end delay of roughly 150ms or
less and a delay variance (jitter) of no more than 30ms.
To ensure that these metrics are met, network operators
must invoke the most reliable QoS mechanisms, which will
typically include ways to prioritize and queue traffic or
reserve network resources end-to-end.
The term "IP telephony" covers a range of technologies,
including Voice over IP (VoIP) and Fax over IP services,
which are carried over both the Internet and private IP-based
networks. IP telephony is part of packet voice that includes
Voice over Asynchronous Transmission Mode (VoATM) and Frame
Relay networks, which run faster than IP but are less common.
IP telephony connects across combinations of PCs, Web-based
telephones and phones connected via public telephone lines
to remote voice gateways. Because information travels in
discrete packets, it need not rely on a continuously available
switched circuit. Consequently, it's easier on bandwidth
and is also cost efficient.
IP telephony is still a "best effort" system challenging
the "guaranteed" quality of service provided by
the Public Switched Telephone Network (PSTN). While the
Internet best-effort service model continues to treat all
network traffic in exactly the same way, there is no consistent
service outcome from the same. When the load level is low,
the network delivers a high-quality service. The best-effort
Internet does not deny entry to traffic, so, as the load
levels increase, the network congestion levels increase,
and service quality levels decline uniformly. This decline
in service is experienced by all traffic passing through
a congestion point, and is not limited to the most recently
admitted traffic flows.
QoS is the key to accelerating the deployment of IP telephony.
Although IP telephony is an appealing technology, QoS had
been concern enough to limit adoption of this technology.
The two most important components of QoS, latency and lost
packets, relate closely to each other.
Every VoIP application involves converting speech into series
of packets, each containing about 30 of voice. After the
speaker-side gateway receives the voice transmission, it
converts it to packets and shuttles these packets onto the
network. Packets traverse the network and reunite at the
receiver's end. The network may lose some packets, and others
arrive too late to use in the reconstructed speech. In either
case, the speech plays back without these packets lost in
Processing and transmission delay all packets, and this
delay causes latency in conversations. The transmission
leg across the network is the longest, especially on the
Internet. Users expect latency of 250ms or less, equivalent
to the delay of a satellite link for international calls
for "toll-quality" service. Unfortunately, the
Internet induces latencies that can far exceed 500ms.
Jitter buffers also contribute to latency. Jitter, or delay
variation, is the result of packets arriving at their destination
at irregular intervals. Bursts of Internet traffic create
jitter problems. This distortion is particularly damaging
to the QoS of VoIP applications. Severe jitter in IP voice
transmissions causes jittery or shaky voice quality. We
can also understand jitters as the speed variation between
quickly and slowly traveling packets. The jitter buffer
stores packets, allowing most of the slower packets to catch
up. The less control in routing, the more jitter that results,
and more jitter means a longer jitter buffer. But a longer
jitter buffer introduces more latency. Too short a jitter
buffer loses too many packets, causing voice quality to
When the network loses a packet, VoIP products "reconstruct"
it. The products cannot determine the information in the
packet, but like CD players smoothing over scratches, VoIP
algorithms produce transitions that are less distracting
than silence. Yet, too many lost packets degrade voice quality
to unacceptable levels. The maximum level of lost packets
for toll-quality service is difficult to define, but 10%
Latency, loss of packets and distortion are critical 'cannot
haves' in an IP network. The industry has evolved from building
interfaces between legacy systems and 'New World' technologies
to one with solutions and products that directly address
communication over a leased line and between closed user
groups. These solutions would translate to routers, soft
switches and IP enabled equipment populating the network
between the source of data transmission and the destination.
What is essential is a maximum throughput despite transportation
losses especially over large networks like the Internet
itself. The throughput rate decreases with increase of delay
and packet loss.
A Mission Critical Criteria
The objective of various Internet QoS efforts is to enhance
this service with a number of selectable service responses.
These service responses may be different from the best-effort
service by some form of superior service response, such
as lower delay, lower jitter, or greater
bandwidth. QoS service responses may be distinguished
by providing a consistent, and therefore predictable,
service response that is unaffected by network congestion
These are quantitative service responses, where the characteristics
of the service can be measured against a constant outcome.
A quantitative service many be one that constrains jitter
to a maximum level, or one that makes a certain bandwidth
available, within parameters of bounded jitter, similar
to a conventional leased line. Such constant-rate services
may be superior to best-effort services when the network
is under load, but they may also offer inferior service
when the network is under negligible load. The essential
attribute of these services is one of consistency.
Why is there a need for relative or consistent service profiles
within the Internet? First is the desire to provide high-quality
support for IP voice and video services, second, the desire
to manage the service response provided to low-speed access
devices, such as Internet mobile wireless devices, and third
is the desire to provide a differentiated Internet access
service, providing a network client with a range of service-quality
levels at a range of prices.
The initial approach to QoS was that of the Integrated Services
architecture where the request is sent to the network as
a reservation. The network in turn would respond with its
ability to carry the additional load by committing to the
reservation. Termination would mean another request or a
signal from the network corresponding its inability to continue
the reservation. The essential feature of this model is
the "all-or-nothing" nature of the service model.
This approach imposes per-application state within the network,
and for large-scale networks, such as the global Internet
itself, this approach alone does not appear to be viable.
The subsequent approach was to examine those mechanisms
that can provide differentiated service outcomes with appropriate
scaling properties. The differentiated services architecture
includes dropping the concept of a per application path
state across the network using instead the concept of aggregated
service mechanisms. Within the aggregated service model,
the network provides a smaller number of different service
classes and aggregates similar service demands from a set
of applications into a single service class.
Aggregated services are typically seen as an entry filter,
where on entry to the network, each packet is classified
into a particular service profile. This classification is
carried within the IP packet header, using six bits from
the deprecated IP Type of Service (ToS) header to carry
the service coding. The network then uses this service code
in the packet header to treat this packet identically to
all other packets within the same service code.
Neither of the two approaches has proved adequate. With
one approach imposing an excessive load in the core of large
networks through the imposition of a per-application path
state and the other while providing superior scaling properties
through the use of aggregated service elements, includes
no concept of control signaling to inform the traffic conditioning
elements of the current state of the network, or the current
per-application requirements, the question is whether a
combination of the two approaches would be the answer.
The telecommunications industry has been challenged. Voice
moved from analog to digital translation and transportation
over a PSTN, making the geographical divided insignificant.
Telecommunication has traversed extensively. However, challenges
defy innovationsbe they technology or otherwise. Global
players are running this race to make voice a run of the
mill entity in the telecom space. Voice will ride the tube
with data and will be just another application on the global
Will tech gurus then speculate 'free voice for all' as a
future? This is one question technology and the burgeoning
riders of the new Telecom Arc will have to address at haste.
end users demand more quality and
reliability, providers are showing more concern towards
IP testing for end-to-end QoS
provide a set of tools that can be used by the network administrator
to manage network resources in a
efficient manner The
for IP Telephony is to ensure delivery of the same quality
of reliable voice conversations that users have come to
expect after a century
of experience over
the PSTN lines
Naresh Wadhwa, Head-SP Sales, Cisco India, can be reached