ETE 605: IP Telephony
Title: IP telephony for reduction of transmission
cost of PSTN/PLMN
Course Instructor: Dr. Mashiur Rahman
Submitted by: Mohammad Faisal karim
ID: 073 714 556
Date of Submission- 15/04/08
Title: IP telephony for reduction of transmission cost of PSTN/PLMN.
Enterprises are rapidly adopting IP telephony for cost savings, productivity gains
and business innovation, but delivering a high quality voice service takes more
than just buying the latest IPtelephony equipment. Successfully deploying IP
telephony to your enterprise also means understanding the requirements for
delivering toll-quality voice over your company’s network infrastructure, and then
appropriately planning for, choosing and deploying the right IP telephony solution.
In this assignment I tried to focus on IP telephony implementation on traditional
PLMN ( Public Land Mobile Network) which is being termed as mobile network
backbone facilities as well as traditional PSTN ( Public Switched Telephone
Netwrok) for their backbone transmission capabilities.
2. PLMN/ PSTN Backbones Used Now:
Traditional Phone company’s now uses E1/T1 lines or STM 1/4/16/64 to connect
their exchanges. For Mobile Operator’ s it the connection for MSC to MSC or
MSC to GMSC( Gateway MSC). For PSTN it used for connection of there
Tandem Exchanges. For Bangladesh consideration we only focused on E1
standard. It is the lowest tributary standard for SDH transmission.
Each E1 connection has the data rate capacity of 2.048 Mbps. It has 32 channels
each of 64 Kbps. An STM -1 has 63 E1 capacity and higher STM’s are multiples
of this E1 capacity. There are two types of connection between the MSC’s –
One for data/Signalling connection and Other for Voice. As the MSC’s only
reconginse 64Kbps voice channels, They can utilize 31 timeslots of E1 connection
for voice. But implementation of ip telephony can change the way that the voice
traffic handled by this MSC
The Backbone Links now used by MSC’s can be viewed as following:
Consider one of the Backbone Teletalk Bangladesh Limited’s Dhaka-Bogra
Dhaka MSC Bogra MSC
3. Why to Switch from TDM to IP:
Big business features with small business pricing. IP Telephony represents a phenomenal
value when compared to
traditional telephony solutions with a level of flexibility yet to be available in high end
Decentralized and Immediate Deployment; With the increasing availability of high
quality, reliable Internet connectivity businesses can now comfortably deploy production
IP Telephony solutions without fear of dialtone outages. Where reasonably well-
engineered existing data networks exist almost immediate deployment of an IP
Telephony solution is possible. The disparate nature of the Internet allows for deploying
or using centralized solutions with end user phones located in any geographic region.
Extremely Cost Effective:
No government regulation or taxes
Often no long distance charges
Usually priced per minute or a monthly flat fee
Simple, unified wiring infrastructure
Flexibility; Since IP Telephony in most deployments uses the Internet as the
communication medium it
becomes an easy task to move, merge or split offices between physical locations.
Working from home is
easy and seamless. There really are no geographic constraints.
Scalable; The pre-existing and distributed nature of the Internet provides the perfect
complex IP Telephony deployments.
Stable and Reliable; There are many high quality, low cost IP Telephony hardware and
products and services to choose from.
Choice; Businesses have an abundance of IP Telephony Providers and Carriers to
choose from since
locality is no longer an issue. Choice breeds competition and competition breeds
excellence and drives low
pricing, a win-win scenario for businesses.
Open Standards; IP Telephony technology is built on industry standardized and open
means an abundance of knowledge resources exist for IT departments and decision
Investment Protection; Open Standards means the equipment you purchase today for
one solution will
most likely work with other solutions tomorrow.
Extended Value with Hosted Solutions:
An immense strategic advantage to small and medium sized businesses
Absolutely no in-house expertise required.
No expensive, proprietary PBX or Key System equipment to purchase
Reduced operating costs
4. Proposed Network Of IP for Voice Trunks:
As the voice trunks/circuits are 64Kbps we can use compression techniques for
this kind of event. As ip telephony support various kinds of codecs which have
low bit rate other than the traditional one such as G.723, G.723.1 , G.726 etc and
implement IP backbone for this. This would save much cost of implementation of
TDM based voice circuit and reduce transmission cost. As it would be IP
backbone network, Network Management features will also be available at much
The network diagram will look like following:
Dhaka Bogra Bogra
MSC compr IP Backbone compr MSC
5. Problem of Implementing IP backbone for Voice:
The problem involve IP backbone for voice is mentioned following:
The first thing that VOIP requires of a network has to do with its basic ability to
deliver most packets on time. In general, TCP/IP networks do not guarantee that every
packet sent will be delivered. Routers along the way have the option of dropping
packets, if necessary, so some packets that are sent will never arrive at their destination.
Since each packet can take a different route from the source device to the destination
device, they can arrive out of order and take a different amount of time to travel the
same distance. Finally, the total time it takes a packet to go from one location to the
other may be very large; too large to allow voice conversations to progress normally.
We will address each of these issues with regard to what is required for voice traffic.
One aspect of network quality is packet loss. This is a quantity that can be
measured by the network analysis tools that we will discuss later. The quantity is the
percentage of packets that are sent from one end of the network connection that do not
reach the other end. Networks with a packet loss of more than 3% are not good
candidates for VOIP, as there will be dropouts in the audio. Packet loss increases
sharply at the point where the network is overloaded with traffic. For this reason, packet
loss testing must be done in conjunction with bandwidth and QOS testing. On a lightly-
loaded network, packet loss may be low, but it may become unacceptably high when
the number of packets reaches the maximum that the network can accommodate.
Another network quality issue to examine is jitter. Each packet of voice
information takes a different amount of time to go from one end of the network to the
other. This variation is called “jitter”. The VOIP equipment on the receiving end is
responsible for putting the packets into a buffer so that they can be played out as an
unbroken stream of audio. The buffer that is used for the purpose is called a “jitter
buffer” and it is a certain length in milliseconds. This length is called the “jitter buffer
depth”. This depth should be about twice the size of the largest jitter value that actually
occurs on the network.In networks that display jitter values that are larger than 50
milliseconds, it is difficult or impossible to play the packets smoothly using a jitter
buffer of reasonable depth. In these poorly-behaved networks, the receiving device will
reset its jitter buffer frequently, leading to noticeable dropouts in audio. It is important
to set the jitter buffer depth in the VOIP devices to match the behavior of
the network. If you set it too low, you will hear dropouts in the audio. If it is too high,
there will be an unnecessary delay in the audio. Dropouts can also be caused by
packet loss and QOS problems, so you cannot assume that the jitter buffer is too small
just because there are dropouts.
The last aspect of network quality that we will examine is latency. This refers to
the amount of time it takes a packet to get from one end of the network to the other. If
this is long (more than 150-200 milliseconds), it can create problems for the VOIP
equipment that lead to an echo in the audio. If it is very long (more than 400
milliseconds roundtrip), then it will interfere with human conversations. The echo
problem only occurs in certain types of VOIP equipment, where audio echoes back to
the sender with some delay. When this delay is small (less than 20 milliseconds) there
is an algorithm in the sending device that recognizes the echo and removes it. When the
delay is very small (less than 28 ms), no echo handling is necessary, since a person will
not notice the echo. When the delay is large (more than the size of the echo canceller’s
buffer), the sending device will have forgotten that it sent that sound so it will not be
able to recognize it as an echo.When a connection is all-digital to a digital phone at the
receiving end, then echo is not always an issue.The problem in conversation, however,
is always present when the round-trip latency (RTL) is more than 400 milliseconds or
The objective in voice network planning is to create a network that will carry
the traffic with delays that users can live with. The objective in creating a voice
network is to build one that will support voice conversations that users will consider
toll-quality.Designing a converged voice-data network requires to meet both of these
Specifying routers, switches, traffic shapers, and ISP partners that will support
the QOS features described in this paper.
Having or finding the expertise necessary to configure that equipment
Planning bandwidth capacity that is the larger of:
Three times the Voice over IP bandwidth
Voice + Data + 30%
Choosing Codecs that meet your voice quality needs while fitting within the
After meeting all these requirement we could say that voice network over the IP
backbone will work well and meet customer’s demand.
1. Voice over IP Technologies, Building the converged network, by Mark A.
2. Configuring Cisco Voice Over IP, By Cisco Inc.
3. IP Telephony: The Good, The Bad, The Ugly by Bill Kervaski [White Paper].
4. Strategies for Successful IP Telephony Implementations, by Peter Brockmann
, 3Com Enterprise.[White Paper].
5. Planning an IP Network for Voice and Data, by NEC Unified Solutions, Inc.
6. Voice Over IP (VoIP) ImplementationGuide for Network Performance
Management by NetScout Systems, Inc.