Mohammad Faisal Kairm(073714556) Assignment 2


Published on

Published in: Business, Technology
  • Be the first to comment

  • Be the first to like this

No Downloads
Total views
On SlideShare
From Embeds
Number of Embeds
Embeds 0
No embeds

No notes for slide

Mohammad Faisal Kairm(073714556) Assignment 2

  1. 1. 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
  2. 2. Title: IP telephony for reduction of transmission cost of PSTN/PLMN. 1. Introduction: 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 –
  3. 3. 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 Backbone Dhaka MSC Bogra MSC TDM Connections 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 enterprise solutions. 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
  4. 4. 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 platform for complex IP Telephony deployments. Stable and Reliable; There are many high quality, low cost IP Telephony hardware and software 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 protocols which means an abundance of knowledge resources exist for IT departments and decision makers. 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 Low risk
  5. 5. 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 cheaper rate. The network diagram will look like following: Dhaka Bogra Bogra Dhaka TDM Media Media TDM MSC compr IP Backbone compr MSC essor essor /Route /Route r r 5. Problem of Implementing IP backbone for Voice: The problem involve IP backbone for voice is mentioned following: Network Quality
  6. 6. 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. Packet Loss 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. Jitter 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
  7. 7. 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. Latency 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 so. 6. Conclusion:
  8. 8. 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 objectives. 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 bandwidth budget. After meeting all these requirement we could say that voice network over the IP backbone will work well and meet customer’s demand.
  9. 9. 7. References: 1. Voice over IP Technologies, Building the converged network, by Mark A. Miller. 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.