Objective: Students will leverage existing tools and equipment in the lab to design and develop a SIP based system to place and receive phone calls. This system will be tested using open source traffic generators to baseline and determine the network performance parameters involved in degrading real time traffic performance. Students will need to support their work with documentation, drawings, description of the network system, protocols and how congestion could be managed. Overview: Students are to design a simple network that connects 4 distinct broadcast domains, distinguishable by IPv4 networks. At minimal, two routers should be connected via a serial link with no more than two networks on each router. LinPhone is recommended as the SIP application. The codec should be determined and confirmed by deep packet analysis. The protocol data unit stack should be notated and documented as well. Phone calls should be placed to test SIP functionality. Network performance should then be tested. Next, all four network performance parameters should be measured, while determining the point of degradation of voice quality across the SIP network system. The application of SIP may or may not have any security concerns, if used over the public Internet. Students should consider what security implications are involved with SIP traffic traversing across the public Internet. The network that is implemented should have documentation to reproduce the network and for anyone to follow what was accomplished in the lab. Prelab Questions 1. What is SIP? What other VoIP protocols are available? What are some differences between SIP and other VoIP protocols? 2. What is inelastic traffic? How does inelastic traffic relate to real time traffic? How does VoIP relate to real time traffic? Is VoIP soft or hard real time traffic? 3. What is an example other real time application types? Requirements: 1. Four distinct IPv4 networks/subnets 2. Minimal of 2 routers connected by serial cables 3. A dynamic routing protocol used for IPv4 connectivity of networks 4. LinPhone is recommended as the SIP application 5. One phone call must be initiated, received and completed between all four networks 6. Baseline performance using ping, IPerf, and the SIP application 7. Determine the point of degradation by testing with voice quality 8. Determine the point of degradation by testing by reviewing network performance parameters 9. Document the point of degradation 10. Design and discuss a Quality of Service plan along with example block diagram and configuration to support managing congestion for the SIP network system Deliverables: 1. Detailed explanation of SIP network system 2. Drawing with the topology of the SIP network system 3. Table of IPv4 addresses and interface names for workstations and network devices 4. Measurements of SIP network system baseline 5. Measurements of SIP network system under congestion/point of degradation 6. Explanation of SIP network system congestion observations from voice quality versus analyzing network performance parameters 7. A Bill of Materials (BOM) of all the devices used to produce this system 8. Configuration of all devices involved including network devices and workstations 9. A Quality of Service plan to address congestion with block diagram, example configuration and discussion of why QoS strategy was chosen 10. A discussion of what VoIP codec was used, its equivalent data rate, and proof of how this codec was confirmed via packet analysis 11. What should be done the improve the voice network 12. Think about ( how to force the degradation to occur) (lab scenario)