rhodes university - department of computer science 1 re-establishing and improving the experimental...
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Rhodes University - Department of Computer Science
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Re-establishing and improving the experimental VoIP link with the
University of Namibia: A Case Study
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Alfredo Terzoli and Mosiuoa Tsietsi
Ricky [email protected]
Rhodes University - Department of Computer Science
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Project Aim/Objective
• Establish a fully working VoIP link between RU and UNAM
Rhodes University - Department of Computer Science
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Approach
• Learn about asterisk and gain practical experience
• Learn about traffic shaping and test traffic shaping techniques on a network test-bed
• Learn about UNAM’s network and their ISP SLA• Deploy iLanga at UNAM• Implement traffic shaping• Test the solution
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RU – UNAM VoIP Link
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Live Network • RU has a 12 224 Kbps Internet connection
• UNAM has a 1 024 Kbps Internet connection
• During working hours there is a high rate of network congestion at UNAM
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Network Test-bed
• Build a network to emulate the existing live network between RU and UNAM• Test call quality under normal and congested conditions• Implement traffic shaping and test call quality under normal and congested conditions
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Results • Iperf program was used to measure
bandwidth and throughput• ManageEngine VQManager was used to
analyse VoIP and network traffic• Call quality is calculated and rated by
MOS (Mean Opinion Score). MOS is described in ITU P.800
• MOS rates are between 1 and 5. MOS above 4 are toll quality and those below 3.6 are not satisfactory
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Results (cont.)
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Results (cont.)
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Deployment Challenges• Geographic separation makes it
challenging to streamline VoIP deployment between UNAM and RU
• UNAM uses a Telecom owned ISP and their bureaucracy has stalled deployment due to blocked VoIP ports
• Low bandwidth was a major headache during installation and testing
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Challenge Solutions
• To resolve the technical and bureacracy surrounding VoIP we opted to use two test sites UNAM and Polytechnic of Namibia (PoN)
• We opted for IAX over SIP to bridge the two PBXs. This is due to security restrictions, codec limitations and a NAT-ed network
• Speex codec works best over IAX between the sites. Speex compresses between 2 – 44Kbps
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Conclusion
• Traffic shaping improves the quality of voice calls over congested networks
• IAX is better to use over congested and NAT-ed networks for audio but not video because there has been a slow development for video handling
• SIP is better to use for video but is difficult to run over NAT-ed networks
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Future work
Although SIP is more widely used than IAX, SIP does not handle NAT-ed networks well.
• Develop clients and codecs that handle video over IAX
• Improve SIP to handle NAT-ed networks better
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Questions?