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Published byDwight Thornton Modified over 9 years ago
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1-800-CALL-H.E.P. Warren Matthews (warrenm@slac.stanford.edu) Les Cottrell (cottrell@slac.stanford.edu) Rebecca Nitzan (nitzan@es.net)
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Overview Review ESnet VoIP testbed Discuss and Review Metrics Define Quality Compare to Real Internet Production VoIP Service
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3.5 Mbps carved from ESnet ATM Backbone
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Hardware Regular Telephone PBX (Nortel) Individual E&M (Ear and Mouth) Analog Voice Trunks. Cisco 3640 VODEC
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Software VODEC –ITU G.729 Policing –Cisco IOS 12.0(4.4) Weighted Fair Queuing (WFQ) –Cisco experimental IOS
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A Typical VoIP call Call set up with TCP Voice transmitted with UDP Managed with RTP
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UDP Packet Header IP UDP RTP
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Aims How can VoIP be used on the Internet –under what loss/RTT (B.E. and priority) –assume some kind of QoS is required to be useful –Telephone companies claim 99.999% availability Metrics –loss, RTT, Jitter, IQR
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Loss Loss of TCP packets results in call failing to initiate, but will be re-sent Loss of UDP packets causes break in conversation –routing change can cause seconds of loss
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Loss Plot RTP seq vs Time Seq 0-255 Time Stamp is random (RFC 1889) Require 56 concurrent calls to fill the pipe. RTP Seq # Time Stamp
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Jitter Jitter is the variation in delay. IETF defines IPDV Audio is particularly sensitive to jitter.
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On the Real Internet
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Differentiated Services Packet Loss, and hence loss of voice is unacceptable Use Weighted Fair Queuing (WFQ) Per-Hop Behavior (PHB) and Expedited Forwarding (EF) require Bandwidth Broker and associated infrastructure Inter-network DiffServ still in the future
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Production Service ESnet –How could/should applications set PHB –Policing –Extend test bed (Italy?) Internet 2 (vBNS/Abilene) Qbone –May include CERN
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Conclusion VoIP is capable of high quality calls. Work in Progress. Future Work ?
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Any Questions ?
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VoIP - uses voice component of H.323
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H.323 infrastructure
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Voice over IP protocols From “VoIP Networking Design” by tdanford.cisco.com http://www.cisco.com/networkers/presentations/voice/1086_038.ppt
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