1-800-CALL-H.E.P. Warren Matthews Les Cottrell Rebecca Nitzan

Overview Review ESnet VoIP testbed Discuss and Review Metrics Define Quality Compare to Real Internet Production VoIP Service

3.5 Mbps carved from ESnet ATM Backbone

Hardware Regular Telephone PBX (Nortel) Individual E&M (Ear and Mouth) Analog Voice Trunks. Cisco 3640 VODEC

Software VODEC –ITU G.729 Policing –Cisco IOS 12.0(4.4) Weighted Fair Queuing (WFQ) –Cisco experimental IOS

A Typical VoIP call Call set up with TCP Voice transmitted with UDP Managed with RTP

UDP Packet Header IP UDP RTP

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 % availability Metrics –loss, RTT, Jitter, IQR

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

Loss Plot RTP seq vs Time Seq Time Stamp is random (RFC 1889) Require 56 concurrent calls to fill the pipe. RTP Seq # Time Stamp

Jitter Jitter is the variation in delay. IETF defines IPDV Audio is particularly sensitive to jitter.

On the Real Internet

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

Production Service ESnet –How could/should applications set PHB –Policing –Extend test bed (Italy?) Internet 2 (vBNS/Abilene) Qbone –May include CERN

Conclusion VoIP is capable of high quality calls. Work in Progress. Future Work ?

Any Questions ?

VoIP - uses voice component of H.323

H.323 infrastructure

Voice over IP protocols From “VoIP Networking Design” by tdanford.cisco.com