1. 3/4/981 Internet Telephony & Internet Performance Issues Les Cottrell SLACSLAC Presented at the XIWT/IPWT meeting San Jose February 4th, 1998 Partially funded by DOE/MICS Field Work Proposal on Internet End-to-end Performance Monitoring (IEPM)

2. 3/4/982 Outline of talk Credits VoIP –Why, how, setup, how to use, protocols, traces, results so far, next Relating to PingER performance measurements –delay, loss, jitter, availability

3. 3/4/983 Credits VoIP –SLAC - Charley Granieri, Dave Millsom –CERN - Olivier Martin, Denise Heagerty –DESY - Michael Ernst, Andrey Bobyshev –FNAL - Phil DeMar, Vyto Grigaliunas –LBNL - Becca Nitzan, Mike Collins Monitoring –Warren Matthews, Les Cottrell - SLAC

4. 3/4/984 Internet Telephony Pilot Goal is to understand issues and see if, when and where VoIP is applicable for HEP collaborations Five National Lab sites (CERN (ch), DESY (de), FNAL(Chicago), LBNL (Berkeley) & SLAC (Silicon Valley)) –Testing/evaluating VoIP between them –Various PBXs including Alcatel, Siemens?, Lucent 5ESS, Nortel SL/1

5. 3/4/985 Typical Set Up Production Internet with no special purpose router code Nortel PBX Alcatel PBX Cisco Gateway Cisco Gateway SLACCERN PSTN G.729

6. 3/4/986 Mechanism User calls access code to get thru PBX to gateway Gateway gives 2nd dial tone, enter site code, then get dial tone from site PBX, enter extension at site Encoding/compression done by gateway (64kbps=>24kbps (8kbps w/o headers)) ~ 60 byte IP packets using UDP & RTP

7. 3/4/987 Headers Overhead in IP packet 20 ms @ 8kbps yields 20 byte payload IP headers 20; UDP header 8; RTP header 12 bytes 2 x payload! Plus media headers

8. 3/4/988 VoIP Packet traces

9. 3/4/989 VoIP Inter Packet Delay distributions Both UDP streams viewed on wire at SLAC end

10. 3/4/9810 Results Tried two vendors’ gateways –voice quality on first poor –voice quality on second very usable Occasional drop out (every few minutes) Delay OK No echo Lots of teething problems: –difficulty in disconnecting when hang-up –hard to get gateway vendors and PBX vendors “all in the same room” & to be able to talk in each others terms Cisco sent 3 people (escalation team=1 PBX expert, 1 VoIP expert, 1 cable expert).

11. 3/4/9811 Scale of Measurements 17 Monitoring sites - 7 in US (5 ESnet, 2 vBNS), 2 in Canada, 6 in Europe (ch, de, dk, hu, it, uk), 2 in Asia (jp, tw) 1129 monitoring-remote-site pairs 379 unique hosts 272 sites 27 countries Data goes back 4 years Metrics include response, jitter, loss, reachability

12. 3/4/9812 CERN Link to US very busy

13. 3/4/9813 QoS All 4 sites set precedence bits for VoIP traffic Precedence only takes effect on CERN-US link What does ping performance look like for major metrics (delay, “jitter”, loss, reachability...)?

14. 3/4/9814 Nov-98 100 byte pings SLAC - CERN Percent loss Response in msec G.114 300msec RTT limit Packet loss = 0.1% Median response = 182 msec

15. 3/4/9815 It’s getting better all the time 1/2

16. 3/4/9816 What about loss? BCR Feb ‘98 & Jan ‘99 shows even with 10% random loss can get almost toll quality Our experience in other areas is to say problems start between 2.5 and 5% packet loss ITU/TIPHON defines a loss of < 3% as being “good” for Internet telephony Consecutive frames lost:12 345 Quality (MOS scale 1-5)4.23.22.42.11.7 Perception goodfairpoor v.poorunsat if 20 msec between VoIP packets, then lose adjacent packets: –every 2 secs for 10% loss, every ~ 30 sec for 2.5% loss

17. 3/4/9817 It’s getting better all the time 2/3

18. 3/4/9818 It’s getting better all the time 3/3 SLAc vBNS ~ 2 * SLAC ESnet

19. 3/4/9819 Countries Expected to be Good

20. 3/4/9820 What about “jitter”? Jitter ~ instantaneous variability of delay ITU/TIPHON defines a one-way jitter of < 75 msec. as providing “good” Internet telephony –< 125msec is “medium” quality and < 225msec is “poor” How may it be measured? –ITU specifies inject packets at regular intervals and measure the variability as received –IETF has the one-way Instantaneous Packet Delay Variability (IPDV) draft –we are experimenting with 3 ways

21. 3/4/9821 RTT frequency histogram snapshot

22. 3/4/9822 Instantaneous inter-packet RTT

23. 3/4/9823 Instantaneous inter-packet RTT magnitude

24. 3/4/9824 Packet Jitter CERN link utilization TIPHON delay jitter threshold

25. 3/4/9825 Jitter Correlations

26. 3/4/9826 Effect of setting precedence on jitter January 9, 1999

27. 3/4/9827 Effect of setting precedence bits January 8-9, 1999

28. 3/4/9828 Reachability Unreachable = all pings are lost Varies a lot from site to site, it’s not phone quality

29. 3/4/9829 Next How does ping jitter relate to VoIP jitter How does one way jitter relate to two-way –compare Surveyor & pingER in more detail Working with CERN & Esnet/LBNL to understand effect of preferential service –Set up test path, look at behavior of important metrics, VoIP and video as vary load and use DiffServ

30. 3/4/9830 More Information & extra info follows WAN Monitoring at SLAC has lots of links –http://www.slac.stanford.edu /comp/net/wan-mon.htmlhttp://www.slac.stanford.edu /comp/net/wan-mon.html Tutorial on WAN Monitoring (including jitter & quality of service thresholds etc.) –http://www.slac.stanford.edu /comp/net/wan-mon/tutorial.htmlhttp://www.slac.stanford.edu /comp/net/wan-mon/tutorial.html PingER History tables –http://www.slac.stanford.edu/ /xorg/iepm/pinger/table.htmlhttp://www.slac.stanford.edu/ /xorg/iepm/pinger/table.html