Subject: RE: [Ieprep] Qualitative statement about quality

From: "Nguyen, An" <[email-masked]>
Date: Sat, 16 Nov 2002 10:07:14 -0500

>I can't quite make out what you're saying here.  600ms was the observed
l>atency between San Luis Obispo and Nairobi, at the time of the call.
>The
>observed latency between San Luis Obisbo and Berkeley was about 200ms
>across four ASes.  As I trace Berkeley to Nairobi right now, I get about
>950ms, 42% drop, and seven ASes in the path.  It's not peak use hours in
>either location at the moment, and this is a path which we use many
>times
>each day for routine calls, when it's qualitatively much worse than it
>is at
>the moment.  The quality of the VoIP calls is much, much higher than
>that
>of PSTN calls, and it's difficult to keep the PSTN calls up...  Initial
>call-completion rate is only about 80%, and most calls are dropped
>within
twenty minutes.  We've never had a failed connection or a dropped call
between this pair of VoIP phones, by contrast.

>router.brk#trace nbi.ispkenya.com
>Translating "nbi.ispkenya.com"...domain server (131.161.3.1) [OK]

>Type escape sequence to abort.
>Tracing the route to nbi.ispkenya.com (212.49.87.2)

>  1 131.13.22.157.zocalo.net (157.22.13.131) [AS715] 4 msec 4 msec 8
>msec
>  2 gige-g2-0.gsr12008.pao.he.net (64.71.128.117) [AS 6939] 4 msec 4
>msec 0 msec
>  3 interconnect-eng.SanJose1.Level3.net (209.245.146.201) [AS 3356] 0
>msec 20 msec 4 msec
>  4 GigabitEthernet3-0.core1.SanJose1.Level3.net (209.244.3.245) [AS
>3356] 4 msec 4 msec 4 msec
>  5 ae0-51.mp1.SanJose1.Level3.net (64.159.2.1) [AS 3356] 4 msec 4 msec
>4 msec
>  6 so-3-0-0.mp1.NewYork1.Level3.net (64.159.1.41) [AS 3356] 80 msec 80
>msec 80 msec
>  7 gige9-0-53.hsipaccess1.NewYork1.Level3.net (64.159.17.71) [AS 3356]
>76 msec 76 msec 76 msec
>  8 unknown.Level3.net (209.244.160.242) [AS 3356] 81 msec 92 msec 80
>msec
>  9 host-66-133-0-22.verestar.net (66.133.0.22) [AS 11908] 156 msec 156
>msec 157 msec
> 10 unknown.Level3.net (64.158.116.117) [AS 11908] 188 msec 184 msec 185
>msec
> 11 host-66-133-6-2.verestar.net (66.133.6.2) [AS 11908] 184 msec 184
>msec 185 msec
> 12  * host-66-133-6-10.verestar.net (66.133.6.10) [AS 11908] 725 msec
>725 msec
> 13 212.49.70.109 [AS 12455] 745 msec 741 msec 753 msec
> 14  * 10.10.2.10 781 msec *
> 15 64.86.231.2 [AS 6453] 978 msec 901 msec 1859 msec

When you did a trace route from 131.13.22.157 (local router) to 
64.86.231.2 [AS 6453], the trace route packet went through 
AS715  -  4 ms (one-way average delay)
AS6939 -  4 ms
AS3358 -  20ms
       -  4ms
       -  4ms
       -  80ms
       -  76ms
       -  92ms
AS11958-  156ms
       -  184ms
       -  184ms
       -  725ms
AS14255-  741ms
10.10.2.10 - 781ms
AS6453 -  901ms
-------------------
Total  = too much

If you add all the delay numbers from the orgination [aS715] to destination
[AS6443], the total will be too much of a delay to provide voiceIP or  any
time-sensitive, interactive applications objectively. The is a certain delay
allowed for a certain voice codec. In order to provide voiceIP service with
some QoS, you have to make sure that your parameters stay within bounds.

>router.brk#ping
>Protocol [ip]:
>Target IP address: 212.49.87.2
>Repeat count [5]: 100
>Datagram size [100]:
>Timeout in seconds [2]:
>Extended commands [n]:
>Sweep range of sizes [n]:
>Type escape sequence to abort.
>Sending 100, 100-byte ICMP Echos to 212.49.87.2, timeout is 2 seconds:
>.!!!!!.!..!!!!.!!...!!!!.!!!.!!...!!!!!.!!...!...!.!!.!...!!.!!!!...!.
>..!..!!.!.!!!!!!.!.!!!!!.!..!.
>Success rate is 58 percent (58/100), round-trip min/avg/max =
>774/959/1262 ms
>router.brk#

I assume you did an ICMP ping on a router you were on to this node
212.4987.2, the minumm round-trip delay is 774, average 959 , and the max
1262 ms. I don't know how far (or how many hops) this ICMP packet traveled,
but the round-trip delay seems a little too much for voiceIp.

Regards,

An

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---

From: Bill Woodcock <[email-masked]>
Date: Sat, 16 Nov 2002 07:41:09 -0800 (PST)

On Sat, 16 Nov 2002, Nguyen, An wrote:
    > When you did a trace route from 131.13.22.157 (local router)

Actually that's the next-hop router...  I did the trace from the router
which the phone is behind, not the phone.  The phone is not a BGP-speaker
itself.

    > If you add all the delay numbers from the orgination to
    > destination, the total will be too much of a delay to provide
    > voiceIP.

I'm not asking you whether it _would_ work.  I'm telling you that it
_does_ work, absolutely repeatably, day in and day out, and it _does_ work
_substantially better than the PSTN_, which nearly inevitably fails on any
call longer than a few minutes duration for the same pair of endpoints.

                                -Bill