WOW.. That is the most complicated Ping I have ever seen.. :) This is the result I got.
# ping -f -i .02 -s 180 -Q 0xb8 xx.xx.xx.xx *PING xx.xx.xx.xx (xx.xx.xx.xx) 180(208) bytes of data. ............. --- xx.xx.xx.xx ping statistics --- 15338 packets transmitted, 15325 received, 0% packet loss, time 352748ms rtt min/avg/max/mdev = 276.499/286.185/310.118/7.248 ms, pipe 15, ipg/ewma 22.999/284.882 ms * The same test with my Present SIP Provider gave me the result below. *10926 packets transmitted, 10913 received, 0% packet loss, time 244048ms rtt min/avg/max/mdev = 289.514/292.668/316.350/2.336 ms, pipe 15, ipg/ewma 22.338/292.941 ms * I suppose the value of mdev is much higher in the first case but 0% packet loss in both the cases. Does this mean that the voice quality is going to be real bad?? Thanks, Najim On Thu, Dec 1, 2011 at 6:33 AM, Adam Moffett <adamli...@plexicomm.net>wrote: > > a ping is the time a packet needs for travelling to a destination and >>> back to you. So the one way latency you are refering to, should be half >>> the time your ping took. >>> >>> In your case this will be 130ms, I would say this is still reasonable. >>> >> I am probably splitting hairs, but that's not always true because there's > no guarantee that the reply traveled the same path as the echo request. If > you dig into BGP issues you'll see sometimes that traffic one direction > takes a different route than traffic the other direction. I don't know of > any simple and accurate way to learn the "one way" latency so I'm surprised > they specified anything other than round trip time. > > > 'Ping time' is not an accurate predictor of SIP quality. >> >> A 'ping' is an ICMP Echo/reply packet and some routers consider them less >> important than 'data' packets and service them on an 'as resources permit' >> basis. >> > That's possibly maybe true if someone's router or connection is overloaded > and they are trying to make up for it with CoS policies while they save up > for an upgrade. Otherwise it's an apology for a crappy network. That's > the brutally honest truth. > > You can make a pretty good prediction with ping. > "sudo ping -f -i .02 -s 180 -Q 0xb8 [ip]" gives a tolerable simulation of > voip traffic. let it run for awhile, then press ctrl+c and see how many > packets were dropped and also check the mdev number. If mdev is low and > packet loss is almost nothing then you can expect decent voice quality. It > may not be a 100% perfect test, but I'll bet you a vast majority of the > time I can do that test and tell you whether it's going to suck. > > latency by itself with low jitter and no packet loss just means delay. > It's a matter of opinion and circumstance how tolerable delay is, but I > think your 230ms ping is at the upper edge of what most people can live > with. Much more than that and you'll be tempted to say 'over' at the end > of sentence. > > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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