dslreports logo
site
 
    All Forums Hot Topics Gallery
spc

spacer




how-to block ads


Search Topic:
uniqs
972
share rss forum feed

minoe

join:2003-11-11
St George Brant, ON

[General] voip.ms Ping vs. routing

Just looking for general advice on choosing a voip.ms server to use.
Is it better to have a) lower ping time, or b) fewer hops in the routing?

Up until now, I've been using the Atlanta server, which has a ping of about 42 ms over 11 hops.

I'm thinking of giving the Toronto or Montreal servers another shot, now that there has been some improvements made (?).

My ping to the Toronto servers are around 15ms but over 10 hops.
My ping to the Montreal servers are around 25ms but over 8 hops.

Any general advice about ping vs. hops or does any of it really mater for overall voice quality?

adatech

join:2010-04-23

1 recommendation

Jitter is much more important than either of those qualities. I'd suggest focusing on that. 100ms with 0ms jitter will always beat 40ms with lots of jitter.

ajeff

join:2007-07-30
Orleans, VT
reply to minoe
Agreed that jitter is more important than ping times. I live in VT on the Canadian border and have used Montreal 1 and 2 and Toronto on occasion but have had best luck with New York. Your results may vary...


crazyk4952
Premium
join:2002-02-04
united state
kudos:1
Reviews:
·CenturyLink
·Vitelity VOIP
·Charter
·Callcentric
reply to minoe
said by minoe:

Just looking for general advice on choosing a voip.ms server to use.
Is it better to have a) lower ping time, or b) fewer hops in the routing?

Up until now, I've been using the Atlanta server, which has a ping of about 42 ms over 11 hops.

I'm thinking of giving the Toronto or Montreal servers another shot, now that there has been some improvements made (?).

My ping to the Toronto servers are around 15ms but over 10 hops.
My ping to the Montreal servers are around 25ms but over 8 hops.

Any general advice about ping vs. hops or does any of it really mater for overall voice quality?

If the atlanta server is working for you, stick with it. You will not notice a few extra milliseconds or 2 less hops.

I am currently using the houston server even though there are 2 servers that are much closer to me. However, I have had less reliability with both of those servers than the houston server.

adatech

join:2010-04-23
said by crazyk4952:

I am currently using the houston server even though there are 2 servers that are much closer to me. However, I have had less reliability with both of those servers than the houston server.

Likewise. I'm in the Bay Area and use Houston. Seattle & LA have been less reliable. My ping to Houston is only around 50ms, and I get 0-3ms jitter.

MartinM
VoIP.ms
Premium,VIP
join:2008-07-21
kudos:3
reply to minoe
Although it's true that Jitter is very important, if Jitter is not an issue for you, assuming you have a good internet connection, the next most important factor is latency. So you can go with that if you don't have the tools to measure Jitter. The most important element is packet loss, way before jitter.

Packet Loss -> Jitter -> Latency

Of course if latency passes a certain threshold, then it becomes as important as the other factors (For example, My friend's fishing boat has Satellite Internet, the servers where I can get 500 latency are much better than the server I get over 600 MS, regardless if the jitter is a bit better, because the conversation becomes too delayed)

So with all that jargon and personal recommendations, what you should do? Try the server closest to you first, and see how it goes. Because you're in Ontario, Toronto or Montreal.

--
Martin - VoiP.ms


ThaiGuy

join:2008-05-10
Thailand
What tools can be used to measure Jitter and are there any guides available to help VoIP'ers to measure Packet Loss, Jitter and Latency?

OZO
Premium
join:2003-01-17
kudos:2
said by ThaiGuy:

What tools can be used to measure Jitter and are there any guides available to help VoIP'ers to measure Packet Loss, Jitter and Latency?

+1

It's be nice to find out a tool, that will do the job. Any suggestions?
--
Keep it simple, it'll become complex by itself...


LHI

@teksavvy.com
reply to ThaiGuy
said by ThaiGuy:

What tools can be used to measure Jitter and are there any guides available to help VoIP'ers to measure Packet Loss, Jitter and Latency?

Try Pingtest.net

rudeboy24

join:2002-10-14
Welland, ON
reply to ThaiGuy

MartinM
VoIP.ms
Premium,VIP
join:2008-07-21
kudos:3
said by rudeboy24:

Here's another one

»myspeed.visualware.com/indexvoip.php

I second that one, very complete and visual graphics are neat. You can also choose your codec. My only con is you need to install java.
--
Martin - VoiP.ms


VexorgTR

join:2012-08-27
Sheffield Lake, OH
kudos:1
As much as I'd like to say the testing is really valid, the Internet is weird in the hops/pings and such... sometimes you just have to TRY a server and see how it works for you.

The tests are fun though.

minoe

join:2003-11-11
St George Brant, ON
reply to minoe
Lots of useful info here. Thanks for your thoughts.

Further to the question of jitter testing, how is it possible to test jitter to a specific location. I've use the pingtest.net and myvoipspeed tests but they only really test the front end of my internet connection, or the connection to their servers.
Are there tools that allow jitter testing of routes to the voip.ms servers?

adatech

join:2010-04-23
I typically will register to a specific server and dial the voip.ms echo test (4443). Play some music into the receiver, and monitor the call status on my Obi for 5-10 minutes.

MartinM
VoIP.ms
Premium,VIP
join:2008-07-21
kudos:3
said by adatech:

I typically will register to a specific server and dial the voip.ms echo test (4443). Play some music into the receiver, and monitor the call status on my Obi for 5-10 minutes.

That's a good test and you can pretty much listen to realtime latency and how's your sound input with your MIC or Phone.
--
Martin - VoiP.ms

OZO
Premium
join:2003-01-17
kudos:2
MartinM See Profile, I think for those of us, who don't use OBi, it'd be nice if you offered a simple application, that makes a test call to SIP server using your dedicated test account, sending specific test audio stream back and forth, measures stats and report them back to user. Application could be based on any free SIP stacks available around. No OBi device or using some personal account would be required. And the test results would be actually accurate. I guess we'd all appreciate such tool, offered by your service
--
Keep it simple, it'll become complex by itself...