how to calculate the roundtrip time, delay and jitter

snipped-for-privacy@gmail.com hath wroth:

Since you didn't bother to mention the operating system, I'll assume you're a Windoze user. Mac and Linux users always seem to remember to mention their operating system.

Time delay can be done with just ping. Start -> run -> cmd ping ip_address

For better resolution, I suggest FPING:

You can use ping to obtain a series of latency values and feed them to a spreadsheet or statistics calculator to obtain variations, which is jitter. A simple histogram of ping results is usually sufficiently interesting.

IPerf will measure jitter but only with UDP packets. See:

will not do it as a "round trip" type of measurement because jitter is normally measured one way. You'll need two computers, one running the server half to get jitter results. Ethereal is a packet sniffer and decoder. It offers nothing in the way of performance measuring tools.

What are you trying to accomplish?

hi Jeff: thanks, yes i m a windows user, 2000 to be more accurate, i need this to measure the performance of voip calls, i don't think ping is what I could use, i need something that can give me the round trip time ,delay and jitter at the time when i m sending the packets, any ideas? if it was free much better. Thanks

flo

Jeff Liebermann wrote:

Welcome to the VoIP benchmarking can of worms. The problem is that VoIP quality measurements are subjective and at best guesswork. Attempts have been made to provide a statistical basis for these measurements. They work, but the results are generally useless because they either measure a short time interval or averaged over too long a period to see momentary dropouts.

You also have the problem of having your test software interact with the VoIP traffic. The only way around this is to use the VoIP packets themselves as the test packets. That requires a specialized VoIP tool. If you use something like IPerf *WHILE* you're sending VoIP packets, you'll get the conglomerated result of both programs traffic, which is useless.

I couldn't find much in the way of free VoIP simulation tools.

Seach Google for "voip simulator".

Also note that the common Softphone products (Skype and Gizmoproject) have built in diagnostics which display some of the performance parameters.

There are some online measuring tools for VoIP. For example:

The server is free for the first 2 weeks. Then, it's $400 to $1,200 per year. Ouch.

I don't think I really answered your question, but my comments might give you a start in the right direction. For VoIP questions, you might get better results in a VoIP specific blog or newsgroup. I found plenty with Google and a few with Google Groups, but I don't which ones are the best for VoIP questions.

Also, you might want to get the shift key on your keyboard repaired. It seems to be non-functional.

Hi Jeff:

Thanks for your answers, I will have a look at the links you showed me. I ll come back later to say what I used in the end, so the topic is finished. Regards

flo

for jitter and traffic stats im using Ethereal. There is an option in stats -> RTP -> analize stream this jitter value is calculated according to the RFC 3550 for RTP

I d> Hi Jeff:

Thanks. I didn't know it would do that. I just tried it with both a GizmoProject and a Skype call using WireShark 0.99.4. I couldn't capture any RTP packets. I'm doing something wrong. These might be of interest:

I'll play with it some more when the phone stops ringing. (I hate mondays).

that is because skype use propietary coding scheemes, which means ethereal cant see whats the payload, If you make the test with a sip based client you can. I have my own server with asterisk, which uses UDP/RTP

Cheers

flo >

didn't know it would do that. I just tried it with both a