[Asterisk-Users] 128 kbs satelite link
scott at evtmedia.com
Wed Dec 17 10:06:03 MST 2003
Regarding "acceptible" latency, I remember reading a survey a year or two
ago, where telephone users were intentionally subjected to latency with
various delays. the results were interesting:
(a) When just latency alone was considered, most users thought that a
latency above 250 msec was "somewhat annoying", and above 400 (I think) was
(b) Then the test was re-run, factoring in the cost of the call. For
example, "would this call quality be acceptible if the call cost was reduced
by 50%" ...etc.... Then the 250 became "acceptible" and the 400 became
"annoying, but usable".
I'm paraphrasing (and recalling the details from memory), but I think the
point is that users will accept a higher latency if the call cost is
significantly lower, which is the case with VoiP in general.
For me, however, maybe I'm picky but a latency above 250 msec I find quite
annoying, and would rather use email to communicate!
Scott M. Stingel
Emerging Voice Technology Inc.
London, England and Palo Alto, California
> -----Original Message-----
> From: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] On Behalf Of
> Steven Critchfield
> Sent: Wednesday, December 17, 2003 3:58 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] 128 kbs satelite link
> On Wed, 2003-12-17 at 08:48, Senad Jordanovic wrote:
> > Hi all,
> > Anyone has experience using * through
> > 128 kbs (or bigger) satelite link?
> > In particular I am interested to hear how many calls could be put
> > through 128Kbs satelite link simultaneously?
> 128k may seem like the bottleneck, but it may not be the true limiting
> factor for you. Ping some machine near or for sure the
> machine you will
> be calling and see what kind of delay you will be dealing
> with on top of
> the packetizing of the audio. Basically add at least 20ms to the ping
> time to get an idea of the base line lag you will have.
> To give you an idea, my 14 hop trip to the gateway I was using is
> between 55-77ms on ping time, add the 20 ms packet size and your get
> 75-97ms base line lag. If you are using a jitter buffer, that adds a
> little more to the time so that it can deal with drop packets.
> I think I heard that satalite first hops are usually in the
> 400ms range.
> Are you prepared for half second or more delays?
> As far as call numbers, 1 for sure any codec if the only part
> that is a
> bottleneck is your link. 2 calls with occasional drop outs with almost
> any codec but ulaw or alaw. 3 calls will be pushing it with just about
> any codec. I could occasionally handle 3 calls with GSM on my
> 256k cable
> modem before call quality was significantly impaired.
> Steven Critchfield <critch at basesys.com>
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
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