Hi all- 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 "unacceptible". (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! Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. London, England and Palo Alto, California URL: www.evtmedia.com > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Steven Critchfield > Sent: Wednesday, December 17, 2003 3:58 PM > To: [EMAIL PROTECTED] > 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 <[EMAIL PROTECTED]> > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
