You need to know more than what you want to sell.
To make a substantial decision, you need to know something about the voice communication worldwide between telcos and gateways. If you do not care about this, may be, you buy a gateway with a wrong layout and you waste a lot of money.
Quality of speech (or voice)
If you did make calls over ISDN in Europe, you are spoiled with an extreemly good quality. Thats not the fact in many US areas. In Germany we have ISDN quality since many many years and we are well satisfied with the 99,9% usage (availability) of the 64kbit/s bandwidth.
If you often make overseas calls, in very rare cases you have a one way connection with a really worse quality like the old reel to reel taperecorders from 1950.
So, whats that ?
If speech ist transmitted over long distances for low money, its compressed to have many connections simultanously on the same line (or channel). Sometimes it is compressed so much like the space shuttle. Then you hear, it is cheap.
About delay time or latency
If you talk to your mother in law, you know, sometimes you should interrupt her, not to spend hundreds of dollars for one call. So usally you do not recognize, that on a long distance call there is a latency of (i.e.) 150 milliseconds from you to the other end (or more).
You need to understand, that even voice is transported from Europe to US or China with lightspeed. Not faster ! Thats impossible. So we from Germany have a smallest latency for IP data packets of 90ms to ths US east-coast and another 65ms to LA at the west-coast. If we test it inside Europe, its below 30ms. Now you have to add 2 x 30ms for coding the speech into IP packets.
So the experts say, the critical limit (where a phonecall becomes unacceptable) is about 200ms and more. Even 150ms is close to bad. The ITU recommends in G.114, that the latency should be lower than 150 ms for acceptable conversation quality.
Where is the latency comming from:
So we have the carriers lines (fiber optics) with its latency and the cutting of the speech and compressing in little packtes of 20ms. SOme DSP cpu´s have a low latency but bad compression rate of 15ms, others have a high compression rate with 35ms.
If you make a call from Frankfurt to New York, the line hat 95ms. And you get 30ms on both ends (compression and decompression), so you have a total of 155ms, close to the limit.
The telcos can do it better. If I call my partner and agent in Fairfield/CA with my wired phone, we must have less than 150ms, ist all clear and almost no delay.
There are some more reasons for processing delay like pre and postprocessing, echo cancellation, noise suppression and filtering.
About compression of speech
There are two ISDN or digital bandwidths on the world. Thats 56 kilobit/s in the US and Canada and 64 kilobit/s in the rest of the world.
And there are two methods of compressing data.
- the lossless compression and
- the non lossless compression.