In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-committed protocols of the public switched telephone network (PSTN).
eQSO is a "client/server" implementation of this technology. Essentially, there is an eQSO server which we host o要 our bandwidth (its domain is Server.eQSO.net) and which "users" (meaning here either eQSO RF Gateways or PC users) connect to from their computers via the internet.
The eQSO software was developed by Paul M0ZPD and was intended specifically for use by Radio Amateurs.RF gateways and PC users can be in different countries all over the World, and this makes it possible to talk to other users in other countries using a radio through your local gateway (assuming you're within range). This has created a new dimension to amateur radio and we hope that it will contribute to the continuing uptake of the service.
The system requires approximately 15kbs (kilo-bytes per second) per audio stream. As a user starts talking, either the gateway he is working through, or the PC Client he is connected through, sends an audio stream to the server. The server then relays by seperate streams the audio to each other client connected to the room.That's makes a system like this bandwidth intensive (if there's 10 people in the room, that requires a constant bandwidth of 150kbs) It's the upstream bandwidth that does most of the work as o要ly o要e user can talk at a time. So the talking user sends a 15kbs audio stream down to server which then has to send 'x' number of 15 kbs streams upstream to the listening users and gateways, 'x' being the number of people connected to the server less o要e (the person talking).
Lack of bandwidth can contribute to what is sometimes (confusingly) called "packet loss" which materialises as gaps in the audio stream. In practice, although you may sometimes hear pauses in someones audio, you won't actually have lost any of their "over", those packets containing the audio data have simply got delayed in their journey over the internet and not lost. Packet 'delay' would actually be a more accurate description.99% of the time the audio coming from a broadband connected user will not get interupted, although it has been noticed from time to time when the internet is exceptionally busy.Packet delay is usually noticeable when the internet is busy at peak times (evenings often between about 7pm and 9pm in the UK for example) and usually o要ly with users who are o要 dial-up modems.