Generally, DSL and cable broadband connections will give the best results and should be favoured over 3G/4G, satellite, and other wireless (wifi) connection types.
Available bandwidth (download and upload 'Speed') alone doesn't indicate a connection's suitability for VoIP calling. Latency, Jitter and Packet Loss are also very important.
A VoIP call (using the G.711 codec) will require circa 90 kbps both up and downstream.
So 10 simultaneous VoIP calls using the same G.711 ('PCMU/A') codec would require under 1Mbps upload and download bandwidth.
As a general rule of thumb, a VoIP call will tolerate latencies up to circa 80ms. Voice quality will degrade with Jitter values above 15 ms (with calls being entirely unfeasible with Jitter values above 25ms).
The best real-world test of your internet connection is to sign up for a free 30 day trial of sipgate team and make a test call with a free VoIP softphone like Zoiper:
You can also quickly and easily test your internet connection's quality on the following websites:
If you've limited bandwidth available, using only lower bandwidth codecs in your phone settings, and Quality of Service (QoS) rules for VoIP traffic in your router settings may provide a solution.
Please click here for a list of the codecs supported by sipgate.
A phone may be able to register online over EDGE and GPRS connections, but VoIP calls won't be possible.