Generally, DSL and cable broadband connections will give the best results and should be favoured over 3G/4G, satellite, and other wireless 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 100ms. Voice quality will degrade with Jitter values above 20 ms (with calls being unfeasible with Jitter values above 30ms).
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.