Frequently Asked Question

There are two common reasons people only get one way audio, or sometimes no audio at all, the first, but least common, is codec incompatibility, check your chosen codec is allowed and supported by your VoIP service provider, if you set preferred to G711A (or A-Law) and second preference to G729, you're a fairly safe bet, your service provider will usually provide you with this information. If you're using our VoIP service, check out our Codecs KB entry.

The second, and most common problem is the devices behind NAT. People often think they only need to forward port 5060 to their device, but VoIP has two components, SIP and RTP.

5060 is the SIP (Session Initiated Protocol) signalling and control channel. When you dial or receive a call, port 5060 takes care of all the setup and finishing/cleaning up of the calls, it does not handle the audio, that's handled by a second protocol, RTP (Realtime Transfer Protocol). 

RTP has no permanent fixed port(s), most SIP providers use massive port blocks, like for example ports 10000 to 20000 on FreePBX, or 16384 to 32767 which is Cisco's favourite.

This means when modifying your routers NAT settings, in port forwarding, you need to forward both ports 5060 (SIP) and 10000-20000 (RTP) to your device, most VoIP providers will inform you when singing up the ports to use.

But now your thinking, wait! If I forward all those ports to my device is that not a massive security risk? Well, damn straight it is! Which is why when port forwarding, you need to create rules, rules that say forward ports 5060/10000:20000 only for your voip.providers.ip.address which you should have got in your welcome message, or from their websites support page, this way your device is only exposed for those ports, to your VoIP service provider.