SipWarpRTP and STUN
SIP on port 5060 only sets the call up. The voice itself travels as RTP, on a separate range of UDP ports. If the call rings and then has silence, or you hear them and they do not hear you, those audio ports are not getting through the router.
RTP min and RTP max
Set a range and, if audio fails, forward that same UDP range on the router to this computer.
- A safe range is min 10000 and max 20000.
- voip.ms documents 10001 through 20000. Use that if you are on voip.ms.
- Min has to be lower than max. A port cannot be higher than 65535. 100001 is not a port.
You do not have to forward the range if calls already have two-way audio. Forward it when they do not.
Why STUN helps
A home router hides this PC behind one public address. STUN is a small server on the internet that tells SipWarp what that public address is, so the other side knows where to send the audio.
Many SIP providers do not hand you a STUN name. Turn STUN on and pick one of these. They are public. If one does not answer, try the next.
| Server | Notes |
|---|---|
| stun.voip.ms | Use this if your account is voip.ms. |
| stun.l.google.com:19302 | Google. Widely reachable. The :19302 is the port. |
| stun1.l.google.com:19302 | A second Google server, if the first one fails. |
| stun.counterpath.net | A long-standing public server used by many softphones. |
| stun.sipgate.net:10000 | Sipgate. Note the port 10000, not 3478. |
STUN fixes the common home-router case. It cannot punch through every locked-down office network. If audio is still one-way after STUN, forward the RTP range.
rport should stay on. It tells the provider to reply to the port your router actually used. DNS SRV can stay off unless the provider gave you a nameserver and told you to use it.