Help With IP Phone SIP/RTP

I am using PepWave MAX_BR1 attempting to set up an IP phone. I get the dial tone and can make a call but with no voice at all. I have set everything I know, any assistance will be greatly appreciated.

Welcome to the forum!
Are you using a 4G connection by any chance?

I would suggest you use speedfusion cloud and send all your VoIp traffic via that. What I expect you are seeing is a lack of inbound routing so the audio can’t get back to you via the cellular connection. Often outbound audio works and inbound audio fails in these situations.

SIP requires port 5060 UDP to establish the connection, and ports 10,000 - 20,000 UDP for the audio stream. Are you registering the phone directly to a VoIP provider, or to a server of your own?

You are correct I am not seeing any inbound traffic

Thanks I am using my own PBX
The UDP ports are connecting its the RTS 10020-20000 not connecting

And yes it’s 4G

In general as you know VoIP doesn’t enjoy NAT. Right now you have two NAT hops. You’re going from LAN to cellular WAN on your BR1 and then your going through Carrier Grade NAT ay your mobile network providers edge.

Instead, host a fusionhub alongside your VoIP PBX, or use Speedfusion Cloud to build a VPN out over 4G to the internet to give you a single NAT hop (LAN (via VPN) via FusionHub/Speedfusion Cloud NAT). Then your VoIP will likely come right.

Milkew, you might have the NAT problem decribed by @MartinLangmaid, but your are mixing up some terminology on the ports. I will explain assuming the PBX server is on your LAN. If you are using a hosted solution the execution is a bit different but the concept is the same.

Port 5060 UDP is used to make the connection. That is called SIP. Technically the industry can use 5060 to 5064 but I don’t know anyone who uses anything other than 5060. Your firewall must allow inbound and outbound traffic to/from the PBX server on that port. Since you said the connection is being made that is likely working now.

If your server is connecting to the VoIP provider by credentials (username/password) that portion is completed. If the VoIP provider is sending calls to your server by IP only authentication (provider sends calls to your IP with no authentication required) then you need to forward inbound 5060 to the LAN address of your server.

Ports 10,000 to 20,000 UDP are used to carry the audio stream. The industry calls this RTP but there isn’t any such thing as an RTP (or RTS) port. Its all UDP (as opposed to TCP which is more common for general internet traffic). Your firewall needs to have port range 10,000 to 20,000 open to your PBX server. One way audio is a common setup problem. If the other party can hear you, they are receiving those ports. If you can’t hear them, the inbound ports are being blocked.

If you are registering phones on a cloud server the concept is the same on the firewall. Port forwarding is typically not required for cloud setup because authentication is done with username/password.

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That sounds correct. This is my first attempt to use cellular for IP phone
I will build a VPN

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