I have a VoIP application that I'm trying to run over the PPTP VPN tunnel on a RV120W router.
The system is a NEC SV8100 PBX communicating with the NEC soft phone (SP310). The system uses SIP to set up the call and for other signaling information. It uses RTP to transmit/receive the audio stream.
The problem I'm having is that there is no audio stream from the soft phone. The SIP communication and the audio stream to the soft phone works fine. The symptom is: from the soft phone, the remote party cannot hear you, but you can hear them.
I did a packet trace on the RV120W and found the following:
PT=ITU-T G.711 PCMU, SSRC=0x7F1621CA, Seq=14361, Time=779040
PT=ITU-T G.711 PCMU, SSRC=0xE943F2E7, Seq=19090, Time=3940936556
192.168.1.252 => NEC PBX
192.168.1.52 => Soft phone connected via PPTP
192.168.1.1 => RV120W
As you can see, the Source IP address is being changed from its original 192.168.1.52 to 192.168.1.1. The NEC PBX is expecting the packet to be coming from the softphone, (192.168.1.52) not the RV120W (192.168.1.1). As a result it ignores the RTP packet from the soft phone and does not relay it to the remote party.
Is there any reason why the RV120W is performing NAT on PPTP packets? Can this be disabled somehow?
Any ideas will be helpful.