I have a Grandstream GXW4232V2 gateway and I’m having an audio issue. I have a new Asterisk (20.6) / FreePBX 17 server and I’m using PJSIP. I’m experiencing one-way audio: VoIP gateway → server – audio works, server → VoIP gateway – no audio.
I monitored SIP, SDP, and RTP packets, and they do reach the gateway’s Ethernet interface (it even captures them in Packet Capture). However, there is still no audio. What I noticed is that in the SDP packet it specifies that the transmission will be on port 17980 on the server, but the server is sending packets from a different source port. Could this be the problem?
Of course, the ports change with every call. I don’t have NAT; the server and the gateway are in different VLANs.
There are no issues with codecs or with the connection between the VLAN networks.
The Wireshark capture taken on the gateway end shows that the GXW at 10.140.3.36 is requesting that Asterisk at 10.???.13.17 send RTP to port 50000. Subsequent RTP packets are sent from gateway port 50000 to Asterisk port 15356 (normal; the default Asterisk RTP port range is 10000-20000), but the RTP from Asterisk has source port 1316.
I am reasonably certain that Asterisk was sending from port 15356, but the intervening networking equipment (firewall, VPN, etc.) modified the source port. You can easily confirm this by taking a concurrent capture at the FreePBX end. One way to do this is to run tcpdump on the Freepbx machine and move the pcap file to your workstation to analyze with Wireshark.
I don’t believe that the GXW has an option to ignore the source port on incoming RTP, so you need to find the “randomize source port” or similar setting in your network gear and turn it off.
