Hello Everyone, Does anyone here experiencing the “Auto changing RTP port” in freeswitch? It actually happened when initiating a call from local SIP going to SIP server installed in the CLOUD. There is no Audio both ends after the SIP call setup. Now, when initiating call from cloud going to local SIP I have no experience in Intermittent RTP. Btw, the device that is registered to the SIP server in the cloud is Polycom IP Phone. Basically, I have SIP server in cloud and a SIP server in local. The local SIP server is using VPN to tunnel the RTP because it is behind NAT. Instead of doing port forwarding. I have tried the following: 1. In the freeswitch, I configured the ”auto_adjust_rtp=true” 2. in the local freeswitch, I configured the internal and external profile to ext-rtp-ip=<VPN IP> If anyone who is experiencing this and able to solve the issue, please give me some point on what I need to check. Thank you. SL Sent from my iPhone