Hi, all I have a server that has installed the FreeSwitch software, and the server has a public IP addr. My clients are android phones, the CSipSimple softphone runs on them. There are some issues when these phone call each other. When phone A calls phone B, only the phone B can listen the audio coming from phone A, but the phone A cannot. What's more, the phone B always need some delays to listen the audio from phone A. Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as http://wiki.freeswitch.org/wiki/NAT_Traversal. But, How to configure the STUN server in FreeSwitch is ambiguous. Then, I tried to make some configurations as below: vars.xml. <X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/> internal.xml <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> external.xml <param name="manage-presence" value="true"/> <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/> <param name="ext-sip-ip" value="$${ external_sip_ip }"/> But, I didn't find where could I set the port of STUN server. So I failed to launch the STUN server. Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch? Thanks and sorry if my English is poor. Best regards!