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!