If the FS is on a public IP you don't need to configure any of those params they are for dealing with FS itself behind nat.
Try running FS the default way and make all the changes on you client.  Sometimes disabling all the NAT features on the client actual can make it work better.



On Tue, Dec 24, 2013 at 7:59 PM, alex <xhyuer@gmail.com> wrote:

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!


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