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</o:shapelayout></xml><![endif]--></head><body lang=ZH-CN link=blue vlink=purple style='text-justify-trim:punctuation'><div class=WordSection1><p class=MsoNormal><span lang=EN-US>Hi, all<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>I have a server that has installed the FreeSwitch software, and the server has a public IP addr.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>My clients are android phones, the CSipSimple softphone runs on them.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>There are some issues when these phone call each other.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>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.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Therefore, I guess I need set a STUN server to deal with the above problem. So I read related web pages, such as <a href="http://wiki.freeswitch.org/wiki/NAT_Traversal">http://wiki.freeswitch.org/wiki/NAT_Traversal</a>. But, How to configure the STUN server in FreeSwitch is ambiguous.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Then, I tried to make some configurations as below:<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>vars.xml.<o:p></o:p></span></p><p class=MsoNormal style='text-indent:21.0pt'><span lang=EN-US><X-PRE-PROCESS cmd="set" data="external_rtp_ip=a.b.c.d"/> <o:p></o:p></span></p><p class=MsoNormal style='text-indent:21.0pt'><span lang=EN-US><X-PRE-PROCESS cmd="set" data="external_sip_ip=a.b.c.d"/> <o:p></o:p></span></p><p class=MsoNormal style='text-indent:21.0pt'><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>internal.xml<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/><o:p></o:p></span></p><p class=MsoNormal style='text-indent:21.0pt'><span lang=EN-US><param name="ext-sip-ip" value="$${ external_sip_ip }"/><o:p></o:p></span></p><p class=MsoNormal style='text-indent:21.0pt'><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>external.xml<o:p></o:p></span></p><p class=MsoNormal style='text-indent:21.0pt'><span lang=EN-US><param name="manage-presence" value="true"/><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US> <param name="ext-rtp-ip" value="$${ external_rtp_ip }"/><o:p></o:p></span></p><p class=MsoNormal style='text-indent:21.0pt'><span lang=EN-US><param name="ext-sip-ip" value="$${ external_sip_ip }"/><o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>But, I didn’t find where could I set the port of STUN server. So I failed to launch the STUN server.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Thus, Could you tell me How to solve the one-way audio problem, and How to configure the STUN server in FreeSwitch?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Thanks and sorry if my English is poor.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US>Best regards!<o:p></o:p></span></p></div></body></html>