[Freeswitch-users] Problems about STUN and one way audio
alex
xhyuer at gmail.com
Wed Dec 25 04:59:49 MSK 2013
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!
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20131225/7a1ce537/attachment.html
Join us at ClueCon 2013 Aug 6-8, 2013
More information about the FreeSWITCH-users
mailing list