[Freeswitch-users] Problems about STUN and one way audio

[Sean Devoy]

I had similar issues with Polycom phones that do not support RPORT.  The fix for me was adding these to the directory entry:

<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
<param name="NDLB-force-rport" value="safe"/>

Sean

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Francis
Sent: Thursday, December 26, 2013 6:44 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Problems about STUN and one way audio

Have you set STUN in the clients?

IIRC, CSipSimple doesn't do Rport, at least it's not an option in the settings, so you also must make sure your firewall opens any rtp ports that FreeSwitch uses.

On 27/12/13 03:56, Anthony Minessale wrote:
> 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 at gmail.com 
> <mailto:xhyuer at 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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> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
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_________________________________________________________________________
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