[Freeswitch-users] firewall and nat

Peter P GMX Prometheus001 at gmx.net
Wed Jan 7 03:39:45 PST 2009

Generally speaking you will need to open an UPD port range for the RTP
stream. This can be configured on FS. Eg. we use 12000-13000 on our system.
Then If you do not hear any sound you may put

    <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>

in your external and internal profile, if FS is natted.

Best regards

kriko schrieb:
> Hello!
> Yesterday I've successfully placed a call between two different domains:
> originate sofia/default/1003 at &bridge(sofia/gateway/
> I didn't hear any audio, but it was kinda working. Today I investigated this more deep and found some issues.
> FS with is a public computer with firewall, but open TCP and UDP 5060, 5080 ports. Freeswitch on this machine
> uses default configuration.
> FS with is a lan computer in a different place, this is where I would like to start a call, the other way would
> be probably too much difficult for now. I've added a gateway entry to this one:
> http://pastebin.com/m2174ead
> Calling from (for e.g. using softphone at ext. 1003) to (ext. 1001 for e.g.) works. Both end
> answers, however I cannot hear audio coming trough. When testing I'm at the computer which is behind a lan, so I'm
> capturing music as audio source on the other side.
> Are there any other ports I should open on public computer?
> With wireshark on the computer behind a lan, I can see RTP going away to, but not the other way.
> There are also issues when e.g. terminating a call on public computer, fs on the other end will never terminate the call since
> SIP messages cannot reach the computer behind lan I guess, but this is second problem.

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