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
Peter
kriko schrieb:
> Hello!
>
> Yesterday I've successfully placed a call between two different domains:
> originate sofia/default/[email protected]
> &bridge(sofia/gateway/212.235.180.41/1001)
>
> I didn't hear any audio, but it was kinda working. Today I investigated this
> more deep and found some issues.
> FS with 212.235.180.41 is a public computer with firewall, but open TCP and
> UDP 5060, 5080 ports. Freeswitch on this machine
> uses default configuration.
>
> FS with 10.99.8.221 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 10.99.8.221 (for e.g. using softphone at ext. 1003) to
> 212.235.180.41 (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
> 212.235.180.41, 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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