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/1003@10.99.8.221 &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.