Thanks for all suggestions. Ufortunately I cannot get it working. Seems like packets are not coming to phone behind nat (freeswitch is on public ip). When registering I can see multiple notify retries like this: send 802 bytes to udp/[10.99.10.6]:5060 at 10:49:31.762605: ------------------------------------------------------------------------ NOTIFY sip:1003@10.99.10.6;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 212.235.180.41:5080;rport;branch=z9hG4bKtNStS2gtr8DNr Max-Forwards: 70 From: <sip:1003@212.235.180.41>;tag=veSr4DmgmFHjr To: <sip:1003@212.235.180.41> Call-ID: cec2b00b-5814-122c-f981-000fea488302 CSeq: 109587536 NOTIFY Contact: <sip:mod_sofia@212.235.180.41:5080> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-10924M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 93 Messages-Waiting: yes Message-Account: sip:1003@212.235.180.41 Voice-Message: 3/0 (0/0) I've opened necessary ports and I've defined custom rtp port range (which goes trough). Does nat should really just work if you register on external profile via port 5080? This is what I'm doing now. The phone on lan is a nokia N95 configured like described here (using port 5080): http://wiki.freeswitch.org/wiki/Nokia_N95 Phone shows registered message, but it takes like a half minute to register, when I'm on home network this happens in a second. On Wed, 07 Jan 2009 12:39:45 +0100, Peter P GMX <Prometheus001@gmx.net> wrote:
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.
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-- kriko