STUN could work but would have problems in this scenario since the FS server is also on a RFC1918 IP, not a public address. You'd need to be using a STUN server running on the same network segment as FS to find the IP:port on that side of the NAT router for it to work. On 7 January 2014 11:06, Francis <sms@icefire.qza.net.au> wrote:
1. Have you turned on STUN on the phone? 2. Does the phone support any kind of NAT keepalive, say 20 seconds or so? 3. Does the phone support RPORT? 4. Is there a firewall on the freeswitch server, and if so, are the appropriate SIP and RTP ports open to new traffic? 5. Does the router have SIP ALG disabled? (yes is a good answer :)
One more question- is the freeswitch server behind a nat as well, or does it have it's own public IP address?
On 7/01/2014 6:38 PM, Mitja wrote:
Hello there,
Im trying to solve NAT related Problem which is caused by missbehaving SIP phone. Up Front: I searched the wiki and the web for all the NAT related sites/mailling list. I think its a error of the invoxia IPhone docking station (the missbehaving SIP phone), thus I did not open a jira for it.
We have a freeswitch server with internet accessibility and a Phone behind a natting router (Fritz!Box). FreeSWITCH (1.2.3.4) <-> (10.11.12.13) NAT (192.168.0.1) <-> (192.168.0.2) SIP Phone
The Phone is able to register and initiate calls. But when someone else to call that endpoint, its unreachable.
I did some research and the problem seems to be that the phone registers with a Contact containing the private IP Adress (192.168.0.2).
freeswitch@1.2.3.4@internal> sofia status profile internal reg
Registrations:
=================================================================================================
Call-ID: aeac7f28-a6a0-4ba1-9d7c-30934924726d User: 270@1.2.3.4 Contact: "user" <sip:270@192.168.0.2:52767;ob> Agent: invoxia-lemonvoice-v6.17.4 Status: Registered(UDP)(unknown) EXP(2014-01-06 13:06:35) EXPSECS(134) Host: ippbx-border IP: 10.11.12.13 Port: 52767 Auth-User: 270 Auth-Realm: 1.2.3.4 MWI-Account: 270@1.2.3.4
Other devices in the same Setup have their NAT IP [+Port] (10.11.12.13:52767) in their Contact Header which then works fine. In this setup however the FreeSWITCH tries to send the INVITE Message to 192.168.0.2 and even when a NAT Table on the firewall manages to map it to 10.11.12.13:52767 the answer from 10.11.12.13 is discarded.
Normally I use bridge with user/270@1.2.3.4 with the dialstring "{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(internal/${dialed_user}@ ${dialed_domain})}".
I tried to force freeswitch to use the NAT IP address by calling bridge with "sofia/internal/sip:270@10.11.12.13:52767;ob" which worked fine. The IP address is of course not static but if I would be able to extract the IP information which is stored in sofia registry (see above) I could arrange my dialplan in a way that would fix that phones missbehaviour.
So after all that describing text (sorry for that), heres my question: Is there a way (an API or Dialplan command) to get the IP Information which is stored the sofia registry.
Thanks in advance
Regards, Mitja
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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