[Freeswitch-users] How to get the network address of registered Endpoint (NAT Problem)

Francis sms at icefire.qza.net.au
Wed Jan 8 00:58:53 MSK 2014


Even if the router is loopback enabled? I'm testing something like this:

Freeswitch(lan)<----------->Router-NAT(Internet)Loopback<---------->Phone(lan)

Setting the phone to STUN/UDP fails with "port unreachable", the phone 
can call but not be called, whereas STUN/TCP works- Freeswitch ignores 
the public IP sent by the phone and uses the router's internal IP instead.
This scenario is obviously different from the OP, but is typical of a 
roaming user switching between wlan and GSM. With the default internal 
profile and external_sip/rtp_ip set to STUN, it works quite well. Quite 
useless though if the phone can't use TCP. I'll be doing some tests from 
another NAT'd network later today to test this further....


On 8/01/2014 4:04 AM, Steven Ayre wrote:
> 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 at icefire.qza.net.au 
> <mailto:sms at 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 at 1.2.3.4@internal> sofia status profile internal reg
>     >
>     > Registrations:
>     >
>     =================================================================================================
>     >
>     > Call-ID:  aeac7f28-a6a0-4ba1-9d7c-30934924726d
>     > User: 270 at 1.2.3.4 <mailto:270 at 1.2.3.4>
>     > Contact:        "user" <sip:270 at 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 at 1.2.3.4 <mailto:270 at 1.2.3.4>
>     >
>     > Other devices in the same Setup have their NAT IP [+Port]
>     > (10.11.12.13:52767 <http://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 <http://10.11.12.13:52767> the answer from
>     10.11.12.13 is discarded.
>     >
>     > Normally I use bridge with user/270 at 1.2.3.4 <mailto:270 at 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 at 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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