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DISCLAIMER: I'm REALLY new to
FreeSwitch, so please take my advice with a grain of salt. I have a similar setup (and problem) - the wiki documentation refers to it as "double nat". Like you, my FS and client are behind different NATs and I can register my remote endpoint and make calls (in my case, to the the FS demo ivr at 5000). Since your external endpoint (spa3102) is registering, you've likely setup your sip profile correctly (ext-sip-ip, ext-rtp-ip, nat settings, etc). According to the documentation, and my limited experience, you have at least 2 options for NATted endpoints. However, I am unable to make the second work. 1) Setup stun on your remote endpoint (spa3102 in your case) 2) Add <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> to the directory xml file that describes your spa3102 endpoint Option 1 worked for me right away (eyebeam in my case) and, as expected, the remote sdp had the correct (remote) IP address, since the endpoint is using stun to correctly identify its IP address to FS. However, option 2 has not made a difference (for me). Is it just me or is it strange that SIP works without stun, but RTP doesn't? I guess I've been spoiled by the way Asterisk handles NAT and was hopeful that NDLB-connectile-dysfunction would behave similarly, so I wouldn't have to tell users to setup stun on their clients. Maybe a FS user with some experience with this type of NAT setup and these settings can help. I'd be interested in knowing how to correctly setup remote NATted endpoints without stun - or, at least, hear from someone that this setting works for them without stun. Anyway, hope this helps you with your SPA3102. Mark Campbell-Smith wrote: Hi! I'm sure this is a NAT issue, but I'm not sure what options to use.I have a Linksys SPA3102, NAT'd on the internet (remotely) and connected to my FS on the otherside of the world, which is also natted. A PAP2T is connected on the same subnet as the FS. The 3102 registers successfully and a call can be set up from the PAP2 to the 3102. However, after FS receives the Remote SDP the audio stops (ring tone stops in my case) 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3646 Channel sofia/internal/sip:[email protected]:56885 entering state [completing][200] 2009-12-20 22:29:57.848463 [DEBUG] sofia.c:3657 Remote SDP: v=0 o=- 18490612 18490612 IN IP4 192.168.1.3 s=- c=IN IP4 192.168.1.3 t=0 0 m=audio 16432 RTP/AVP 2 100 101 a=rtpmap:2 G726-32/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 I notice that the ip address in the o and c fields indicate a local IP address. Should this IP address be an external IP address of the 3102 instead? Thanks _______________________________________________ FreeSWITCH-users mailing list [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org |
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