|Summary:||ASTERISK-10525: NAT settings ignored on calls recieved to [general]|
|Reporter:||John Todd (jtodd)||Labels:|
|Date Opened:||2007-10-14 13:25:33||Date Closed:||2011-06-07 14:08:04|
|Environment:||Attachments:||( 0) bad-sip-rtp-path.tcap|
|Description:||I have "nat=yes" set in the [general] section of sip.conf, but RTP media continues to go (on the *->UA flow) to the "theoretical address" instead of the "recieved address" on calls received from anonymous hosts to my * server. |
****** ADDITIONAL INFORMATION ******
If anyone _really_ wants SIP headers and a media flow, I can provide it. It's pretty easy to reproduce, though.
|Comments:||By: Joshua C. Colp (jcolp) 2007-10-15 08:38:13|
Yeah, I would like to see the SIP debug and an rtp debug. I just tested it between my home, through NAT, to my public Asterisk running latest trunk with no matching user/pass and it worked fine. I did, however, remember that rizzo did some NAT stuff in SIP that depended on the signalling awhile back though... maybe that is it.
By: John Todd (jtodd) 2007-10-16 22:49:33
SIP tcap file and RTP debug included. I can't provide a SIP debug on Asterisk at the moment, since the tester is not available (this isn't my client I'm testing.)
To sift through that tcap file, just look for SIP packets going to 22.214.171.124 and then RTP going to 10.211.55.4.
By: Olle Johansson (oej) 2007-11-19 09:41:23.000-0600
Is 1.4 any different? Any chance to test?
By: Joshua C. Colp (jcolp) 2008-01-15 20:58:03.000-0600
The code that I believe caused this has now been disabled for now.