| Summary: | ASTERISK-12684: r140417 broke sip nat interoperability | ||
| Reporter: | mdu113 (mdu113) | Labels: | |
| Date Opened: | 2008-09-04 15:35:48 | Date Closed: | 2008-09-05 10:51:10 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) 13424.patch ( 1) sip_trace.txt | |
| Description: | The scenario is very simple. I have 2 sip endpoints: xyz011101 and xyz010001, both on a private LAN (192.168.0.0/24) behind the same nat device (cisco router) and asterisk on a public ip. This setup has been working for years, but after changes made in r140417, asterisk seems to ignore the source port the device was registered from and always sends requests to registration_ip:5060, which of course doesn't work. ****** ADDITIONAL INFORMATION ****** Please see attached sip_trace.txt. As you can see xyz010001 is registered from 66.114.80.25:1054, but asterisk sends INVITEs to 66.114.80.25:5060 If I revert that commit (140417) then everything's back to normal. | ||
| Comments: | By: Russell Bryant (russell) 2008-09-05 07:18:18 FYI, since this change was done by you. By: Mark Michelson (mmichelson) 2008-09-05 09:43:56 Ah yes, thanks for pointing to the revision. I see where my error in logic is. I'll make a patch to test ASAP. By: Mark Michelson (mmichelson) 2008-09-05 09:52:13 I've uploaded 13424.patch for testing. The idea behind the commit which broke things for you was that if someone specifies a port when dialing, we should never ignore it. The problem is that I also made the inverse happen. If no port is specified when dialing a peer, then dial the standard SIP port, 5060. The patch I have uploaded changes the behavior to be exactly what it should be. If a port is specified when dialing a peer, use that port. If no port is specified, then use the port the peer registered from. By: mdu113 (mdu113) 2008-09-05 10:45:54 Yes, that patch fixes the problem. Thanks. By: Mark Michelson (mmichelson) 2008-09-05 10:46:52 Excellent. I will get this committed then. By: Digium Subversion (svnbot) 2008-09-05 10:51:07 Repository: asterisk Revision: 141217 U branches/1.4/channels/chan_sip.c ------------------------------------------------------------------------ r141217 | mmichelson | 2008-09-05 10:51:07 -0500 (Fri, 05 Sep 2008) | 14 lines Commit 140417 had a logic flaw in it which caused port 5060 to always be used when dialing a peer if no explicit port was specified. This broke the behavior of implicitly using the port from which the peer registered if no port is specified. This commit fixes the logic flaw. (closes issue ASTERISK-12684) Reported by: mdu113 Patches: 13424.patch uploaded by putnopvut (license 60) Tested by: mdu113 ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=141217 | ||