| Summary: | ASTERISK-18659: If connection address in SDP content equals "sent-by" address of the Via header, then send RTP media to same address as SIP responses | ||
| Reporter: | Jack Bates (jablko) | Labels: | |
| Date Opened: | 2011-10-03 16:32:15 | Date Closed: | 2011-11-21 14:18:13.000-0600 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
| Versions: | SVN | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Attachments: | ( 0) 18659.patch | |
| Description: | In response to SIP INVITE where the connection address in SDP content equals the "sent-by" address of the Via header, I propose that if "nat = yes" is configured, then send RTP media to the same address as SIP responses (and continue to send RTP media to the media port in SDP content) e.g. in response to SIP INVITE: * transport source address 184.66.112.45 and port 44859 * "sent-by" address of the Via header 192.168.1.6 and port 44859 * connection address in SDP content 192.168.1.6 and media port 45112 - connection address in SDP content and "sent-by" address of the Via header are both 192.168.1.6, so if "nat = yes" is configured, then send RTP media to the same address as SIP responses: 184.66.112.45 (and continue to send RTP media to the media port in SDP content: 45112) This works around the case that the client is behind a NAT or firewall, and Asterisk is also behind a NAT or firewall, in which case "connection oriented media" or "comedia" requires manually mucking with the firewall | ||
| Comments: | By: Jack Bates (jablko) 2011-10-03 16:37:19.355-0500 I tested this patch with Android SIP client behind a NAT, and Asterisk running on Amazon EC2, behind Amazon EC2 NAT I am keen to make any changes that you suggest! By: Leif Madsen (lmadsen) 2011-10-04 10:29:21.935-0500 This sounds like something you should bring up on the asterisk-dev mailing list for discussion. If people are in favour of this approach, then I would suggest the step after that to be posting it to https://reviewboard.asterisk.org Thanks for the submission! By: Leif Madsen (lmadsen) 2011-11-01 14:30:56.406-0500 http://lists.digium.com/pipermail/asterisk-dev/2011-October/051848.html <-- started by Kevin Fleming By: Michael Spiceland (mspiceland) 2011-11-04 15:38:26.341-0500 This is not related to the security issue that Leif added above (confirmed by Mr. Kevin Fleming himself). By: Michael Spiceland (mspiceland) 2011-11-04 15:41:04.761-0500 Can you take this issue to the list? It needs more discussion. By: Leif Madsen (lmadsen) 2011-11-21 14:18:06.992-0600 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||