[asterisk-dev] Asterisk not ACKing some 407 Proxy Auth Required requests?
Bryan Boatright
asterisk at omega71.com
Fri Jul 24 07:26:23 CDT 2009
I posted this to asterisk-users about a week ago and didn't get any kind
of response. Since then I've found this bug:
https://issues.asterisk.org/view.php?id=14239
This bug seems to describe my situation, except that the fix for it is
on the "other" side of the dialog but I seem to have evidence that the
dropped ACK in question in that bug is not simply lost by the network
but isn't actually sent by Asterisk (is that acceptable?). I've shown
my original post to asterisk-users below. I guess what I am looking for
is an explanation for why my server dropped the ACK (and if that is
acceptable to do from a protocol perspective) and whether you agree that
my provider should apply the patch in bug 14239 (when I asked that
question of the provider, it said "no").
I have a problem that has developed within about the past 3 months with
my backup outgoing SIP provider (I am not sure when this problem started
since it involves only my backup provider which is used rarely).
The problem is that most (not all) outgoing calls fail during the
earliest stages of call setup, specifically after the provider sends
back a "407 Proxy Auth Required" response. The problem is that my
server is failing to ACK this response most of the time (as determined
my looking at a Wireshark capture as summarized below). When my server
sends the second INVITE (with credentials) the provider sends back a
"491 Request Pending" which causes my server to terminate the call with
BYE. I've verified this behavior with Asterisk 1.4.23 through 1.4.25.1
which goes back to the time when calls placed with this provider were
working reliably. Again, I only have this issue with one provider for
some reason (not that I am using very many) and that provider says it
has no idea why this is happening.
Shown below are two calls placed via the same provider to the same
destination within 10 seconds of each other. The first call fails with
my server failing to send the ACK to the 407, but the second call
succeeds with my server sending the ACK to the 407. These calls were
placed with me hitting the redial key on the phone until I got one to go
through (less than a 10% success rate).
(In case the list software munges this report, you can view it here:
http://www.omega71.com/temp/vitelity_problem_bad_good_example_edited.txt )
This trace was generated by wireshark and edited by me to remove
identifying information, but other than that it should be considered
complete and accurate. Does anyone have any idea what might be wrong here?
FAILED CALL (no ACK to 407 response):
Conv.| Time | MY_HOST | 64.2.142.215.GIGe-net.vitel.net|
6 |186.532 | INVITE SDP ( telephone-event) |SIP From: sip:ME at MY_IP_ADDR <http://lists.digium.com/mailman/listinfo/asterisk-users> To:sip:15555551212 at outbound.vitelity.net <http://lists.digium.com/mailman/listinfo/asterisk-users>
| |(5060) ------------------> (5060) |
6 |186.575 | 407 Proxy Authentication Required |SIP Status
| |(5060) <------------------ (5060) |
6 |186.780 | INVITE SDP ( telephone-event) |SIP From: sip:ME at MY_IP_ADDR <http://lists.digium.com/mailman/listinfo/asterisk-users> To:sip:15555551212 at outbound.vitelity.net <http://lists.digium.com/mailman/listinfo/asterisk-users>
| |(5060) ------------------> (5060) |
6 |186.825 | 491 Request Pending |SIP Status
| |(5060) <------------------ (5060) |
6 |186.825 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
6 |186.826 | BYE | |SIP Request
| |(5060) ------------------> (5060) |
6 |186.872 | 487 Request Terminated |SIP Status
| |(5060) <------------------ (5060) |
6 |186.873 | 200 OK | |SIP Status
| |(5060) <------------------ (5060) |
SUCCESSFUL CALL (ACK to 407 response):
---------------------------------------------------------
7 |193.318 | INVITE SDP ( telephone-event) |SIP From: sip:ME at MY_IP_ADDR <http://lists.digium.com/mailman/listinfo/asterisk-users> To:sip:15555551212 at outbound.vitelity.net <http://lists.digium.com/mailman/listinfo/asterisk-users>
| |(5060) ------------------> (5060) |
7 |193.361 | 407 Proxy Authentication Required |SIP Status
| |(5060) <------------------ (5060) |
7 |193.361 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
7 |193.361 | INVITE SDP ( telephone-event) |SIP From: sip:ME at MY_IP_ADDR <http://lists.digium.com/mailman/listinfo/asterisk-users> To:sip:15555551212 at outbound.vitelity.net <http://lists.digium.com/mailman/listinfo/asterisk-users>
| |(5060) ------------------> (5060) |
7 |193.414 | 100 Trying| |SIP Status
| |(5060) <------------------ (5060) |
7 |195.529 | 183 Session Progress SDP ( telephone-event) |SIP Status
| |(5060) <------------------ (5060) |
7 |195.536 | RTP (g711U) |RTP Num packets:969 Duration:19.360s SSRC:0x74A81A70
| |(10024) <------------------ (12350) |
7 |195.689 | RTP (g711U) |RTP Num packets:959 Duration:19.220s SSRC:0x5614CA99
| |(10024) ------------------> (12350) |
7 |214.918 | RTP (g711U) |RTP Num packets:626 Duration:12.497s SSRC:0x77780128
| |(10024) <------------------ (12350) |
7 |214.920 | 200 OK SDP ( telephone-event) |SIP Status
| |(5060) <------------------ (5060) |
7 |214.920 | ACK | |SIP Request
| |(5060) ------------------> (5060) |
7 |214.929 | RTP (g711U) |RTP Num packets:586 Duration:11.780s SSRC:0x39936132
| |(10024) ------------------> (12350) |
7 |227.385 | BYE | |SIP Request
| |(5060) ------------------> (5060) |
7 |227.425 | 200 OK | |SIP Status
| |(5060) <------------------ (5060) |
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