[asterisk-bugs] [Asterisk 0012101]: SIP channel hung due to CANCEL ReliableXmit (ReTX)
noreply at bugs.digium.com
noreply at bugs.digium.com
Fri May 30 15:07:34 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12101
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Reported By: MVF
Assigned To:
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Project: Asterisk
Issue ID: 12101
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: confirmed
Asterisk Version: 1.4.18
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 02-28-2008 13:23 CST
Last Modified: 05-30-2008 15:07 CDT
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Summary: SIP channel hung due to CANCEL ReliableXmit (ReTX)
Description:
I've detected a problem that make asterisk keep sip channels up forever
even when the call has been disconnected, the problem keep UDP ports and
memory taken making asterisk to drop new calls when the sip channels hung
increase.
Reviewing the "sip show history" of one of these hung channels I see that
the call is cancel by calling party while is on ringing state, after that
asterisk send CANCEL to the peer but it didn't get an answer (maybe because
of network problems). After I get some bad responses from peer, maybe also
due to network problems that asterisk reply with an ACK but the history
shows a new ReliableXmit timeout message every 30 seconds.
(originating peer)
* SIP Call
1. TxReqRel INVITE / 102 INVITE - -UNKNOWN-
2. Rx SIP/2.0 / 102 INVITE / 100 Trying
3. Rx SIP/2.0 / 102 INVITE / 100 Trying
4. Rx SIP/2.0 / 102 INVITE / 180 Ringing
5. Rx SIP/2.0 / 102 INVITE / 180 Ringing
6. Cancel Cause Normal Clearing
7. SchedDestroy 32000 ms
8. TxReqRel CANCEL / 102 CANCEL - -UNKNOWN-
9. SchedDestroy 32000 ms
10. ReTx 1000 CANCEL sip:17898702397 at 20.20.20.25 SIP/2.0
11. ReTx 2000 CANCEL sip:17898702397 at 20.20.20.25 SIP/2.0
12. Rx SIP/2.0 / 102 CANCEL / 100 Trying
13. ReliableXmit timeout
14. Rx SIP/2.0 / 102 INVITE / 180 Ringing
15. Rx SIP/2.0 / 102 INVITE / 486 Busy Here
16. TxReq ACK / 102 ACK - -UNKNOWN-
17. ReliableXmit timeout
18. ReliableXmit timeout
19. ReliableXmit timeout
20. ReliableXmit timeout
21. ReliableXmit timeout
22. ReliableXmit timeout
23. ReliableXmit timeout
24. ReliableXmit timeout
25. ReliableXmit timeout
26. ReliableXmit timeout
27. ReliableXmit timeout
28. ReliableXmit timeout
29. ReliableXmit timeout
30. ReliableXmit timeout
31. ReliableXmit timeout
32. ReliableXmit timeout
33. ReliableXmit timeout
34. ReliableXmit timeout
35. ReliableXmit timeout
36. ReliableXmit timeout
37. ReliableXmit timeout
38. ReliableXmit timeout
39. ReliableXmit timeout
40. ReliableXmit timeout
41. ReliableXmit timeout
42. ReliableXmit timeout
43. ReliableXmit timeout
44. ReliableXmit timeout
45. ReliableXmit timeout
46. ReliableXmit timeout
47. ReliableXmit timeout
48. ReliableXmit timeout
49. ReliableXmit timeout
50. ReliableXmit timeout
I got a capture for the sip transaction messages that generate these hung
sip channels, please take a look at it in the attached file including two
captures (20080227_sip_channel_hung_at_cancel_ReTX.txt), note that there is
no sip messages sent after the last ACK, the "ReliableXmit timeout"
messages appear in the history of the call filling the 50 slot buffer and
even deleting the old history messages. The "sip show channels" command
shows the channel hung forever.
20.20.20.25 1789870239 185d051b629 00102/00000 0x0 (nothing) No
(d) Tx: ACK
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dd1916 - 05-30-08 15:07
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I tried the patch on our 1.4.17 but I still have several stuck channels
here is the sip show channel
* SIP Call
Curr. trans. direction: Outgoing
Call-ID:
94b138f0-a06000e-13c4-45026-11f95f-536b76fa-11f95f
Owner channel ID: <none>
Our Codec Capability: 4
Non-Codec Capability (DTMF): 1
Their Codec Capability: 4
Joint Codec Capability: 4
Format: 0x0 (nothing)
MaxCallBR: 384 kbps
Theoretical Address: 10.6.0.14:5060
Received Address: 10.6.0.14:5060
SIP Transfer mode: open
NAT Support: No
Audio IP: 172.16.100.101 (local)
Our Tag: as32f92656
Their Tag:
94b16bc8-a06000e-13c4-45026-11f95f-723aba38-11f95f
SIP User agent: ARRIS-TM402P release v.05.02.0X SN/0013113FC54F
Username: 4185558698
Peername: 4185558698
Original uri: sip:4185558698 at 10.6.0.14:5060
Caller-ID: 4185558698
Need Destroy: 2
Last Message: Rx: BYE
Promiscuous Redir: No
Route: sip:4185558698 at 10.6.0.14:5060
DTMF Mode: rfc2833
SIP Options: replaces replace 100rel timer join
here is the sip show history
1. Rx INVITE / 1 INVITE / sip:5550927 at here.com:5060
2. AuthChal Auth challenge sent for - nc 0
3. TxRespRel SIP/2.0 / 1 INVITE - 407 Proxy Authentication Required
4. SchedDestroy 32000 ms
5. Rx ACK / 1 ACK / sip:5550927 at here.com:5060
6. Rx INVITE / 2 INVITE / sip:5550927 at royaume.com:5060
7. CancelDestroy
8. Invite New call:
94b138f0-a06000e-13c4-45026-11f95f-536b76fa-11f95f
9. AuthOK Auth challenge succesful for 4185558698
10. NewChan Channel SIP/4185558698-0194b850 - from
94b138f0-a06000e-13c4-45
11. TxResp SIP/2.0 / 2 INVITE - 100 Trying
12. TxResp SIP/2.0 / 2 INVITE - 180 Ringing
13. TxRespRel SIP/2.0 / 2 INVITE - 200 OK
14. Rx ACK / 2 ACK / sip:5550927 at 172.16.100.101
15. ReInv Re-invite sent
16. TxReqRel INVITE / 102 INVITE - -UNKNOWN-
17. Rx SIP/2.0 / 102 INVITE / 100 Trying
18. Rx SIP/2.0 / 102 INVITE / 200 OK
19. TxReq ACK / 102 ACK - -UNKNOWN-
20. ReInv Re-invite sent
21. TxReqRel INVITE / 103 INVITE - -UNKNOWN-
22. Rx SIP/2.0 / 103 INVITE / 100 Trying
23. Rx BYE / 3 BYE / sip:5550927 at 172.16.100.101
24. RTCPaudio
Quality:ssrc=1371236592;themssrc=25523736;lp=0;rxjitter=0.01273
25. TxResp SIP/2.0 / 3 BYE - 200 OK
26. Hangup Cause Normal Clearing
Issue History
Date Modified Username Field Change
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05-30-08 15:07 dd1916 Note Added: 0087574
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