[asterisk-bugs] [Asterisk 0014584]: [patch] Asterisk does not stop retransmission
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jun 18 09:33:30 CDT 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=14584
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Reported By: klaus3000
Assigned To: mmichelson
======================================================================
Project: Asterisk
Issue ID: 14584
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: minor
Priority: normal
Status: ready for testing
Target Version: 1.4.27
Asterisk Version: 1.4.23
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-03-02 09:22 CST
Last Modified: 2009-06-18 09:33 CDT
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Summary: [patch] Asterisk does not stop retransmission
Description:
Scenario:
<pre>
client Asterisk
------INV,407,ACK-------->
------INV---------------->
-----INV---->
<---183------
<-------183---------------
<---180------
<-------180---------------
--------CANCEL----------->
X----487, 200-(loss)----
---CANCEL retransmission->
<----487, 200-------------
---ACK------------------->
<----487------------------
<----487------------------
<----487------------------
<----487------------------
<----487------------------
</pre>
======================================================================
----------------------------------------------------------------------
(0106626) klaus3000 (reporter) - 2009-06-18 09:33
https://issues.asterisk.org/view.php?id=14584#c106626
----------------------------------------------------------------------
Hi!
With this patch Asterisk 1.4.24 crashes on reception of the retransmitted
CANCEL:
<--- Transmitting (no NAT) to 83.136.32.132:5061 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
83.136.32.132:5061;branch=z9hG4bK-10506-1-4;received=83.136.32.132
From: <sip:klaus at 83.136.32.165>;tag=1
To: <sip:01505641636 at 83.136.32.165:5060>;tag=as5434e245
Call-ID: 1-10506 at 83.136.32.132
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:01505641636 at 83.136.32.165>
Content-Length: 0
<------------>
<--- SIP read from 83.136.32.132:5061 --->
CANCEL sip:01505641636 at 83.136.32.165:5060 SIP/2.0
Via: SIP/2.0/UDP 83.136.32.132:5061;branch=z9hG4bK-10506-1-8
From: <sip:klaus at 83.136.32.165>;tag=1
To: <sip:01505641636 at 83.136.32.165:5060>
Call-ID: 1-10506 at 83.136.32.132
CSeq: 2 CANCEL
Max-Forwards: 70
<------------->
--- (7 headers 0 lines) ---
Sending to 83.136.32.132 : 5061 (no NAT)
<--- Reliably Transmitting (no NAT) to 83.136.32.132:5061 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
83.136.32.132:5061;branch=z9hG4bK-10506-1-4;received=83.136.32.132
From: <sip:klaus at 83.136.32.165>;tag=1
To: <sip:01505641636 at 83.136.32.165:5060>;tag=as5434e245
Call-ID: 1-10506 at 83.136.32.132
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
[Jun 18 16:32:23] NOTICE[23306]: chan_sip.c:2067 __sip_reliable_xmit: Hey,
I just set the response code for this packet to 487
<--- Transmitting (no NAT) to 83.136.32.132:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
83.136.32.132:5061;branch=z9hG4bK-10506-1-8;received=83.136.32.132
From: <sip:klaus at 83.136.32.165>;tag=1
To: <sip:01505641636 at 83.136.32.165:5060>;tag=as5434e245
Call-ID: 1-10506 at 83.136.32.132
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
-- Hungup 'DAHDI/1-1'
== Spawn extension (direct, 01505641636, 9) exited non-zero on
'SIP/klaus-08fb17f8'
<--- SIP read from 83.136.32.132:5061 --->
CANCEL sip:01505641636 at 83.136.32.165:5060 SIP/2.0
Via: SIP/2.0/UDP 83.136.32.132:5061;branch=z9hG4bK-10506-1-12
From: <sip:klaus at 83.136.32.165>;tag=1
To: <sip:01505641636 at 83.136.32.165:5060>
Call-ID: 1-10506 at 83.136.32.132
CSeq: 2 CANCEL
Max-Forwards: 70
<------------->
--- (7 headers 0 lines) ---
Sending to 83.136.32.132 : 5061 (no NAT)
Scheduling destruction of SIP dialog '1-10506 at 83.136.32.132' in 32000 ms
(Method: CANCEL)
Segmentation fault (core dumped)
https://issues.asterisk.org/view.php?id=0 0xb61b7cde in handle_request_cancel
(p=0x8fb17f8, req=0xb6170050) at
chan_sip.c:15366
15366 prev_pkt->next = pkt->next;
(gdb) bt
https://issues.asterisk.org/view.php?id=0 0xb61b7cde in handle_request_cancel
(p=0x8fb17f8, req=0xb6170050) at
chan_sip.c:15366
https://issues.asterisk.org/view.php?id=1 0xb61bb2e2 in handle_request
(p=0x8fb17f8, req=0xb6170050,
sin=0xb6170040, recount=0xb6170034, nounlock=0xb6170038) at
chan_sip.c:16050
https://issues.asterisk.org/view.php?id=2 0xb61bc10f in sipsock_read
(id=0x8f7d7a0, fd=47, events=1, ignore=0x0)
at chan_sip.c:16293
https://issues.asterisk.org/view.php?id=3 0x080b0b54 in ast_io_wait
(ioc=0x8f70d00, howlong=999) at io.c:279
https://issues.asterisk.org/view.php?id=4 0xb61bcadc in do_monitor (data=0x0)
at chan_sip.c:16512
https://issues.asterisk.org/view.php?id=5 0x0810588d in dummy_start
(data=0x8f7d690) at utils.c:856
https://issues.asterisk.org/view.php?id=6 0xb7f1e4c0 in start_thread () from
/lib/i686/cmov/libpthread.so.0
https://issues.asterisk.org/view.php?id=7 0xb7e306de in clone () from
/lib/i686/cmov/libc.so.6
Issue History
Date Modified Username Field Change
======================================================================
2009-06-18 09:33 klaus3000 Note Added: 0106626
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