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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/1126/">https://reviewboard.asterisk.org/r/1126/</a>
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<div>Review request for Asterisk Developers.</div>
<div>By Alec Davis.</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">in 1.8.2.3 and prior to trunk r306216, when the REFER transfer is complete then a deadlock occurs.
Reason being sip_set_rtp_peer locks pvt, then pbx_builtin_getvar_helper tries to lock chan
Violating locking order.
=== Thread ID: -1292625008 (do_monitor started at [24470] chan_sip.c restart_monitor())
=== ---> Lock #0 (chan_sip.c): MUTEX 23964 handle_request_do &netlock 0xb6796e80 (1)
=== ---> Lock #1 (channel.c): MUTEX 6211 ast_do_masquerade channels 0x8d4e0c8 (1)
=== ---> Lock #2 (channel.c): MUTEX 6214 ast_do_masquerade original 0xbd98f48 (1)
=== ---> Lock #3 (channel.c): MUTEX 6234 ast_do_masquerade clonechan 0xb24bf7d0 (1)
=== ---> Waiting for Lock #4 (chan_sip.c): MUTEX 6164 sip_fixup p 0xb24bab10 (1)
=== --- ---> Locked Here: chan_sip.c line 27632 (sip_set_rtp_peer)
=== -------------------------------------------------------------------
===
=== Thread ID: -1315861616 (pbx_thread started at [ 5035] pbx.c ast_pbx_start())
=== ---> Lock #0 (chan_sip.c): MUTEX 27632 sip_set_rtp_peer p 0xb24bab10 (1)
=== ---> Waiting for Lock #1 (pbx.c): MUTEX 9467 pbx_builtin_getvar_helper chan 0xb24bf7d0 (1)
=== --- ---> Locked Here: channel.c line 6234 (ast_do_masquerade)
multiple other bug reports reveal the sip_set_rtp_peer pvt locking issue.
Solution:
inc pvt refcount to prevent the possibility of it disappearing while unlocked.
unlock the pvt
call transmit_reinvite_with_sdp (which finishes up invoking pbx_builtin_getvar_helper)
lock the pvt
dec pvt refcount
Applies to 1.8.2.3 and trunk. Both verified deadlocks.
Presumably 1.8 branch
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Testing: trunk
3 SIP phones.
A calls B, and B answers on line 1.
B puts A on hold by selecting line2.
B calls C, and C answers.
B initiates transfer of line1 to line2, phone uses REFER.
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/view.php?id=18837">18837</a>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>trunk/channels/chan_sip.c <span style="color: grey">(308720)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/1126/diff/" style="margin-left: 3em;">View Diff</a></p>
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