[asterisk-dev] [Code Review] fix Deadlock with attended transfers of SIP calls

[Alec Davis]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1126/
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(Updated 2011-02-25 12:12:26.595347)


Review request for Asterisk Developers.


Changes
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fixed typo.


Summary
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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

r306216 tries to solve it, but causes more channel locking issues 
see https://issues.asterisk.org/view.php?id=18837#132337


This addresses bugs 18468, 18491, 18734, and 18837.
    https://issues.asterisk.org/view.php?id=18468
    https://issues.asterisk.org/view.php?id=18491
    https://issues.asterisk.org/view.php?id=18734
    https://issues.asterisk.org/view.php?id=18837


Diffs (updated)
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  trunk/channels/chan_sip.c 308944 

Diff: https://reviewboard.asterisk.org/r/1126/diff


Testing
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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.


Thanks,

Alec

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