Review Board 1.7.16


fix Deadlock with attended transfers of SIP calls

Review Request #1126 - Created Feb. 23, 2011 and submitted

Alec Davis
trunk
18468, 18491, 18734, 18837
Reviewers
asterisk-dev
Asterisk
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
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.
Review request changed
Updated (Feb. 25, 2011, 6:12 a.m.)
fixed typo.
Ship it!
Posted (Feb. 25, 2011, 6:19 a.m.)
Looks good.

https://reviewboard.asterisk.org/ runs on a server provided by Digium, Inc. and uses bandwidth donated to the open source Asterisk community by API Digital Communications in Huntsville, AL USA.
Please report problems with this site to asteriskteam@digium.com.