[asterisk-dev] [Code Review] SIP: INVITE w/Replaces deadlock fix

[David Vossel]

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https://reviewboard.asterisk.org/r/371/
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Review request for Asterisk Developers.


Summary
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This patch cleans up the locking logic in chan_sip.c's handle_invite_replaces() function as well as making use of ast_do_masquerade() rather than forcing the masquerade on an ast_read().  The code had several redundant unlocks that would result in 'freed more times than we've locked!' errors. I cleaned these up as well as moving all the unlock logic to the end of the function.  This patch should also resolve the issue people were having with the replacecall channel never being unlocked with one legged calls.

This is very tricky code, as far as I can tell through my testing and tracing back through the code, the function starts with 4 locks held, the the sip_pvt, it's channel, and the replace sip_pvt and it's channel.  The function should end with only the original sip_pvt being locked as it will be unlocked by sipsock_read().


This addresses bug 0015151.
    https://issues.asterisk.org/view.php?id=0015151


Diffs
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  /branches/1.4/channels/chan_sip.c 218511 

Diff: https://reviewboard.asterisk.org/r/371/diff


Testing
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Tested attended transfers using Replaces header.  Tried both waiting for transfer to pickup and then hanging up, as well as hanging up while transfer was still ringing.  Both work. no more mutex unlock errors.


Thanks,

David