[asterisk-dev] [Code Review] Guard against retransmitting a BYE forever

David Vossel reviewboard at asterisk.org
Thu Jan 20 17:28:30 CST 2011


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Ship it!


This looks correct to me!

- David


On 2011-01-18 16:52:16, Terry Wilson wrote:
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> (Updated 2011-01-18 16:52:16)
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> Review request for Asterisk Developers and David Vossel.
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> Summary
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> In the case of an attended transfer (A calls B, A atxfers to C) where A becomes unreachable before replying to Asterisk's BYE, Asterisk can sometimes retransmit the BYE indefinitely. This is because __sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0], SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out, it will not ever be marked as ALREADYGONE, so when __sip_autodestruct is called again, we end up starting the cycle over.
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> This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt in the case of a BYE that has timed out. This should prevent Asterisk from trying to transmit new BYE messages in the future.
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> Diffs
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>   /branches/1.4/channels/chan_sip.c 302087 
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> Diff: https://reviewboard.asterisk.org/r/1077/diff
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> Testing
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> Modified Asterisk to sleep for 2 seconds before sending the transfer NOTIFY to give me time to unplug the network cable from the transfering phone. Registered 3 phones A, B, and C. A calls B, A atxfer to C. I often (but not always) got infinite BYEs. After the patch I did not.
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> Thanks,
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> Terry
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>

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