[asterisk-dev] [Code Review] 4032: PJSIP: Force transport on contact rewrite

Mark Michelson reviewboard at asterisk.org
Tue Sep 30 10:19:32 CDT 2014


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Ship it!



branches/12/res/res_pjsip_nat.c
<https://reviewboard.asterisk.org/r/4032/#comment23892>

    Just set uri->transport_param.slen = 0 instead of creating the empty_string constant.


- Mark Michelson


On Sept. 30, 2014, 12:19 p.m., opticron wrote:
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/4032/
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> (Updated Sept. 30, 2014, 12:19 p.m.)
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> 
> Review request for Asterisk Developers.
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> Repository: Asterisk
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> Description
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> If contact rewriting is enabled but the contact differs in transport from what is actually being used, messages after the initial INVITE transaction can be sent to an incorrect transport/port combination. In the case where this bug occurred the remote party never received a BYE since it was sent to the remote party's TCP port over UDP.
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> Diffs
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>   branches/12/res/res_pjsip_nat.c 424094 
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> Diff: https://reviewboard.asterisk.org/r/4032/diff/
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> Testing
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> Ensured that this patch allowed the BYE to be sent properly.
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> Thanks,
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> opticron
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>

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