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

[opticron]

> On Sept. 29, 2014, 2:47 p.m., Joshua Colp wrote:
> > branches/12/res/res_pjsip_nat.c, line 49
> > <https://reviewboard.asterisk.org/r/4032/diff/1/?file=67680#file67680line49>
> >
> >     What will this do for UDP transport? For those the transport should be empty.

"udp" is also a valid transport along with "tls", "ws", and others. Is there a benefit to leaving it blank if the transport is UDP?


- opticron


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On Sept. 29, 2014, 2:25 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. 29, 2014, 2:25 p.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> 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.
> 
> 
> Diffs
> -----
> 
>   branches/12/res/res_pjsip_nat.c 424094 
> 
> Diff: https://reviewboard.asterisk.org/r/4032/diff/
> 
> 
> Testing
> -------
> 
> Ensured that this patch allowed the BYE to be sent properly.
> 
> 
> Thanks,
> 
> opticron
> 
>

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