[asterisk-dev] [Code Review] SIP various transport type issues

[Russell Bryant]

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Ship it!



/trunk/channels/chan_sip.c
<http://reviewboard.digium.com/r/278/#comment2074>

    space after if


- Russell


On 2009-06-16 10:09:07, David Vossel wrote:
> 
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> (Updated 2009-06-16 10:09:07)
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> 
> Review request for Asterisk Developers.
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> 
> Summary
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> This is a combination of patches from vrban, mmichelson, and myself relating to (issue #13865).
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> What this patch addresses:
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> 1. ast_sip_ouraddrfor() by default binds to the UDP address/port reguardless if the sip->pvt is of type UDP or not.  Now when no remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's transport type, attempting to set the address and port to the correct TCP/TLS bindings if necessary.
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> 2.  It is not necessary to send the port number in the Contact header unless the port is non-standard for the transport type.  This patch fixes this and removes the todo note.
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> 3.  In sip_alloc(), the default dialog built always uses transport type UDP.  Now sip_alloc() looks at the sip_request (if present) and determines what transport type to use by default.
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> 4.  When changing the transport type of a sip_socket, the file descriptor must be set to -1 and in some cases the tcptls_session's ref count must be decremented and set to NULL.  I've encountered several issues associated with this process and have created a function, set_socket_transport(), to handle the setting of the socket type.
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> 
> This addresses bug 13865.
>     https://issues.asterisk.org/view.php?id=13865
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> 
> Diffs
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>   /trunk/channels/chan_sip.c 200941 
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> Diff: http://reviewboard.digium.com/r/278/diff
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> Testing
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> It appears through issue notes that vrban has tested items 1-3.  I have only reviewed/cleaned up a few possible errors and done sanity checks on them.  I have tested item 4.  The function was previously used only for changing a peer's transport type.  I simply expanded it to be used every time any socket's transport type is changed
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> Thanks,
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> David
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>