[Asterisk-code-review] res pjsip: Implement additional SIP RFCs for Google Voice tr... (asterisk[master])

[Joshua Colp]

Joshua Colp has posted comments on this change. ( https://gerrit.asterisk.org/9505 )

Change subject: res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
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Patch Set 7:

(1 comment)

https://gerrit.asterisk.org/#/c/9505/7/include/asterisk/res_pjsip.h
File include/asterisk/res_pjsip.h:

https://gerrit.asterisk.org/#/c/9505/7/include/asterisk/res_pjsip.h@127
PS7, Line 127: 	/*!
             : 	 * Set when this transport is a flow of signaling to a target
             : 	 * \since 17.0.0
             : 	 */
             : 	int flow;
> Any reason not to make this a new ast_transport enum type instead of a separate field?
The thought crossed my mind but flows are a res_pjsip concept... I could add one just for a "connection" I suppose.



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Gerrit-Project: asterisk
Gerrit-Branch: master
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Gerrit-Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58
Gerrit-Change-Number: 9505
Gerrit-PatchSet: 7
Gerrit-Owner: Nick French <naf at ou.edu>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Kevin Harwell <kharwell at digium.com>
Gerrit-Reviewer: Michael L. Young <elgueromexicano at gmail.com>
Gerrit-Reviewer: Nick French <naf at ou.edu>
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