[asterisk-dev] [Code Review]: res_sip and res_sip_session design review

Mark Michelson reviewboard at asterisk.org
Tue Jan 8 15:13:39 CST 2013



> On Jan. 8, 2013, 3:13 p.m., Mark Michelson wrote:
> > I've updated the wiki pages to have the recommended changes from Joshua and from Saul on the dev list
> > 
> > * There is a function to append body data and a function to add multipart bodies.
> > * SDP handling has been changed to use handlers of individual stream types instead of handlers for entire SDPs.
> > * ast_sip_session_send_reinvite() now has an optional callback to be called when the response is received.
> > * ast_sip_session now supports session cookies.
> > * "session extensions" are now "session supplements" in all cases.
> > * Errors regarding missing parameters in functions have been cleared up.

Also added session_begin() and session_end() callbacks to session supplements.


- Mark


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/2251/#review7639
-----------------------------------------------------------


On Dec. 20, 2012, 1:17 p.m., Mark Michelson wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2251/
> -----------------------------------------------------------
> 
> (Updated Dec. 20, 2012, 1:17 p.m.)
> 
> 
> Review request for Asterisk Developers, Matt Jordan and jcolp.
> 
> 
> Summary
> -------
> 
> This is a proposal for a res_sip and res_sip_session API for use in the new SIP channel driver. The pages are located here:
> 
> https://wiki.asterisk.org/wiki/display/AST/res_sip+design
> https://wiki.asterisk.org/wiki/display/AST/res_sip_session+design
> 
> Please let me know what you think of these.
> 
> There are a few things that are not here and that probably should
> * A struct called ast_sip_endpoint is referenced in a few places, but it is not defined. This is because a SIP endpoint is more-or-less defined by the DAL, which is currently under development by Mr. Joshua Colp. Once endpoint configuration and related structures are defined, they can be added in to these pages.
> * There are no functions in res_sip_session for iterating over SDP media streams or attributes, nor are there any functions for aiding in creating SDPs. These likely should exist, but I have not placed them here now since I have difficulty seeing what parameters will be necessary nor what they might return.
> 
> 
> Diffs
> -----
> 
> 
> Diff: https://reviewboard.asterisk.org/r/2251/diff
> 
> 
> Testing
> -------
> 
> The wiki page renders properly.
> 
> 
> Thanks,
> 
> Mark
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20130108/47bc4e53/attachment-0001.htm>


More information about the asterisk-dev mailing list