[asterisk-dev] [Code Review] Pimp SIP Media generification

Joshua Colp reviewboard at asterisk.org
Mon Mar 11 10:15:04 CDT 2013


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Minor comments.


team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2380/#comment15384>

    This will fall apart when media streams are introduced after initial setup. A note indicating such would be good.



team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2380/#comment15385>

    This should only be in the RFC_4733 section, since that is where it is relevant.



team/group/pimp_my_sip/channels/chan_gulp.c
<https://reviewboard.asterisk.org/r/2380/#comment15386>

    Same here.



team/group/pimp_my_sip/res/res_sip_session.c
<https://reviewboard.asterisk.org/r/2380/#comment15387>

    This is going to fall apart in the future. Multiplexing of streams over the same port is a thing for WebRTC.


- Joshua


On March 11, 2013, 9:57 a.m., opticron wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2380/
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> 
> (Updated March 11, 2013, 9:57 a.m.)
> 
> 
> Review request for Asterisk Developers, Mark Michelson and Joshua Colp.
> 
> 
> Summary
> -------
> 
> Abstract media type restrictions out of res_sip_session and move them to chan_gulp where they're actually needed.  Due to the change, the sdp handler callback structure has been modified to accept a ast_sip_session_media struct and had a destroy function added and ast_sip_session_media_position has been removed from res_sip_session.h
> 
> This will need updates when Pimp SIP NAT goes in.
> 
> 
> This addresses bug ASTERISK-21184.
>     https://issues.asterisk.org/jira/browse/ASTERISK-21184
> 
> 
> Diffs
> -----
> 
>   team/group/pimp_my_sip/channels/chan_gulp.c 382643 
>   team/group/pimp_my_sip/include/asterisk/res_sip_session.h 382643 
>   team/group/pimp_my_sip/res/res_sip_sdp_audio.c 382643 
>   team/group/pimp_my_sip/res/res_sip_session.c 382643 
> 
> Diff: https://reviewboard.asterisk.org/r/2380/diff
> 
> 
> Testing
> -------
> 
> Tested with call scenarios from SDP_offer_answer integration test using a quickly hacked together video sdp handler which may or may not work properly but responds like it should (cp res/res_sip_sdp_audio.c res/res_sip_sdp_video.c;sed -i 's/AUDIO/VIDEO/;s/audio/video/' res/res_sip_sdp_video.c)
> 
> 
> Thanks,
> 
> opticron
> 
>

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