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

[Joshua Colp]

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Ship it!


One minor comment.


team/group/pimp_my_sip/res/res_sip_session.c
<https://reviewboard.asterisk.org/r/2380/#comment15438>

    Instead of mallocing you could just return the pj_str_t value and deal with that. It will be valid for the duration. It also means you can get rid of media_has_mid and use media_get_mid.


- Joshua


On March 11, 2013, 4:47 p.m., opticron wrote:
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> (Updated March 11, 2013, 4:47 p.m.)
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> 
> Review request for Asterisk Developers, Mark Michelson and Joshua Colp.
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> Summary
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> 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
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> This will need updates when Pimp SIP NAT goes in.
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> This addresses bug ASTERISK-21184.
>     https://issues.asterisk.org/jira/browse/ASTERISK-21184
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> 
> Diffs
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>   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 
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> Diff: https://reviewboard.asterisk.org/r/2380/diff
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> Testing
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> 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)
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> Thanks,
> 
> opticron
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>

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