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

[opticron]

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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, 4:47 p.m.)


Review request for Asterisk Developers, Mark Michelson and Joshua Colp.


Changes
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Update to handle WebRTC-style RTP muxing in the SDP validation routine.


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

This will need updates when Pimp SIP NAT goes in.


This addresses bug ASTERISK-21184.
    https://issues.asterisk.org/jira/browse/ASTERISK-21184


Diffs (updated)
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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 

Diff: https://reviewboard.asterisk.org/r/2380/diff


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)


Thanks,

opticron

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