[asterisk-dev] [Code Review] 3671: media_formats: res_rtp_asterisk and other fixes

[Corey Farrell]

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https://reviewboard.asterisk.org/r/3671/
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(Updated June 25, 2014, 1:54 p.m.)


Status
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This change has been marked as submitted.


Review request for Asterisk Developers, Joshua Colp and Matt Jordan.


Changes
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Committed in revision 417211


Repository: Asterisk


Description
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This update gives media_formats the ability to receive a call using chan_sip.  Possibly other channel drivers might work, I haven't tried them.

* ast_format_cap_is_compatible_format needs to be checked against AST_FORMAT_CMP_NOT_EQUAL, not zero/non-zero.  All calls to ast_format_cap_is_compatible_format were fixed.
* res_rtp_asterisk was updated by Matt Jordan, along with related changes to codec.c, codec.h, format.c, format.c and codec_builtin.c.
* Switch ast_format_copy from function to macro to ao2_bump.  This allows REF_DEBUG to give better results.


Diffs
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  /team/group/media_formats-reviewed-trunk/res/res_speech.c 417190 
  /team/group/media_formats-reviewed-trunk/res/res_rtp_asterisk.c 417190 
  /team/group/media_formats-reviewed-trunk/main/translate.c 417190 
  /team/group/media_formats-reviewed-trunk/main/frame.c 417190 
  /team/group/media_formats-reviewed-trunk/main/format.c 417190 
  /team/group/media_formats-reviewed-trunk/main/codec_builtin.c 417190 
  /team/group/media_formats-reviewed-trunk/main/codec.c 417190 
  /team/group/media_formats-reviewed-trunk/main/channel.c 417190 
  /team/group/media_formats-reviewed-trunk/main/bridge.c 417190 
  /team/group/media_formats-reviewed-trunk/include/asterisk/format.h 417190 
  /team/group/media_formats-reviewed-trunk/include/asterisk/codec.h 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_unistim.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_skinny.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_sip.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_pjsip.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_oss.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_nbs.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_motif.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_mgcp.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_jingle.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_h323.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_gtalk.c 417190 
  /team/group/media_formats-reviewed-trunk/channels/chan_alsa.c 417190 
  /team/group/media_formats-reviewed-trunk/addons/chan_ooh323.c 417190 
  /team/group/media_formats-reviewed-trunk/addons/chan_mobile.c 417190 

Diff: https://reviewboard.asterisk.org/r/3671/diff/


Testing
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Called from Asterisk 11 to a test server with this code, I was able to hear the 'invalid' message, everything seemed during the call.  I received TONS of ao2 frack's when stopping Asterisk.  The sip.conf peer on both Asterisk servers was setup for disallow=all / allow=ulaw.


Thanks,

Corey Farrell

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