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

[Joshua Colp]

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Ship it!


Ship It!

- Joshua Colp


On June 25, 2014, 12:14 p.m., Corey Farrell wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3671/
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> 
> (Updated June 25, 2014, 12:14 p.m.)
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> 
> Review request for Asterisk Developers, Joshua Colp and Matt Jordan.
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> Repository: Asterisk
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> 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.
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> * 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.
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> 
> 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/
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> 
> 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.
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> 
> Thanks,
> 
> Corey Farrell
> 
>

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