[asterisk-dev] [Code Review] Fix problems when RTP packet frame size is changed

[vadim at mbdsys.com]

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/branches/1.4/main/frame.c
<http://reviewboard.digium.com/r/184/#comment1191>

    I wonder, if this code is secure against s->data buffer overflow



/branches/1.4/main/frame.c
<http://reviewboard.digium.com/r/184/#comment1192>

    assert(size < SMOOTHER_SIZE) ?



/branches/1.4/main/frame.c
<http://reviewboard.digium.com/r/184/#comment1193>

    assert(bytes <= SMOOTHER_SIZE) ?


- vadim


On 2009-03-04 17:32:49, Kevin Fleming wrote:
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> (Updated 2009-03-04 17:32:49)
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> 
> Review request for Asterisk Developers.
> 
> 
> Summary
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> During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
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> This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
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> 
> Diffs
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>   /branches/1.4/include/asterisk/frame.h 180297 
>   /branches/1.4/main/frame.c 180297 
>   /branches/1.4/main/rtp.c 180297 
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> Diff: http://reviewboard.digium.com/r/184/diff
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> 
> Testing
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> Compile testing only.
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> Thanks,
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> Kevin
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>