[asterisk-dev] [Code Review] Allow Setting Bitsize and make SRTP optional chan_sip

[irroot]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1173/
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(Updated 2011-05-21 14:37:28.337981)


Review request for Asterisk Developers.


Changes
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Ok changed the code to reflect that the bit size 32 refers to the auth tag not the crypto key strength.

a note will [MUST] be added to the config/changes that SRTP on its own offers no security only
encryption of the audio to the casual RTP tap if the SIP is not encrypted ie with TLS the key is exposed.


Summary
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change the encruption option to tristate with optional bit setting
also make this a global option.

qwell sugests a second option for bitlen have no problem with that.


This addresses bug 19335.
    https://issues.asterisk.org/view.php?id=19335


Diffs (updated)
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  /team/irroot/t38gateway-trunk/channels/chan_sip.c 319935 
  /team/irroot/t38gateway-trunk/channels/sip/include/sdp_crypto.h 319935 
  /team/irroot/t38gateway-trunk/channels/sip/include/sip.h 319935 
  /team/irroot/t38gateway-trunk/channels/sip/include/srtp.h 319935 
  /team/irroot/t38gateway-trunk/channels/sip/sdp_crypto.c 319935 

Diff: https://reviewboard.asterisk.org/r/1173/diff


Testing
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Thanks,

irroot

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