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Allow Setting Auth Tag Bit length Based on invite or config option
Update the SIP SRTP API to allow use of 32 or 80 bit taglen. Curently only 80 bit is supported. The outgoing invite will use the taglen of the incoming invite preventing one-way audio. (Closes issue ASTERISK-17895) Review: https://reviewboard.asterisk.org/r/1173/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@336936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -353,9 +353,10 @@
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#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
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#define SIP_PAGE3_SRTP_TAG_32 (1 << 1) /*!< DP: Use a 32bit auth tag in INVITE not 80bit */
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#define SIP_PAGE3_FLAGS_TO_COPY \
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(SIP_PAGE3_SNOM_AOC)
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(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32)
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/*@}*/
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