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
This commit is contained in:
Gregory Nietsky
2011-09-20 16:51:59 +00:00
parent 5f1a062fa6
commit a31b5ce87e
7 changed files with 36 additions and 13 deletions

View File

@@ -353,9 +353,10 @@
#define SIP_PAGE3_SNOM_AOC (1 << 0) /*!< DPG: Allow snom aoc messages */
#define SIP_PAGE3_SRTP_TAG_32 (1 << 1) /*!< DP: Use a 32bit auth tag in INVITE not 80bit */
#define SIP_PAGE3_FLAGS_TO_COPY \
(SIP_PAGE3_SNOM_AOC)
(SIP_PAGE3_SNOM_AOC | SIP_PAGE3_SRTP_TAG_32)
/*@}*/