Merged revisions 48964 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48965
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