Merged revisions 48964 via svnmerge from
authorJoshua Colp <jcolp@digium.com>
Tue, 26 Dec 2006 04:34:07 +0000 (04:34 +0000)
committerJoshua Colp <jcolp@digium.com>
Tue, 26 Dec 2006 04:34:07 +0000 (04:34 +0000)
commit7f61b822c17ccadac726172a2b120e8c9d029abf
tree756df2c8bb71afc320d31e4d30afe941fb099ef6
parentb3ab5300776cb22075d6add23ec27d6a968e0f5c
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 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c
include/asterisk/rtp.h
main/rtp.c