Merge in the RTP engine API.
authorJoshua Colp <jcolp@digium.com>
Thu, 2 Apr 2009 17:20:52 +0000 (17:20 +0000)
committerJoshua Colp <jcolp@digium.com>
Thu, 2 Apr 2009 17:20:52 +0000 (17:20 +0000)
commit63de8343958b91c8836c5e6ddf1c0106b40e9fe6
tree8a8042738e1c444e5988a648b795c4d2b02febd1
parent08971ce2056f4e035b4b37324c7f184370cd0ec6
Merge in the RTP engine API.

This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.

Review: http://reviewboard.digium.com/r/209/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
24 files changed:
UPGRADE.txt
apps/app_dial.c
channels/chan_agent.c
channels/chan_bridge.c
channels/chan_gtalk.c
channels/chan_h323.c
channels/chan_jingle.c
channels/chan_local.c
channels/chan_mgcp.c
channels/chan_sip.c
channels/chan_skinny.c
channels/chan_unistim.c
configs/sip.conf.sample
include/asterisk/_private.h
include/asterisk/rtp.h [deleted file]
include/asterisk/rtp_engine.h [new file with mode: 0644]
include/asterisk/stun.h [new file with mode: 0644]
main/Makefile
main/asterisk.c
main/loader.c
main/rtp.c [deleted file]
main/rtp_engine.c [new file with mode: 0644]
main/stun.c [new file with mode: 0644]
res/res_rtp_asterisk.c [new file with mode: 0644]