Use an API call (ast_rtp_get_bridged) to return the RTP stream we are bridged to...
authorJoshua Colp <jcolp@digium.com>
Wed, 30 Aug 2006 03:16:03 +0000 (03:16 +0000)
committerJoshua Colp <jcolp@digium.com>
Wed, 30 Aug 2006 03:16:03 +0000 (03:16 +0000)
commit12b6ec4e11f872ba59d8ffe72f479d3fd14bb00f
tree45b7b58240ee6153fe6044d3ec6d36d6c9013c93
parentda7d969ae10e89c3fb2719dbac5099e5d09c56ef
Use an API call (ast_rtp_get_bridged) to return the RTP stream we are bridged to, and also use it in chan_sip so we know to ignore the no RTP activity checking

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@41316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c
include/asterisk/rtp.h
main/rtp.c