chan_sip: Fix RTCP port for SRFLX ICE candidates
authorKinsey Moore <kmoore@digium.com>
Fri, 1 Nov 2013 12:40:40 +0000 (12:40 +0000)
committerKinsey Moore <kmoore@digium.com>
Fri, 1 Nov 2013 12:40:40 +0000 (12:40 +0000)
commit98dea21bc14a16831710beaadb3b859b7b7a0637
tree0892301be5bc5d940b5beb197984aebd01a17274
parent4053f36a71fcb0d8354a9023c9514d51c0011892
chan_sip: Fix RTCP port for SRFLX ICE candidates

This corrects one-way audio between Asterisk and Chrome/jssip as a
result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX
ICE candidates. This also exposes an ICE component enumeration to
extract further details from candidates.

(closes issue ASTERISK-21383)
Reported by: Shaun Clark
Review: https://reviewboard.asterisk.org/r/2967/
........

Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 402348 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/chan_sip.c
include/asterisk/rtp_engine.h
res/res_rtp_asterisk.c