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authorJoshua Colp <jcolp@digium.com>2014-08-24 17:22:48 +0000
committerJoshua Colp <jcolp@digium.com>2014-08-24 17:22:48 +0000
commitcee660dadf1e809bbc9fc452d711e0da89b006bb (patch)
treeb5b9e730726238eede8da3ad894baeb84d328651 /res/res_pjsip_transport_websocket.c
parentdcfffce66dfd156f419b04744297e99a464b666f (diff)
chan_sip: Use the server reflexive ICE candidate RTCP port as provided.
This code originally worked around an issue within res_rtp_asterisk itself. The wrong socket was being used for the STUN check for RTCP, causing the port to be the same as RTP. This was subsequently fixed and the RTCP port provided for the ICE candidate is correct and does not need to be incremented. ASTERISK-23997 #close Reported by: Badalian Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav (license 5249) ........ Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 421911 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421912 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip_transport_websocket.c')
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