]> git.ipfire.org Git - thirdparty/asterisk.git/commit
res_rtp_asterisk: Fails to resume WebRTC call from hold
authorKevin Harwell <kharwell@digium.com>
Thu, 9 Jan 2014 16:52:57 +0000 (16:52 +0000)
committerKevin Harwell <kharwell@digium.com>
Thu, 9 Jan 2014 16:52:57 +0000 (16:52 +0000)
commitbce38c0cc59a3388ef15d0de3e41508be946aa5f
tree6efdc9911c9adec29f5ebe7fe759ec26d53f7842
parent50b2d6eec162de15370544c8feab6165a340e66a
res_rtp_asterisk: Fails to resume WebRTC call from hold

In ast_rtp_ice_start if the ice session create check list failed, start check
was never initiated and ice_started was never set to true.  Upon re-entering
the function (for instance, [un]hold) it would try to create the check list
again with duplicate remote candidates.

Fixed so that if the create check list fails the necessary data structures
are properly re-initialized for any subsequent retries.

Note, it was decided to not stop ice support (by calling ast_rtp_ice_stop) on a
check list failure because it possible things might still work.  However, a
debug message was added to help with any future troubleshooting.

(closes issue ASTERISK-22911)
Reported by: Vytis Valentinavičius
Patches:
     works_on_my_machine.patch uploaded by xytis (license 6558)
........

Merged revisions 405234 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 405235 from http://svn.asterisk.org/svn/asterisk/branches/12

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res/res_rtp_asterisk.c