Merged revisions 370619 via svnmerge from

Merged revisions 370619 via svnmerge from
file:///srv/subversion/repos/asterisk/branches/10

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  r370619 | mmichelson | 2012-07-31 10:31:57 -0500 (Tue, 31 Jul 2012) | 34 lines

  Help mitigate potential reinvite glare scenarios.

  When Asterisk servers are set up back-to-back, and
  direct media is to be used betweeen endpoints, it is
  fairly common for the two Asterisk servers to send
  direct media reinvites to each other simultaneously.
  This results in 491s and ACKs being exchanged between
  the servers. While the media eventually gets set up
  properly, the problem is that there can be a noticeable
  delay for the streams to stabilize.

  This patch adds a new directmedia option called "outgoing".
  With this set, an immediate direct media reinvite will only
  be sent if the call direction is outgoing. For incoming
  dialogs, an immediate direct media reinvite will not be sent,
  but further "reactionary" direct media reinvites may be sent.

  For those who are having some deja vu, that's because this
  patch was originally committed to trunk since there is a
  new configuration option added. After seeing a bug report
  about audio being slow to set up on SIP calls, it became
  apparent that this patch would be the best solution for
  resolving the issue. The patch is unintrusive and will
  have no effect unless the option is explicitly enabled.

  (closes issue AST-896)
  reported by Thomas Arimont

  (closes issue ASTERISK-19857)
  reported by Matt Jordan
  ........

  Merged revisions 370618 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10-digiumphones@370633 65c4cc65-6c06-0410-ace0-fbb531ad65f3