Multiple revisions 402345,405234,409129-409130,409565,413008,417141,417677

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  r402345 | kmoore | 2013-11-01 05:31:49 -0700 (Fri, 01 Nov 2013) | 11 lines
  
  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/
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  r405234 | kharwell | 2014-01-09 08:49:55 -0800 (Thu, 09 Jan 2014) | 19 lines
  
  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)
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  r409129 | jrose | 2014-02-27 11:19:02 -0800 (Thu, 27 Feb 2014) | 15 lines
  
  res_rtp_asterisk: Fix checklist creating problems in ICE sessions
  
  Prior to this patch, local candidate lists including SRFLX would fail to start
  properly when building ICE candidate check lists. This patch fixes that problem
  by making sure that each SRFLX candidate is associated with the proper
  base address so that the check list can create matches properly.
  This patch was written by jcolp. The issue will be left open to await testing
  by the issue participants.
  
  (issue ASTERISK-23213)
  Reported by: Andrea Suisani
  Review: https://reviewboard.asterisk.org/r/3256/
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  r409130 | jrose | 2014-02-27 11:38:10 -0800 (Thu, 27 Feb 2014) | 8 lines
  
  res_rtp_asterisk: correct build error from r409129
  
  Accidentally placed a declaration below functional code
  
  (issue ASTERISK-23213)
  Reported by: Andrea Suisani
  Review: https://reviewboard.asterisk.org/r/3256/
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  r409565 | jrose | 2014-03-04 08:40:39 -0800 (Tue, 04 Mar 2014) | 9 lines
  
  res_rtp_asterisk: Fix one way audio problems with hold/unhold when using ICE
  
  ICE sessions will now be restarted if sessions are changed to use new sets of
  remote candidates.
  
  (closes issue ASTERISK-22911)
  Reported by: Vytis Valentinavičius
  Review: https://reviewboard.asterisk.org/r/3275/
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  r413008 | mjordan | 2014-04-25 10:47:21 -0700 (Fri, 25 Apr 2014) | 14 lines
  
  res_rtp_asterisk: Add support for DTLS handshake retransmissions
  
  On congested networks, it is possible for the DTLS handshake messages to get
  lost. This patch adds a timer to res_rtp_asterisk that will periodically
  check to see if the handshake has succeeded. If not, it will retransmit the
  DTLS handshake.
  
  Review: https://reviewboard.asterisk.org/r/3337
  
  ASTERISK-23649 #close
  Reported by: Nitesh Bansal
  patches:
    dtls_retransmission.patch uploaded by Nitesh Bansal (License 6418)
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  r417141 | file | 2014-06-23 11:49:14 -0700 (Mon, 23 Jun 2014) | 5 lines
  
  res_rtp_asterisk: Return the length of data written when sending via ICE instead of 0.
  
  ASTERISK-23834 #close
  Reported by: Richard Kenner
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  r417677 | file | 2014-06-30 12:42:18 -0700 (Mon, 30 Jun 2014) | 12 lines
  
  res_rtp_asterisk: Add SHA-256 support for DTLS and perform DTLS negotiation on RTCP.
  
  This change fixes up DTLS support in res_rtp_asterisk so it can accept and provide
  a SHA-256 fingerprint, so it occurs on RTCP, and so it occurs after ICE negotiation
  completes. Configuration options to chan_sip have also been added to allow behavior
  to be tweaked (such as forcing the AVP type media transports in SDP).
  
  ASTERISK-22961 #close
  Reported by: Jay Jideliov
  
  Review: https://reviewboard.asterisk.org/r/3679/
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Merged revisions 402345,405234,409129-409130,409565,413008,417141,417677 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/certified/branches/11.6@417724 65c4cc65-6c06-0410-ace0-fbb531ad65f3