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Merged revisions 252089 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/trunk

........
  r252089 | twilson | 2010-03-12 16:04:51 -0600 (Fri, 12 Mar 2010) | 20 lines
  
  Only change the RTP ssrc when we see that it has changed
  
  This change basically reverts the change reviewed in
  https://reviewboard.asterisk.org/r/374/ and instead limits the
  updating of the RTP synchronization source to only those times when we
  detect that the other side of the conversation has changed the ssrc.
  
  The problem is that SRCUPDATE control frames are sent many times where
  we don't want a new ssrc, including whenever Asterisk has to send DTMF
  in a normal bridge. This is also not the first time that this mistake
  has been made. The initial implementation of the ast_rtp_new_source
  function also changed the ssrc--and then it was removed because of
  this same issue. Then, we put it back in again to fix a different
  issue. This patch attempts to only change the ssrc when we see that
  the other side of the conversation has changed the ssrc.
  
  It also renames some functions to make their purpose more clear.
  
  Review: https://reviewboard.asterisk.org/r/540/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@252175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Terry Wilson
2010-03-13 00:30:04 +00:00
parent a247e69d65
commit 529e8af144
9 changed files with 79 additions and 54 deletions
Download Patch File Download Diff File Expand all files Collapse all files
include/asterisk/frame.h
+3 -1
View File
@@ -85,7 +85,8 @@ struct ast_codec_pref {
\arg \b HOLD Call is placed on hold
\arg \b UNHOLD Call is back from hold
\arg \b VIDUPDATE Video update requested
\arg \b SRCUPDATE The source of media has changed
\arg \b SRCUPDATE The source of media has changed (RTP marker bit must change)
\arg \b SRCCHANGE Media source has changed (RTP marker bit and SSRC must change)
*/
@@ -290,6 +291,7 @@ enum ast_control_frame_type {
AST_CONTROL_UNHOLD = 17, /*!< Indicate call is left from hold */
AST_CONTROL_VIDUPDATE = 18, /*!< Indicate video frame update */
AST_CONTROL_SRCUPDATE = 20, /*!< Indicate source of media has changed */
AST_CONTROL_SRCCHANGE = 21, /*!< Media has changed and requires a new RTP SSRC */
};
#define AST_SMOOTHER_FLAG_G729 (1 << 0)
include/asterisk/rtp.h
+4 -3
View File
@@ -179,10 +179,11 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
/*! \brief When changing sources, don't generate a new SSRC */
void ast_rtp_set_constantssrc(struct ast_rtp *rtp);
/*! \brief Indicate that we need to set the marker bit */
void ast_rtp_update_source(struct ast_rtp *rtp);
void ast_rtp_new_source(struct ast_rtp *rtp);
/*! \brief Indicate that we need to set the marker bit and change the ssrc */
void ast_rtp_change_source(struct ast_rtp *rtp);
/*! \brief Setting RTP payload types from lines in a SDP description: */
void ast_rtp_pt_clear(struct ast_rtp* rtp);