Terry Wilson 68d1ded8dd 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/trunk@252089 65c4cc65-6c06-0410-ace0-fbb531ad65f3		2010-03-12 22:04:51 +00:00
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include	Only change the RTP ssrc when we see that it has changed	2010-03-12 22:04:51 +00:00
config_parser.c	fixes some test description formatting inconsistencies so log file looks nice	2010-02-11 21:57:37 +00:00
dialplan_functions.c	Make all of the various rtpqos parameters in this branch available from the CHANNEL function.	2010-02-17 06:25:15 +00:00
reqresp_parser.c	chan_sip parse code refactoring plus two new unit tests	2010-02-15 15:45:02 +00:00