Only change the RTP ssrc when we see that it has changed
authorTerry Wilson <twilson@digium.com>
Fri, 12 Mar 2010 22:04:51 +0000 (22:04 +0000)
committerTerry Wilson <twilson@digium.com>
Fri, 12 Mar 2010 22:04:51 +0000 (22:04 +0000)
commit68d1ded8dd40f646017da31c1fd3b4a0cc23c375
tree24326afc8f1cbf64c5dc15d7013b19991584bf86
parent828bdd8929000a8bba2be8ca5d4f65144a90b9dd
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
12 files changed:
addons/chan_ooh323.c
channels/chan_h323.c
channels/chan_mgcp.c
channels/chan_sip.c
channels/chan_skinny.c
channels/sip/include/sip.h
configs/sip.conf.sample
include/asterisk/frame.h
include/asterisk/rtp_engine.h
main/channel.c
main/rtp_engine.c
res/res_rtp_asterisk.c