diff options
| author | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-03-13 00:30:04 +0000 |
|---|---|---|
| committer | twilson <twilson@f38db490-d61c-443f-a65b-d21fe96a405b> | 2010-03-13 00:30:04 +0000 |
| commit | 8b997b6ae37cf53d86a1918be28dfc999d4d1ee0 (patch) | |
| tree | 2f54dbec9891bf51682d9e18092aa7d98ebe3384 /channels/chan_h323.c | |
| parent | 3be293b2e1a22b1b3b42f7952ecb72a8773522c5 (diff) | |
Merged revisions 252089 via svnmerge from
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: http://svn.digium.com/svn/asterisk/branches/1.4@252175 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_h323.c')
| -rw-r--r-- | channels/chan_h323.c | 6 |
1 files changed, 5 insertions, 1 deletions
diff --git a/channels/chan_h323.c b/channels/chan_h323.c index 700fdd174..f8a929d6e 100644 --- a/channels/chan_h323.c +++ b/channels/chan_h323.c @@ -918,7 +918,11 @@ static int oh323_indicate(struct ast_channel *c, int condition, const void *data res = 0; break; case AST_CONTROL_SRCUPDATE: - ast_rtp_new_source(pvt->rtp); + ast_rtp_update_source(pvt->rtp); + res = 0; + break; + case AST_CONTROL_SRCCHANGE: + ast_rtp_change_source(pvt->rtp); res = 0; break; case AST_CONTROL_PROCEEDING: |