[asterisk-commits] twilson: branch group/srtp_reboot r259800 - in /team/group/srtp_reboot: chann...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Apr 28 15:09:19 CDT 2010
Author: twilson
Date: Wed Apr 28 15:09:16 2010
New Revision: 259800
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=259800
Log:
Cleanup whitespace
Modified:
team/group/srtp_reboot/channels/chan_sip.c
team/group/srtp_reboot/main/channel.c
team/group/srtp_reboot/res/res_srtp.c
Modified: team/group/srtp_reboot/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/srtp_reboot/channels/chan_sip.c?view=diff&rev=259800&r1=259799&r2=259800
==============================================================================
--- team/group/srtp_reboot/channels/chan_sip.c (original)
+++ team/group/srtp_reboot/channels/chan_sip.c Wed Apr 28 15:09:16 2010
@@ -5009,7 +5009,6 @@
/* We're replacing a call. */
p->options->replaces = ast_var_value(current);
}
-
}
/* Check to see if we should try to force encryption */
Modified: team/group/srtp_reboot/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/team/group/srtp_reboot/main/channel.c?view=diff&rev=259800&r1=259799&r2=259800
==============================================================================
--- team/group/srtp_reboot/main/channel.c (original)
+++ team/group/srtp_reboot/main/channel.c Wed Apr 28 15:09:16 2010
@@ -4625,7 +4625,7 @@
if ((ds = ast_channel_datastore_find(r, &secure_call_info, NULL))) {
struct ast_secure_call_store *encrypt = ds->data;
ops[0][1] = encrypt->signaling;
- ops[1][1] = encrypt->media;
+ ops[1][1] = encrypt->media;
} else {
return 0;
}
Modified: team/group/srtp_reboot/res/res_srtp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/srtp_reboot/res/res_srtp.c?view=diff&rev=259800&r1=259799&r2=259800
==============================================================================
--- team/group/srtp_reboot/res/res_srtp.c (original)
+++ team/group/srtp_reboot/res/res_srtp.c Wed Apr 28 15:09:16 2010
@@ -293,7 +293,7 @@
if (srtp->cb && srtp->cb->no_ctx) {
if (ast_rtp_instance_get_stats(srtp->rtp, &stats, AST_RTP_INSTANCE_STAT_REMOTE_SSRC)) {
break;
- }
+ }
if (srtp->cb->no_ctx(srtp->rtp, stats.remote_ssrc, srtp->data) < 0) {
break;
}
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