[svn-commits] file: branch 1.4 r78172 - /branches/1.4/main/rtp.c

Mon Aug 6 10:27:24 CDT 2007

Author: file
Date: Mon Aug  6 10:27:24 2007
New Revision: 78172

URL: http://svn.digium.com/view/asterisk?view=rev&rev=78172
Log:
(closes issue #10355)
Reported by: wdecarne
Now that we pass through RTP timestamp information we need to make the allowed timestamp skew considerably less. There are situations where a source may change and due to the timestamp difference the receiver will experience an audio gap since we did not indicate by setting the marker bit that the source changed.

Modified:
    branches/1.4/main/rtp.c

Modified: branches/1.4/main/rtp.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/main/rtp.c?view=diff&rev=78172&r1=78171&r2=78172
==============================================================================

--- branches/1.4/main/rtp.c (original)
+++ branches/1.4/main/rtp.c Mon Aug  6 10:27:24 2007
@@ -57,7 +57,7 @@
 #include "asterisk/unaligned.h"
 #include "asterisk/utils.h"
 
-#define MAX_TIMESTAMP_SKEW	640
+#define MAX_TIMESTAMP_SKEW	20
 
 #define RTP_SEQ_MOD     (1<<16) 	/*!< A sequence number can't be more than 16 bits */
 #define RTCP_DEFAULT_INTERVALMS   5000	/*!< Default milli-seconds between RTCP reports we send */


