Merged revisions 78172 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.4

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r78172 | file | 2007-08-06 12:27:24 -0300 (Mon, 06 Aug 2007) | 4 lines

(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.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78173 65c4cc65-6c06-0410-ace0-fbb531ad65f3

main/rtp.c

@@ -59,7 +59,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
 #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 */