Merged revisions 78172 via svnmerge from

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

........
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.

........

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

diff --git a/main/rtp.c b/main/rtp.c
index c3c7acaedc3faccf3e8790abac212da0ed15d076..d8d99012f0db76f693492a2a96e7078ebd00f571 100644 (file)
--- a/main/rtp.c
+++ b/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 */