use existing sample size in frame instead of recalculating it (issue #5390)

git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6781 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 1 insertions, 1 deletions

diff --git a/rtp.c b/rtp.c
index 78cdfb09a..468445e5f 100755
--- a/rtp.c
+++ b/rtp.c
@@ -1231,7 +1231,7 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
 	ms = calc_txstamp(rtp, &f->delivery);
 	/* Default prediction */
 	if (f->subclass < AST_FORMAT_MAX_AUDIO) {
-                pred = rtp->lastts + ast_codec_get_samples(f);
+                pred = rtp->lastts + f->samples;
 
 		/* Re-calculate last TS */
 		rtp->lastts = rtp->lastts + ms * 8;