Merged revisions 221086 via svnmerge from

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

........
  r221086 | twilson | 2009-09-30 09:49:11 -0500 (Wed, 30 Sep 2009) | 25 lines
  
  Change the SSRC by default when our media stream changes
  
  Be default, change SSRC when doing an audio stream changes Asterisk doesn't
  honor marker bit when reinvited to already-bridged RTP streams,resulting in
  far-end stack discarding packets with "old" timestamps that areactually part of
  a new stream.  This patch sends AST_CONTROL_SRCUPDATE whenever there is a
  reinvite, unless the 'constantssrc' is set to true in sip.conf.
  
  The original issue reported to Digium support detailed the following situation:
  ITSP <-> Asterisk 1.4.26.2 <-> SIP-based Application Server Call comes in
  fromITSP, Asterisk dials the app server which sends a re-invite back
  toAsterisk--not to negotiate to send media directly to the ITSP, but to
  indicatethat it's changing the stream it's sending to Asterisk.  The app
  servergenerates a new SSRC, sequence numbers, timestamps, and sets the marker
  bit on the new stream.  Asterisk passes through the teimstamp of the new stream,
  butdoes not reset the SSRC, sequence numbers, or set the marker bit.
  
  When the timestamp on the new stream is older than the timestamp on the
  originalstream, the ITSP (which doesn't know there has been any change) discards
  the newframes because it thinks they are too old.  This patch addresses this by
  changing the SSRC on a stream update unless constantssrc=true is set in
  sip.conf.
  
  Review: https://reviewboard.asterisk.org/r/374/
........


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@221266 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 14 insertions, 0 deletions

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 3abb6c686..42cce3786 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -103,6 +103,7 @@ enum strict_rtp_state {
 #define FLAG_NAT_INACTIVE_NOWARN        (1 << 1)
 #define FLAG_NEED_MARKER_BIT            (1 << 3)
 #define FLAG_DTMF_COMPENSATE            (1 << 4)
+#define FLAG_CONSTANT_SSRC              (1 << 5)
 
 /*! \brief RTP session description */
 struct ast_rtp {
@@ -253,6 +254,7 @@ static int ast_rtp_destroy(struct ast_rtp_instance *instance);
 static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
 static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
 static void ast_rtp_new_source(struct ast_rtp_instance *instance);
+static void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance);
 static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
 static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
 static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
@@ -275,6 +277,7 @@ static struct ast_rtp_engine asterisk_rtp_engine = {
 	.dtmf_begin = ast_rtp_dtmf_begin,
 	.dtmf_end = ast_rtp_dtmf_end,
 	.new_source = ast_rtp_new_source,
+	.constant_ssrc_set = ast_rtp_set_constantssrc,
 	.write = ast_rtp_write,
 	.read = ast_rtp_read,
 	.prop_set = ast_rtp_prop_set,
@@ -653,6 +656,13 @@ static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
 	return 0;
 }
 
+void ast_rtp_set_constantssrc(struct ast_rtp_instance *instance)
+{
+	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
+
+	ast_set_flag(rtp, FLAG_CONSTANT_SSRC);
+}
+
 static void ast_rtp_new_source(struct ast_rtp_instance *instance)
 {
 	struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
@@ -660,6 +670,10 @@ static void ast_rtp_new_source(struct ast_rtp_instance *instance)
 	/* We simply set this bit so that the next packet sent will have the marker bit turned on */
 	ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
 
+	if (!ast_test_flag(rtp, FLAG_CONSTANT_SSRC)) {
+		rtp->ssrc = ast_random();
+	}
+
 	return;
 }