[asterisk-commits] res rtp asterisk: enable rtcp & QOS stats on native bridge (asterisk[14])

[SVN commits to the Asterisk project]

Jenkins2 has submitted this change and it was merged. ( https://gerrit.asterisk.org/6185 )

Change subject: res_rtp_asterisk: enable rtcp & QOS stats on native bridge
......................................................................

res_rtp_asterisk: enable rtcp & QOS stats on native bridge

Asterisk wasn't generating or forwarding RTCP packets when native
bridge was activated.  Also the stats weren't available via
CHANNEL(qos). Now the RTCP stats are always calculated.

ASTERISK-27158 #close

Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b
---
M res/res_rtp_asterisk.c
1 file changed, 22 insertions(+), 10 deletions(-)

Approvals:
  Joshua Colp: Looks good to me, but someone else must approve
  George Joseph: Looks good to me, approved
  Jenkins2: Approved for Submit



diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 8121dc9..fe1d1e3 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4853,9 +4853,6 @@
 		return -1;
 	}
 
-	rtp->rxcount++;
-	rtp->rxoctetcount += (len - hdrlen);
-
 	/* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
 	if (ast_rtp_codecs_find_payload_code(ast_rtp_instance_get_codecs(instance1), bridged_payload) == -1) {
 		ast_debug(1, "Unsupported payload type received \n");
@@ -5091,13 +5088,6 @@
 		}
 	}
 
-	/* If we are directly bridged to another instance send the audio directly out */
-	instance1 = ast_rtp_instance_get_bridged(instance);
-	if (instance1
-		&& !bridge_p2p_rtp_write(instance, instance1, rtpheader, res, hdrlen)) {
-		return &ast_null_frame;
-	}
-
 	/* If the version is not what we expected by this point then just drop the packet */
 	if (version != 2) {
 		return &ast_null_frame;
@@ -5202,6 +5192,28 @@
 		rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
 	}
 
+
+	/* If we are directly bridged to another instance send the audio directly out,
+	 * but only after updating core information about the received traffic so that
+	 * outgoing RTCP reflects it.
+	 */
+	instance1 = ast_rtp_instance_get_bridged(instance);
+	if (instance1
+		&& !bridge_p2p_rtp_write(instance, instance1, rtpheader, res, hdrlen)) {
+		struct timeval rxtime;
+		struct ast_frame *f;
+
+		/* Update statistics for jitter so they are correct in RTCP */
+		calc_rxstamp(&rxtime, rtp, timestamp, mark);
+
+		/* When doing P2P we don't need to raise any frames about SSRC change to the core */
+		while ((f = AST_LIST_REMOVE_HEAD(&frames, frame_list)) != NULL) {
+			ast_frfree(f);
+		}
+
+		return &ast_null_frame;
+	}
+
 	if (rtp_debug_test_addr(&addr)) {
 		ast_verbose("Got  RTP packet from    %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d)\n",
 			    ast_sockaddr_stringify(&addr),

-- 
To view, visit https://gerrit.asterisk.org/6185
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: 14
Gerrit-MessageType: merged
Gerrit-Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b
Gerrit-Change-Number: 6185
Gerrit-PatchSet: 1
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
Gerrit-Reviewer: George Joseph <gjoseph at digium.com>
Gerrit-Reviewer: Jenkins2
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-commits/attachments/20170809/77212413/attachment-0001.html>