[Asterisk-code-review] res rtp asterisk: enable rtcp & QOS stats on native bridge (asterisk[master])

[Torrey Searle]

Torrey Searle has uploaded this change for review. ( https://gerrit.asterisk.org/6190


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(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/90/6190/1

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 65ab902..ce899df 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -5088,9 +5088,6 @@
 		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");
@@ -5362,13 +5359,6 @@
 		ast_rtp_dtmf_continuation(instance);
 	}
 
-	/* 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;
-	}
-
 	/* Pull out the various other fields we will need */
 	payloadtype = (seqno & 0x7f0000) >> 16;
 	padding = seqno & (1 << 29);
@@ -5467,6 +5457,28 @@
 		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/6190
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-MessageType: newchange
Gerrit-Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b
Gerrit-Change-Number: 6190
Gerrit-PatchSet: 1
Gerrit-Owner: Torrey Searle <tsearle at gmail.com>
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