From: Torrey Searle <torrey@voxbone.com>
Date: Wed, 26 Jul 2017 14:17:02 +0000 (+0200)
Subject: res_rtp_asterisk: enable rtcp & QOS stats on native bridge
X-Git-Tag: 13.18.0-rc1~100^2
X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=3608f96ea378d6b37bcf26e6ca182077140a7b90;p=thirdparty%2Fasterisk.git

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
---

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index a93bb77582..bf0d9cc829 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4854,9 +4854,6 @@ static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance,
 		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");
@@ -5092,13 +5089,6 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		}
 	}
 
-	/* 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;
@@ -5203,6 +5193,28 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
 		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),