Fix up some of chan_sip's usage of the RTP engine API.

The get_local_address() function for an RTP instance was used when building an
SDP, but the results were not honored.  The RTP engine activate() function was
not being used once we have determined that media will now flow.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250917 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

@@ -8791,7 +8791,19 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
 		dest->sin_port = p->redirip.sin_port;
 		dest->sin_addr = p->redirip.sin_addr;
 	} else {
-		dest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr : p->ourip.sin_addr;
+		/*
+		 * Audio Destination IP:
+		 *
+		 * 1. Specifically configured media address.
+		 * 2. Local address as specified by the RTP engine.
+		 * 3. The local IP as defined by chan_sip.
+		 *
+		 * Audio Destination Port:
+		 *
+		 * 1. Provided by the RTP engine.
+		 */
+		dest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr :
+					(sin->sin_addr.s_addr ? sin->sin_addr : p->ourip.sin_addr);
 		dest->sin_port = sin->sin_port;
 	}
 	if (needvideo) {
@@ -8800,7 +8812,19 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
 			vdest->sin_addr = p->vredirip.sin_addr;
 			vdest->sin_port = p->vredirip.sin_port;
 		} else {
-			vdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr : p->ourip.sin_addr;
+			/*
+			 * Video Destination IP:
+			 *
+			 * 1. Specifically configured media address.
+			 * 2. Local address as specified by the RTP engine.
+			 * 3. The local IP as defined by chan_sip.
+			 *
+			 * Video Destination Port:
+			 *
+			 * 1. Provided by the RTP engine.
+			 */
+			vdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr :
+						(vsin->sin_addr.s_addr ? vsin->sin_addr : p->ourip.sin_addr);
 			vdest->sin_port = vsin->sin_port;
 		}
 	}
@@ -8810,7 +8834,19 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
 			tdest->sin_addr = p->tredirip.sin_addr;
 			tdest->sin_port = p->tredirip.sin_port;
 		} else {
-			tdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr : p->ourip.sin_addr;
+			/*
+			 * Text Destination IP:
+			 *
+			 * 1. Specifically configured media address.
+			 * 2. Local address as specified by the RTP engine.
+			 * 3. The local IP as defined by chan_sip.
+			 *
+			 * Text Destination Port:
+			 *
+			 * 1. Provided by the RTP engine.
+			 */
+			tdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr :
+						(tsin->sin_addr.s_addr ? tsin->sin_addr : p->ourip.sin_addr);
 			tdest->sin_port = tsin->sin_port;
 		}
 	}
@@ -9235,6 +9271,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
 			ast_debug(1, "Setting framing from config on incoming call\n");
 			ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &p->prefs);
 		}
+		ast_rtp_instance_activate(p->rtp);
 		try_suggested_sip_codec(p);
 		if (p->t38.state == T38_ENABLED) {
 			add_sdp(&resp, p, oldsdp, TRUE, TRUE);
@@ -16671,6 +16708,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 				/* Queue a progress frame only if we have SDP in 180 or 182 */
 				ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
 			}
+			ast_rtp_instance_activate(p->rtp);
 		}
 		check_pendings(p);
 		break;
@@ -16708,6 +16746,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 				/* Queue a progress frame */
 				ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
 			}
+			ast_rtp_instance_activate(p->rtp);
 		} else {
 			/* Alcatel PBXs are known to send 183s with no SDP after sending
 			 * a 100 Trying response. We're just going to treat this sort of thing
@@ -16730,6 +16769,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
 					/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
 					/* For re-invites, we try to recover */
 					ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+			ast_rtp_instance_activate(p->rtp);
 		}
 
 		if (!req->ignore && p->owner && (get_rpid(p, req) || !reinvite)) {