bridge / res_pjsip_sdp_rtp: Fix issues with media not being reinvited during direct media.

This change fixes two issues:

1. During a swap operation bridging added the new channel before having the swap channel
leave. This was not handled in bridge_native_rtp and could result in a channel not getting
reinvited back to Asterisk. After this change the swap channel will leave first and the
new channel will then join.

2. If a re-invite was received after a session had been established any upstream elements
(such as bridge_native_rtp) were not notified that they may want to re-evaluate things.
After this change an UPDATE_RTP_PEER control frame is queued when this situation occurs
and upstream can react.

AST-1524 #close

Review: https://reviewboard.asterisk.org/r/4378/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431157 65c4cc65-6c06-0410-ace0-fbb531ad65f3

2 files changed, 20 insertions, 4 deletions

diff --git a/main/bridge_channel.c b/main/bridge_channel.c
index 25a1a5153..ac72c8bc2 100644
--- a/main/bridge_channel.c
+++ b/main/bridge_channel.c
@@ -1994,6 +1994,19 @@ int bridge_channel_internal_push(struct ast_bridge_channel *bridge_channel)
 			bridge->uniqueid, bridge_channel, ast_channel_name(bridge_channel->chan));
 		return -1;
 	}
+
+	if (swap) {
+		int dissolve = ast_test_flag(&bridge->feature_flags, AST_BRIDGE_FLAG_DISSOLVE_EMPTY);
+
+		/* This flag is cleared so the act of this channel leaving does not cause it to dissolve if need be */
+		ast_clear_flag(&bridge->feature_flags, AST_BRIDGE_FLAG_DISSOLVE_EMPTY);
+
+		ast_bridge_channel_leave_bridge(swap, BRIDGE_CHANNEL_STATE_END_NO_DISSOLVE, 0);
+		bridge_channel_internal_pull(swap);
+
+		ast_set2_flag(&bridge->feature_flags, dissolve, AST_BRIDGE_FLAG_DISSOLVE_EMPTY);
+	}
+
 	bridge_channel->in_bridge = 1;
 	bridge_channel->just_joined = 1;
 	AST_LIST_INSERT_TAIL(&bridge->channels, bridge_channel, entry);
@@ -2015,10 +2028,6 @@ int bridge_channel_internal_push(struct ast_bridge_channel *bridge_channel)
 		bridge->uniqueid);
 
 	ast_bridge_publish_enter(bridge, bridge_channel->chan, swap ? swap->chan : NULL);
-	if (swap) {
-		ast_bridge_channel_leave_bridge(swap, BRIDGE_CHANNEL_STATE_END_NO_DISSOLVE, 0);
-		bridge_channel_internal_pull(swap);
-	}
 
 	/* Clear any BLINDTRANSFER and ATTENDEDTRANSFER since the transfer has completed. */
 	pbx_builtin_setvar_helper(bridge_channel->chan, "BLINDTRANSFER", NULL);
diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c
index 0f5eec8ef..f7dbf08e6 100644
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -1179,6 +1179,10 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a
 
 	/* audio stream handles music on hold */
 	if (media_type != AST_MEDIA_TYPE_AUDIO) {
+		if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
+			&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
+			ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
+		}
 		return 1;
 	}
 
@@ -1198,6 +1202,9 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session, struct a
 		ast_queue_unhold(session->channel);
 		ast_queue_frame(session->channel, &ast_null_frame);
 		session_media->held = 0;
+	} else if ((pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE)
+		&& (session->inv_session->state == PJSIP_INV_STATE_CONFIRMED)) {
+		ast_queue_control(session->channel, AST_CONTROL_UPDATE_RTP_PEER);
 	}
 
 	/* This purposely resets the encryption to the configured in case it gets added later */