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

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/
........

Merged revisions 431157 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431158 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/main/bridge_channel.c b/main/bridge_channel.c
index 25a1a51..ac72c8b 100644 (file)
--- 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 4077caa..a3ccd19 100644 (file)
--- a/res/res_pjsip_sdp_rtp.c
+++ b/res/res_pjsip_sdp_rtp.c
@@ -1180,6 +1180,10 @@ 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;
        }
 
@@ -1199,6 +1203,9 @@ 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->remotely_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 */