chan_sip.c: Fix channel staging assertion failure.

The failing assertion ensures that the final snapshot gets generated so
CDR records can get finalized.  The only place where a channel staging
snapshot flag could be left set is in chan_sip.c:handle_request_bye().
The function could return before clearing the flag because the channel
could dissappear while the function had to have the channel unlocked.

* Fixed handle_request_bye() channel snapshot staging coverage area to not
have a return in the middle of it and be unable to clear the staging flag.

* Pushed the channel snapshot staging coverage area into
ast_rtp_instance_set_stats_vars() to ensure that the staging is not
interrutped.

* Made callers of ast_rtp_instance_set_stats_vars() not call it with any
channels or channel driver private locks held to eliminate the deadlock
potential.  The callers must hold references to the passed in channel and
rtp objects.

* Eliminated sip_hangup() trying to get the bridge peer.  It is futile at
this point because the channel could never be in a bridge.

Review: https://reviewboard.asterisk.org/r/3431/
........

Merged revisions 412385 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412386 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 35 insertions, 7 deletions

diff --git a/main/rtp_engine.c b/main/rtp_engine.c
index fd058a403..48372303a 100644
--- a/main/rtp_engine.c
+++ b/main/rtp_engine.c
@@ -1305,36 +1305,64 @@ char *ast_rtp_instance_get_quality(struct ast_rtp_instance *instance, enum ast_r
 
 void ast_rtp_instance_set_stats_vars(struct ast_channel *chan, struct ast_rtp_instance *instance)
 {
-	char quality_buf[AST_MAX_USER_FIELD], *quality;
-	RAII_VAR(struct ast_channel *, bridge, ast_channel_bridge_peer(chan), ast_channel_cleanup);
-
-	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY, quality_buf, sizeof(quality_buf)))) {
+	char quality_buf[AST_MAX_USER_FIELD];
+	char *quality;
+	struct ast_channel *bridge = ast_channel_bridge_peer(chan);
+
+	ast_channel_lock(chan);
+	ast_channel_stage_snapshot(chan);
+	ast_channel_unlock(chan);
+	if (bridge) {
+		ast_channel_lock(bridge);
+		ast_channel_stage_snapshot(bridge);
+		ast_channel_unlock(bridge);
+	}
+
+	quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY,
+		quality_buf, sizeof(quality_buf));
+	if (quality) {
 		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOS", quality);
 		if (bridge) {
 			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSBRIDGED", quality);
 		}
 	}
 
-	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf)))) {
+	quality = ast_rtp_instance_get_quality(instance,
+		AST_RTP_INSTANCE_STAT_FIELD_QUALITY_JITTER, quality_buf, sizeof(quality_buf));
+	if (quality) {
 		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSJITTER", quality);
 		if (bridge) {
 			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSJITTERBRIDGED", quality);
 		}
 	}
 
-	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf)))) {
+	quality = ast_rtp_instance_get_quality(instance,
+		AST_RTP_INSTANCE_STAT_FIELD_QUALITY_LOSS, quality_buf, sizeof(quality_buf));
+	if (quality) {
 		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSLOSS", quality);
 		if (bridge) {
 			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSLOSSBRIDGED", quality);
 		}
 	}
 
-	if ((quality = ast_rtp_instance_get_quality(instance, AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf)))) {
+	quality = ast_rtp_instance_get_quality(instance,
+		AST_RTP_INSTANCE_STAT_FIELD_QUALITY_RTT, quality_buf, sizeof(quality_buf));
+	if (quality) {
 		pbx_builtin_setvar_helper(chan, "RTPAUDIOQOSRTT", quality);
 		if (bridge) {
 			pbx_builtin_setvar_helper(bridge, "RTPAUDIOQOSRTTBRIDGED", quality);
 		}
 	}
+
+	ast_channel_lock(chan);
+	ast_channel_stage_snapshot_done(chan);
+	ast_channel_unlock(chan);
+	if (bridge) {
+		ast_channel_lock(bridge);
+		ast_channel_stage_snapshot_done(bridge);
+		ast_channel_unlock(bridge);
+		ast_channel_unref(bridge);
+	}
 }
 
 int ast_rtp_instance_set_read_format(struct ast_rtp_instance *instance, struct ast_format *format)