[Asterisk-code-review] Crash during "pjsip show channelstats" execution (...asterisk[13])

[Salah Ahmed]

Salah Ahmed has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/13044


Change subject: Crash during "pjsip show channelstats" execution
......................................................................

Crash during "pjsip show channelstats" execution

During execution "pjsip show channelstats" cli command by an
external module asterisk crashed. It seems this is a separate
thread running to fetch and print rtp stats. The crash happened on
the ao2_lock method, just before it going to read the rtp stats on
a rtp instance. According to gdb backtrace log, it seems the
session media was already cleaned up at that moment.

ASTERISK-28578

Change-Id: Ia918bfa3c8119060633e39b14852e8d983e0417e
---
M channels/pjsip/cli_commands.c
1 file changed, 26 insertions(+), 10 deletions(-)



  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/44/13044/1

diff --git a/channels/pjsip/cli_commands.c b/channels/pjsip/cli_commands.c
index 514cf70..b3cf8f6 100644
--- a/channels/pjsip/cli_commands.c
+++ b/channels/pjsip/cli_commands.c
@@ -343,39 +343,54 @@
 	struct ast_sip_cli_context *context = arg;
 	const struct ast_channel_snapshot *snapshot = obj;
 	struct ast_channel *channel = ast_channel_get_by_name(snapshot->name);
-	struct ast_sip_channel_pvt *cpvt = channel ? ast_channel_tech_pvt(channel) : NULL;
-	struct chan_pjsip_pvt *pvt = cpvt ? cpvt->pvt : NULL;
-	struct ast_sip_session_media *media = pvt ? pvt->media[SIP_MEDIA_AUDIO] : NULL;
+	struct ast_sip_channel_pvt *cpvt = NULL;
+	struct chan_pjsip_pvt *pvt = NULL;
+	struct ast_sip_session_media *media = NULL;
+	struct ast_rtp_instance *rtp = NULL;
 	struct ast_rtp_instance_stats stats;
 	char *print_name = NULL;
 	char *print_time = alloca(32);
 	char codec_in_use[7];
+	int stats_res = -1;
 
 	ast_assert(context->output_buffer != NULL);
 
+	if (!channel) {
+		ast_str_append(&context->output_buffer, 0, " %s not valid\n", snapshot->name);
+		return -1;
+	}
+
+	ast_channel_lock(channel);
+
+	cpvt = ast_channel_tech_pvt(channel);
+	pvt = cpvt ? cpvt->pvt : NULL;
+	media = pvt ? pvt->media[SIP_MEDIA_AUDIO] : NULL;
+
 	if (!media || !media->rtp) {
 		ast_str_append(&context->output_buffer, 0, " %s not valid\n", snapshot->name);
+		ast_channel_unlock(channel);
 		ao2_cleanup(channel);
 		return 0;
 	}
 
+	rtp = ao2_bump(media->rtp);
 	codec_in_use[0] = '\0';
 
-	if (channel) {
-		ast_channel_lock(channel);
-		if (ast_channel_rawreadformat(channel)) {
-			ast_copy_string(codec_in_use, ast_format_get_name(ast_channel_rawreadformat(channel)), sizeof(codec_in_use));
-		}
-		ast_channel_unlock(channel);
+	if (ast_channel_rawreadformat(channel)) {
+		ast_copy_string(codec_in_use, ast_format_get_name(ast_channel_rawreadformat(channel)), sizeof(codec_in_use));
 	}
 
+	stats_res = ast_rtp_instance_get_stats(rtp, &stats, AST_RTP_INSTANCE_STAT_ALL);
+
+	ast_channel_unlock(channel);
+
 	print_name = ast_strdupa(snapshot->name);
 	/* Skip the PJSIP/.  We know what channel type it is and we need the space. */
 	print_name += 6;
 
 	ast_format_duration_hh_mm_ss(ast_tvnow().tv_sec - snapshot->creationtime.tv_sec, print_time, 32);
 
-	if (ast_rtp_instance_get_stats(media->rtp, &stats, AST_RTP_INSTANCE_STAT_ALL)) {
+	if (stats_res == -1) {
 		ast_str_append(&context->output_buffer, 0, "%s direct media\n", snapshot->name);
 	} else {
 		ast_str_append(&context->output_buffer, 0,
@@ -400,6 +415,7 @@
 		);
 	}
 
+	ao2_cleanup(rtp);
 	ao2_cleanup(channel);
 
 	return 0;

-- 
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Change-Id: Ia918bfa3c8119060633e39b14852e8d983e0417e
Gerrit-Change-Number: 13044
Gerrit-PatchSet: 1
Gerrit-Owner: Salah Ahmed <txrubel at gmail.com>
Gerrit-MessageType: newchange
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