Always specify which RTP engine is desired for a new RTP instance.

This fixes a crash reported in #asterisk-dev where chan_mgcp unexpectedly
allocated an RTP instance from res_rtp_multicast, since by not specifying an
engine, you get the first one in the list of engines.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211732 f38db490-d61c-443f-a65b-d21fe96a405b

6 files changed, 7 insertions, 7 deletions

diff --git a/channels/chan_gtalk.c b/channels/chan_gtalk.c
index 6ad6ed904..a0cc42408 100644
--- a/channels/chan_gtalk.c
+++ b/channels/chan_gtalk.c
@@ -947,7 +947,7 @@ static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const
 		tmp->initiator = 1;
 	}
 	/* clear codecs */
-	if (!(tmp->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL))) {
+	if (!(tmp->rtp = ast_rtp_instance_new("asterisk", sched, &bindaddr, NULL))) {
 	  ast_log(LOG_ERROR, "Failed to create a new RTP instance (possibly an invalid bindaddr?)\n");
 	  ast_free(tmp);
 	  return NULL;
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index a4135a976..897416a42 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -958,7 +958,7 @@ static int __oh323_rtp_create(struct oh323_pvt *pvt)
 		ast_log(LOG_ERROR, "Unable to locate local IP address for RTP stream\n");
 		return -1;
 	}
-	pvt->rtp = ast_rtp_instance_new(NULL, sched, &our_addr, NULL);
+	pvt->rtp = ast_rtp_instance_new("asterisk", sched, &our_addr, NULL);
 	if (!pvt->rtp) {
 		ast_mutex_unlock(&pvt->lock);
 		ast_log(LOG_WARNING, "Unable to create RTP session: %s\n", strerror(errno));
diff --git a/channels/chan_jingle.c b/channels/chan_jingle.c
index da98986eb..71113f621 100644
--- a/channels/chan_jingle.c
+++ b/channels/chan_jingle.c
@@ -772,7 +772,7 @@ static struct jingle_pvt *jingle_alloc(struct jingle *client, const char *from,
 		ast_copy_string(tmp->them, idroster, sizeof(tmp->them));
 		tmp->initiator = 1;
 	}
-	tmp->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+	tmp->rtp = ast_rtp_instance_new("asterisk", sched, &bindaddr, NULL);
 	tmp->parent = client;
 	if (!tmp->rtp) {
 		ast_log(LOG_WARNING, "Out of RTP sessions?\n");
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index d74fc9daf..cc0125031 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -2614,7 +2614,7 @@ static void start_rtp(struct mgcp_subchannel *sub)
 		sub->rtp = NULL;
 	}
 	/* Allocate the RTP now */
-	sub->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+	sub->rtp = ast_rtp_instance_new("asterisk", sched, &bindaddr, NULL);
 	if (sub->rtp && sub->owner)
 		ast_channel_set_fd(sub->owner, 0, ast_rtp_instance_fd(sub->rtp, 0));
 	if (sub->rtp) {
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index c289f3015..546c7eb5e 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -3632,9 +3632,9 @@ static void start_rtp(struct skinny_subchannel *sub)
 
 	ast_mutex_lock(&sub->lock);
 	/* Allocate the RTP */
-	sub->rtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+	sub->rtp = ast_rtp_instance_new("asterisk", sched, &bindaddr, NULL);
 	if (hasvideo)
-		sub->vrtp = ast_rtp_instance_new(NULL, sched, &bindaddr, NULL);
+		sub->vrtp = ast_rtp_instance_new("asterisk", sched, &bindaddr, NULL);
 
 	if (sub->rtp) {
 		ast_rtp_instance_set_prop(sub->rtp, AST_RTP_PROPERTY_RTCP, 1);
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index bb90301f9..8de4ada33 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -2061,7 +2061,7 @@ static void start_rtp(struct unistim_subchannel *sub)
 	/* Allocate the RTP */
 	if (unistimdebug)
 		ast_verb(0, "Starting RTP. Bind on %s\n", ast_inet_ntoa(sout.sin_addr));
-	sub->rtp = ast_rtp_instance_new(NULL, sched, &sout, NULL);
+	sub->rtp = ast_rtp_instance_new("asterisk", sched, &sout, NULL);
 	if (!sub->rtp) {
 		ast_log(LOG_WARNING, "Unable to create RTP session: %s binaddr=%s\n",
 				strerror(errno), ast_inet_ntoa(sout.sin_addr));