[asterisk-commits] kharwell: branch 12 r403767 - in /branches/12: bridges/ channels/ include/ast...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Dec 13 12:24:52 CST 2013
Author: kharwell
Date: Fri Dec 13 12:24:47 2013
New Revision: 403767
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=403767
Log:
bridge_native_rtp: Deadlock during 4-way conference creation
The change contains a slightly adjusted patch that was on the issue
(submitted by kmoore). A fix was made by adding in a bridge lock
while calling bridge_start/stop from the framehook callback. Since
the framehook callback is not called from the bridging core the bridge
is not locked, but needs to be before calling bridge_start.
(closes issue ASTERISK-22749)
Reported by: Kinsey Moore
Review: https://reviewboard.asterisk.org/r/3066/
Patches:
lock_inversion.diff uploaded by kmoore (license 6273)
Modified:
branches/12/bridges/bridge_native_rtp.c
branches/12/channels/chan_pjsip.c
branches/12/channels/chan_sip.c
branches/12/include/asterisk/channel.h
branches/12/main/channel.c
Modified: branches/12/bridges/bridge_native_rtp.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/bridges/bridge_native_rtp.c?view=diff&rev=403767&r1=403766&r2=403767
==============================================================================
--- branches/12/bridges/bridge_native_rtp.c (original)
+++ branches/12/bridges/bridge_native_rtp.c Fri Dec 13 12:24:47 2013
@@ -112,7 +112,16 @@
return audio_glue0_res;
}
-static int native_rtp_bridge_start(struct ast_bridge *bridge, struct ast_channel *target)
+/*!
+ * \internal
+ * \brief Start native RTP bridging of two channels
+ *
+ * \param bridge The bridge that had native RTP bridging happening on it
+ * \param target If remote RTP bridging, the channel that is unheld.
+ *
+ * \note Bridge must be locked when calling this function.
+ */
+static void native_rtp_bridge_start(struct ast_bridge *bridge, struct ast_channel *target)
{
struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels);
struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
@@ -128,17 +137,11 @@
RAII_VAR(struct ast_format_cap *, cap1, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK), ast_format_cap_destroy);
if (c0 == c1) {
- return 0;
- }
-
+ return;
+ }
+
+ ast_channel_lock_both(c0->chan, c1->chan);
native_type = native_rtp_bridge_get(c0->chan, c1->chan, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
-
- if (glue0->get_codec) {
- glue0->get_codec(c0->chan, cap0);
- }
- if (glue1->get_codec) {
- glue1->get_codec(c1->chan, cap1);
- }
switch (native_type) {
case AST_RTP_GLUE_RESULT_LOCAL:
@@ -155,6 +158,12 @@
break;
case AST_RTP_GLUE_RESULT_REMOTE:
+ if (glue0->get_codec) {
+ glue0->get_codec(c0->chan, cap0);
+ }
+ if (glue1->get_codec) {
+ glue1->get_codec(c1->chan, cap1);
+ }
/* If we have a target, it's the channel that received the UNHOLD or UPDATE_RTP_PEER frame and was told to resume */
if (!target) {
@@ -180,7 +189,8 @@
break;
}
- return 0;
+ ast_channel_unlock(c0->chan);
+ ast_channel_unlock(c1->chan);
}
static void native_rtp_bridge_stop(struct ast_bridge *bridge, struct ast_channel *target)
@@ -202,6 +212,7 @@
return;
}
+ ast_channel_lock_both(c0->chan, c1->chan);
native_type = native_rtp_bridge_get(c0->chan, c1->chan, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
switch (native_type) {
@@ -241,6 +252,9 @@
ast_debug(2, "Discontinued RTP bridging of '%s' and '%s' - media will flow through Asterisk core\n",
ast_channel_name(c0->chan), ast_channel_name(c1->chan));
+
+ ast_channel_unlock(c0->chan);
+ ast_channel_unlock(c1->chan);
}
/*! \brief Frame hook that is called to intercept hold/unhold */
@@ -252,16 +266,23 @@
return f;
}
- ast_channel_lock(chan);
bridge = ast_channel_get_bridge(chan);
- ast_channel_unlock(chan);
if (bridge) {
+ /* native_rtp_bridge_start/stop are not being called from bridging
+ core so we need to lock the bridge prior to calling these functions
+ Unfortunately that means unlocking the channel, but as it
+ should not be modified this should be okay...hopefully */
+ ast_channel_unlock(chan);
+ ast_bridge_lock(bridge);
if (f->subclass.integer == AST_CONTROL_HOLD) {
native_rtp_bridge_stop(bridge, chan);
} else if ((f->subclass.integer == AST_CONTROL_UNHOLD) || (f->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
native_rtp_bridge_start(bridge, chan);
}
+ ast_bridge_unlock(bridge);
+ ast_channel_lock(chan);
+
}
return f;
@@ -412,7 +433,8 @@
return -1;
}
- return native_rtp_bridge_start(bridge, NULL);
+ native_rtp_bridge_start(bridge, NULL);
+ return 0;
}
static void native_rtp_bridge_unsuspend(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
Modified: branches/12/channels/chan_pjsip.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_pjsip.c?view=diff&rev=403767&r1=403766&r2=403767
==============================================================================
--- branches/12/channels/chan_pjsip.c (original)
+++ branches/12/channels/chan_pjsip.c Fri Dec 13 12:24:47 2013
@@ -292,14 +292,11 @@
struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_session *session = channel->session;
int changed = 0;
- struct ast_channel *bridge_peer;
/* Don't try to do any direct media shenanigans on early bridges */
- bridge_peer = ast_channel_bridge_peer(chan);
- if ((rtp || vrtp || tpeer) && !bridge_peer) {
+ if ((rtp || vrtp || tpeer) && !ast_channel_is_bridged(chan)) {
return 0;
}
- ast_channel_cleanup(bridge_peer);
if (nat_active && session->endpoint->media.direct_media.disable_on_nat) {
return 0;
Modified: branches/12/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_sip.c?view=diff&rev=403767&r1=403766&r2=403767
==============================================================================
--- branches/12/channels/chan_sip.c (original)
+++ branches/12/channels/chan_sip.c Fri Dec 13 12:24:47 2013
@@ -32667,11 +32667,8 @@
struct sip_pvt *p;
int changed = 0;
- /* Lock the channel and the private safely. */
- ast_channel_lock(chan);
p = ast_channel_tech_pvt(chan);
if (!p) {
- ast_channel_unlock(chan);
return -1;
}
sip_pvt_lock(p);
@@ -32679,7 +32676,6 @@
/* I suppose it could be argued that if this happens it is a bug. */
ast_debug(1, "The private is not owned by channel %s anymore.\n", ast_channel_name(chan));
sip_pvt_unlock(p);
- ast_channel_unlock(chan);
return 0;
}
@@ -32688,14 +32684,12 @@
!ast_channel_is_bridged(chan) &&
!sip_cfg.directrtpsetup) {
sip_pvt_unlock(p);
- ast_channel_unlock(chan);
return 0;
}
if (p->alreadygone) {
/* If we're destroyed, don't bother */
sip_pvt_unlock(p);
- ast_channel_unlock(chan);
return 0;
}
@@ -32704,7 +32698,6 @@
*/
if (nat_active && !ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
sip_pvt_unlock(p);
- ast_channel_unlock(chan);
return 0;
}
@@ -32770,7 +32763,6 @@
*/
ast_clear_flag(&p->flags[2], SIP_PAGE3_DIRECT_MEDIA_OUTGOING);
sip_pvt_unlock(p);
- ast_channel_unlock(chan);
return 0;
}
@@ -32791,7 +32783,6 @@
/* Reset lastrtprx timer */
p->lastrtprx = p->lastrtptx = time(NULL);
sip_pvt_unlock(p);
- ast_channel_unlock(chan);
return 0;
}
Modified: branches/12/include/asterisk/channel.h
URL: http://svnview.digium.com/svn/asterisk/branches/12/include/asterisk/channel.h?view=diff&rev=403767&r1=403766&r2=403767
==============================================================================
--- branches/12/include/asterisk/channel.h (original)
+++ branches/12/include/asterisk/channel.h Fri Dec 13 12:24:47 2013
@@ -4214,6 +4214,8 @@
* \note The returned peer channel is the current peer in the
* bridge when called.
*
+ * \note Absolutely _NO_ channel locks should be held when calling this function.
+ *
* \retval NULL Channel not in a bridge or the bridge is not two-party.
* \retval non-NULL Reffed peer channel at time of calling.
*/
Modified: branches/12/main/channel.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/main/channel.c?view=diff&rev=403767&r1=403766&r2=403767
==============================================================================
--- branches/12/main/channel.c (original)
+++ branches/12/main/channel.c Fri Dec 13 12:24:47 2013
@@ -6869,13 +6869,11 @@
}
ast_debug(1, "Done Masquerading %s (%d)\n", ast_channel_name(original), ast_channel_state(original));
+ ast_channel_unlock(original);
if ((bridged = ast_channel_bridge_peer(original))) {
- ast_channel_unlock(original);
ast_indicate(bridged, AST_CONTROL_SRCCHANGE);
ast_channel_unref(bridged);
- } else {
- ast_channel_unlock(original);
}
ast_indicate(original, AST_CONTROL_SRCCHANGE);
More information about the asterisk-commits
mailing list