asterisk/main/core_local.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013 Digium, Inc.
*
* Richard Mudgett <rmudgett@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*!
* \file
* \brief Local proxy channel driver.
*
* \author Richard Mudgett <rmudgett@digium.com>
*
* See Also:
* \arg \ref AstCREDITS
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
/* ------------------------------------------------------------------- */
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/cli.h"
#include "asterisk/manager.h"
#include "asterisk/devicestate.h"
#include "asterisk/astobj2.h"
#include "asterisk/bridge.h"
#include "asterisk/core_unreal.h"
#include "asterisk/core_local.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/_private.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/stream.h"
#include "asterisk/translate.h"
/*** DOCUMENTATION
<manager name="LocalOptimizeAway" language="en_US">
<synopsis>
Optimize away a local channel when possible.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
<parameter name="Channel" required="true">
<para>The channel name to optimize away.</para>
</parameter>
</syntax>
<description>
<para>A local channel created with "/n" will not automatically optimize away.
Calling this command on the local channel will clear that flag and allow
it to optimize away if it's bridged or when it becomes bridged.</para>
</description>
</manager>
<managerEvent language="en_US" name="LocalBridge">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when two halves of a Local Channel form a bridge.</synopsis>
<syntax>
<channel_snapshot prefix="LocalOne"/>
<channel_snapshot prefix="LocalTwo"/>
<parameter name="Context">
<para>The context in the dialplan that Channel2 starts in.</para>
</parameter>
<parameter name="Exten">
<para>The extension in the dialplan that Channel2 starts in.</para>
</parameter>
<parameter name="LocalOptimization">
<enumlist>
<enum name="Yes"/>
<enum name="No"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="LocalOptimizationBegin">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when two halves of a Local Channel begin to optimize
themselves out of the media path.</synopsis>
<syntax>
<channel_snapshot prefix="LocalOne"/>
<channel_snapshot prefix="LocalTwo"/>
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
<channel_snapshot prefix="Source"/>
<parameter name="DestUniqueId">
<para>The unique ID of the bridge into which the local channel is optimizing.</para>
</parameter>
<parameter name="Id">
<para>Identification for the optimization operation.</para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">LocalOptimizationEnd</ref>
<ref type="manager">LocalOptimizeAway</ref>
</see-also>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="LocalOptimizationEnd">
<managerEventInstance class="EVENT_FLAG_CALL">
<synopsis>Raised when two halves of a Local Channel have finished optimizing
themselves out of the media path.</synopsis>
<syntax>
<channel_snapshot prefix="LocalOne"/>
<channel_snapshot prefix="LocalTwo"/>
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
<parameter name="Success">
<para>Indicates whether the local optimization succeeded.</para>
</parameter>
<parameter name="Id">
<para>Identification for the optimization operation. Matches the <replaceable>Id</replaceable>
from a previous <literal>LocalOptimizationBegin</literal></para>
</parameter>
</syntax>
<see-also>
<ref type="managerEvent">LocalOptimizationBegin</ref>
<ref type="manager">LocalOptimizeAway</ref>
</see-also>
</managerEventInstance>
</managerEvent>
***/
static const char tdesc[] = "Local Proxy Channel Driver";
static struct ao2_container *locals;
static struct ast_channel *local_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
static struct ast_channel *local_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
static int local_call(struct ast_channel *ast, const char *dest, int timeout);
static int local_hangup(struct ast_channel *ast);
static int local_devicestate(const char *data);
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
static void local_optimization_started_cb(struct ast_unreal_pvt *base, struct ast_channel *source,
enum ast_unreal_channel_indicator dest, unsigned int id);
static void local_optimization_finished_cb(struct ast_unreal_pvt *base, int success, unsigned int id);
static int local_setoption(struct ast_channel *chan, int option, void *data, int datalen);
static struct ast_manager_event_blob *local_message_to_ami(struct stasis_message *msg);
/*!
* @{ \brief Define local channel message types.
*/
STASIS_MESSAGE_TYPE_DEFN(ast_local_bridge_type,
.to_ami = local_message_to_ami,
);
STASIS_MESSAGE_TYPE_DEFN(ast_local_optimization_begin_type,
.to_ami = local_message_to_ami,
);
STASIS_MESSAGE_TYPE_DEFN(ast_local_optimization_end_type,
.to_ami = local_message_to_ami,
);
/*! @} */
/*! \brief Callbacks from the unreal core when channel optimization occurs */
struct ast_unreal_pvt_callbacks local_unreal_callbacks = {
.optimization_started = local_optimization_started_cb,
.optimization_finished = local_optimization_finished_cb,
};
/* PBX interface structure for channel registration */
static struct ast_channel_tech local_tech = {
.type = "Local",
.description = tdesc,
.requester = local_request,
.requester_with_stream_topology = local_request_with_stream_topology,
.send_digit_begin = ast_unreal_digit_begin,
.send_digit_end = ast_unreal_digit_end,
.call = local_call,
.hangup = local_hangup,
.answer = ast_unreal_answer,
.read_stream = ast_unreal_read,
.write = ast_unreal_write,
.write_stream = ast_unreal_write_stream,
.exception = ast_unreal_read,
.indicate = ast_unreal_indicate,
.fixup = ast_unreal_fixup,
.send_html = ast_unreal_sendhtml,
.send_text = ast_unreal_sendtext,
.devicestate = local_devicestate,
.queryoption = ast_unreal_queryoption,
.setoption = local_setoption,
};
/*! What to do with the ;2 channel when ast_call() happens. */
enum local_call_action {
/* The ast_call() will run dialplan on the ;2 channel. */
LOCAL_CALL_ACTION_DIALPLAN,
/* The ast_call() will impart the ;2 channel into a bridge. */
LOCAL_CALL_ACTION_BRIDGE,
/* The ast_call() will masquerade the ;2 channel into a channel. */
LOCAL_CALL_ACTION_MASQUERADE,
};
/*! Join a bridge on ast_call() parameters. */
struct local_bridge {
/*! Bridge to join. */
struct ast_bridge *join;
/*! Channel to swap with when joining bridge. */
struct ast_channel *swap;
/*! Features that are specific to this channel when pushed into the bridge. */
struct ast_bridge_features *features;
};
/*!
* \brief the local pvt structure for all channels
*
* The local channel pvt has two ast_chan objects - the "owner" and the "next channel", the outbound channel
*
* ast_chan owner -> local_pvt -> ast_chan chan
*/
struct local_pvt {
/*! Unreal channel driver base class values. */
struct ast_unreal_pvt base;
/*! Additional action arguments */
union {
/*! Make ;2 join a bridge on ast_call(). */
struct local_bridge bridge;
/*! Make ;2 masquerade into this channel on ast_call(). */
struct ast_channel *masq;
} action;
/*! What to do with the ;2 channel on ast_call(). */
enum local_call_action type;
/*! Context to call */
char context[AST_MAX_CONTEXT];
/*! Extension to call */
char exten[AST_MAX_EXTENSION];
};
void ast_local_lock_all(struct ast_channel *chan, void **tech_pvt,
struct ast_channel **base_chan, struct ast_channel **base_owner)
bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
2016-03-01 22:18:21 +00:00
{
struct local_pvt *p = ast_channel_tech_pvt(chan);
*tech_pvt = NULL;
*base_chan = NULL;
*base_owner = NULL;
bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
2016-03-01 22:18:21 +00:00
if (p) {
*tech_pvt = ao2_bump(p);
ast_unreal_lock_all(&p->base, base_chan, base_owner);
bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
2016-03-01 22:18:21 +00:00
}
}
void ast_local_unlock_all(void *tech_pvt, struct ast_channel *base_chan,
struct ast_channel *base_owner)
bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
2016-03-01 22:18:21 +00:00
{
if (base_chan) {
ast_channel_unlock(base_chan);
ast_channel_unref(base_chan);
bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
2016-03-01 22:18:21 +00:00
}
if (base_owner) {
ast_channel_unlock(base_owner);
ast_channel_unref(base_owner);
bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
2016-03-01 22:18:21 +00:00
}
if (tech_pvt) {
struct local_pvt *p = tech_pvt;
ao2_unlock(&p->base);
ao2_ref(tech_pvt, -1);
bridge.c: Crash during attended transfer when missing a local channel half It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880
2016-03-01 22:18:21 +00:00
}
}
struct ast_channel *ast_local_get_peer(struct ast_channel *ast)
{
struct local_pvt *p = ast_channel_tech_pvt(ast);
struct local_pvt *found;
struct ast_channel *peer;
if (!p) {
return NULL;
}
found = p ? ao2_find(locals, p, 0) : NULL;
if (!found) {
/* ast is either not a local channel or it has alredy been hungup */
return NULL;
}
ao2_lock(found);
if (ast == p->base.owner) {
peer = p->base.chan;
} else if (ast == p->base.chan) {
peer = p->base.owner;
} else {
peer = NULL;
}
if (peer) {
ast_channel_ref(peer);
}
ao2_unlock(found);
ao2_ref(found, -1);
return peer;
}
/*! \brief Adds devicestate to local channels */
static int local_devicestate(const char *data)
{
int is_inuse = 0;
int res = AST_DEVICE_INVALID;
char *exten = ast_strdupa(data);
char *context;
char *opts;
struct local_pvt *lp;
struct ao2_iterator it;
/* Strip options if they exist */
opts = strchr(exten, '/');
if (opts) {
*opts = '\0';
}
context = strchr(exten, '@');
if (!context) {
ast_log(LOG_WARNING,
"Someone used Local/%s somewhere without a @context. This is bad.\n", data);
return AST_DEVICE_INVALID;
}
*context++ = '\0';
it = ao2_iterator_init(locals, 0);
for (; (lp = ao2_iterator_next(&it)); ao2_ref(lp, -1)) {
ao2_lock(lp);
if (!strcmp(exten, lp->exten)
&& !strcmp(context, lp->context)) {
res = AST_DEVICE_NOT_INUSE;
if (lp->base.owner
&& ast_test_flag(&lp->base, AST_UNREAL_CARETAKER_THREAD)) {
is_inuse = 1;
}
}
ao2_unlock(lp);
if (is_inuse) {
res = AST_DEVICE_INUSE;
ao2_ref(lp, -1);
break;
}
}
ao2_iterator_destroy(&it);
if (res == AST_DEVICE_INVALID) {
ast_debug(3, "Checking if extension %s@%s exists (devicestate)\n", exten, context);
if (ast_exists_extension(NULL, context, exten, 1, NULL)) {
res = AST_DEVICE_NOT_INUSE;
}
}
return res;
}
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
static struct ast_multi_channel_blob *local_channel_optimization_blob(struct local_pvt *p,
struct ast_json *json_object)
{
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
struct ast_multi_channel_blob *payload;
RAII_VAR(struct ast_channel_snapshot *, local_one_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, local_two_snapshot, NULL, ao2_cleanup);
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
local_one_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(p->base.owner));
if (!local_one_snapshot) {
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
return NULL;
}
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
local_two_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(p->base.chan));
if (!local_two_snapshot) {
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
return NULL;
}
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
payload = ast_multi_channel_blob_create(json_object);
if (!payload) {
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
return NULL;
}
ast_multi_channel_blob_add_channel(payload, "1", local_one_snapshot);
ast_multi_channel_blob_add_channel(payload, "2", local_two_snapshot);
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
return payload;
}
/*! \brief Callback for \ref ast_unreal_pvt_callbacks \p optimization_started */
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
static void local_optimization_started_cb(struct ast_unreal_pvt *base, struct ast_channel *source,
enum ast_unreal_channel_indicator dest, unsigned int id)
{
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
RAII_VAR(struct ast_json *, json_object, ast_json_null(), ast_json_unref);
RAII_VAR(struct ast_multi_channel_blob *, payload, NULL, ao2_cleanup);
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
struct local_pvt *p = (struct local_pvt *)base;
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
if (!ast_local_optimization_begin_type()) {
return;
}
json_object = ast_json_pack("{s: i, s: I}",
"dest", dest, "id", (ast_json_int_t)id);
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
if (!json_object) {
return;
}
payload = local_channel_optimization_blob(p, json_object);
if (!payload) {
return;
}
if (source) {
RAII_VAR(struct ast_channel_snapshot *, source_snapshot, NULL, ao2_cleanup);
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
source_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(source));
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
if (!source_snapshot) {
return;
}
ast_multi_channel_blob_add_channel(payload, "source", source_snapshot);
}
msg = stasis_message_create(ast_local_optimization_begin_type(), payload);
if (!msg) {
return;
}
stasis_publish(ast_channel_topic(p->base.owner), msg);
}
/*! \brief Callback for \ref ast_unreal_pvt_callbacks \p optimization_finished */
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
static void local_optimization_finished_cb(struct ast_unreal_pvt *base, int success, unsigned int id)
{
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
RAII_VAR(struct ast_json *, json_object, ast_json_null(), ast_json_unref);
RAII_VAR(struct ast_multi_channel_blob *, payload, NULL, ao2_cleanup);
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
struct local_pvt *p = (struct local_pvt *)base;
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
if (!ast_local_optimization_end_type()) {
return;
}
json_object = ast_json_pack("{s: i, s: I}", "success", success, "id", (ast_json_int_t)id);
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
if (!json_object) {
return;
}
payload = local_channel_optimization_blob(p, json_object);
if (!payload) {
return;
}
msg = stasis_message_create(ast_local_optimization_end_type(), payload);
if (!msg) {
return;
}
stasis_publish(ast_channel_topic(p->base.owner), msg);
}
static struct ast_manager_event_blob *local_message_to_ami(struct stasis_message *message)
{
struct ast_multi_channel_blob *obj = stasis_message_data(message);
struct ast_json *blob = ast_multi_channel_blob_get_json(obj);
struct ast_channel_snapshot *local_snapshot_one;
struct ast_channel_snapshot *local_snapshot_two;
RAII_VAR(struct ast_str *, local_channel_one, NULL, ast_free);
RAII_VAR(struct ast_str *, local_channel_two, NULL, ast_free);
RAII_VAR(struct ast_str *, event_buffer, NULL, ast_free);
const char *event;
local_snapshot_one = ast_multi_channel_blob_get_channel(obj, "1");
local_snapshot_two = ast_multi_channel_blob_get_channel(obj, "2");
if (!local_snapshot_one || !local_snapshot_two) {
return NULL;
}
event_buffer = ast_str_create(1024);
local_channel_one = ast_manager_build_channel_state_string_prefix(local_snapshot_one, "LocalOne");
local_channel_two = ast_manager_build_channel_state_string_prefix(local_snapshot_two, "LocalTwo");
if (!event_buffer || !local_channel_one || !local_channel_two) {
return NULL;
}
if (stasis_message_type(message) == ast_local_optimization_begin_type()) {
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
struct ast_channel_snapshot *source_snapshot;
RAII_VAR(struct ast_str *, source_str, NULL, ast_free);
const char *dest_uniqueid;
source_snapshot = ast_multi_channel_blob_get_channel(obj, "source");
if (source_snapshot) {
source_str = ast_manager_build_channel_state_string_prefix(source_snapshot, "Source");
if (!source_str) {
return NULL;
}
}
dest_uniqueid = ast_json_object_get(blob, "dest") == AST_UNREAL_OWNER ?
local_snapshot_one->base->uniqueid : local_snapshot_two->base->uniqueid;
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
event = "LocalOptimizationBegin";
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
if (source_str) {
ast_str_append(&event_buffer, 0, "%s", ast_str_buffer(source_str));
}
ast_str_append(&event_buffer, 0, "DestUniqueId: %s\r\n", dest_uniqueid);
ast_str_append(&event_buffer, 0, "Id: %u\r\n", (unsigned int) ast_json_integer_get(ast_json_object_get(blob, "id")));
} else if (stasis_message_type(message) == ast_local_optimization_end_type()) {
event = "LocalOptimizationEnd";
Massively clean up app_queue. This essentially makes app_queue usable again. From reviewboard: * Reporting of transfers and call completion is done by creating stasis subscriptions and listening for specific events in order to determine when the call is finished (either via a transfer or hangup). * Dial end messages have been added where they were previously missing. * Queue stats are properly being updated again once calls have finished. * AgentComplete stasis messages and AMI events are now occurring again. * Mixmonitor starting has been factored into its own function and uses the Mixmonitor API now instead of using ast_pbx_run() In addition to the changes in app_queue, there are several supplementary changes as well: * Queue logging now differentiates between attended and blind transfers. A note about this is in the CHANGES file. * Local channel optimization events now report more information. This includes which of the two local channels involved is the destination of the optimization, the channel that is replacing the destination local channel, and an identifier so that begin and end events can be matched to each other. The end events are now sent whether the optimization was successful or not and includes an indicator of whether the optimization was successful. * Changes were made to features and bridging_basic so that additional flags may be set on a bridge. This is necessary because the queue requires that its bridge only allows move-swap local channel optimizations into the bridge. (closes issue ASTERISK-21517) Reported by Matt Jordan (closes issue ASTERISK-21943) Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2694 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397451 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-22 18:52:41 +00:00
ast_str_append(&event_buffer, 0, "Success: %s\r\n", ast_json_integer_get(ast_json_object_get(blob, "success")) ? "Yes" : "No");
ast_str_append(&event_buffer, 0, "Id: %u\r\n", (unsigned int) ast_json_integer_get(ast_json_object_get(blob, "id")));
} else if (stasis_message_type(message) == ast_local_bridge_type()) {
event = "LocalBridge";
ast_str_append(&event_buffer, 0, "Context: %s\r\n", ast_json_string_get(ast_json_object_get(blob, "context")));
ast_str_append(&event_buffer, 0, "Exten: %s\r\n", ast_json_string_get(ast_json_object_get(blob, "exten")));
ast_str_append(&event_buffer, 0, "LocalOptimization: %s\r\n", ast_json_is_true(ast_json_object_get(blob, "can_optimize")) ? "Yes" : "No");
} else {
return NULL;
}
return ast_manager_event_blob_create(EVENT_FLAG_CALL, event,
"%s"
"%s"
"%s",
ast_str_buffer(local_channel_one),
ast_str_buffer(local_channel_two),
ast_str_buffer(event_buffer));
}
/*!
* \internal
* \brief Post the \ref ast_local_bridge_type \ref stasis message
* \since 12.0.0
*
* \param p local_pvt to raise the local bridge message
*/
static void publish_local_bridge_message(struct local_pvt *p)
{
RAII_VAR(struct ast_multi_channel_blob *, multi_blob, NULL, ao2_cleanup);
RAII_VAR(struct ast_json *, blob, NULL, ast_json_unref);
RAII_VAR(struct stasis_message *, msg, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, one_snapshot, NULL, ao2_cleanup);
RAII_VAR(struct ast_channel_snapshot *, two_snapshot, NULL, ao2_cleanup);
struct ast_channel *owner;
struct ast_channel *chan;
if (!ast_local_bridge_type()) {
return;
}
ast_unreal_lock_all(&p->base, &chan, &owner);
blob = ast_json_pack("{s: s, s: s, s: b}",
"context", p->context,
"exten", p->exten,
"can_optimize", !ast_test_flag(&p->base, AST_UNREAL_NO_OPTIMIZATION));
if (!blob) {
goto end;
}
multi_blob = ast_multi_channel_blob_create(blob);
if (!multi_blob) {
goto end;
}
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
one_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(owner));
if (!one_snapshot) {
goto end;
}
stasis: Reduce creation of channel snapshots to improve performance During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-13 18:24:49 +00:00
two_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(chan));
if (!two_snapshot) {
goto end;
}
ast_multi_channel_blob_add_channel(multi_blob, "1", one_snapshot);
ast_multi_channel_blob_add_channel(multi_blob, "2", two_snapshot);
msg = stasis_message_create(ast_local_bridge_type(), multi_blob);
if (!msg) {
goto end;
}
stasis_publish(ast_channel_topic(owner), msg);
end:
ast_channel_unlock(owner);
ast_channel_unref(owner);
ast_channel_unlock(chan);
ast_channel_unref(chan);
ao2_unlock(&p->base);
}
int ast_local_setup_bridge(struct ast_channel *ast, struct ast_bridge *bridge, struct ast_channel *swap, struct ast_bridge_features *features)
{
struct local_pvt *p;
struct local_pvt *found;
int res = -1;
/* Sanity checks. */
if (!ast || !bridge) {
ast_bridge_features_destroy(features);
return -1;
}
ast_channel_lock(ast);
p = ast_channel_tech_pvt(ast);
ast_channel_unlock(ast);
found = p ? ao2_find(locals, p, 0) : NULL;
if (found) {
ao2_lock(found);
if (found->type == LOCAL_CALL_ACTION_DIALPLAN
&& found->base.owner
&& found->base.chan
&& !ast_test_flag(&found->base, AST_UNREAL_CARETAKER_THREAD)) {
ao2_ref(bridge, +1);
if (swap) {
ast_channel_ref(swap);
}
found->type = LOCAL_CALL_ACTION_BRIDGE;
found->action.bridge.join = bridge;
found->action.bridge.swap = swap;
found->action.bridge.features = features;
res = 0;
} else {
ast_bridge_features_destroy(features);
}
ao2_unlock(found);
ao2_ref(found, -1);
}
return res;
}
int ast_local_setup_masquerade(struct ast_channel *ast, struct ast_channel *masq)
{
struct local_pvt *p;
struct local_pvt *found;
int res = -1;
/* Sanity checks. */
if (!ast || !masq) {
return -1;
}
ast_channel_lock(ast);
p = ast_channel_tech_pvt(ast);
ast_channel_unlock(ast);
found = p ? ao2_find(locals, p, 0) : NULL;
if (found) {
ao2_lock(found);
if (found->type == LOCAL_CALL_ACTION_DIALPLAN
&& found->base.owner
&& found->base.chan
&& !ast_test_flag(&found->base, AST_UNREAL_CARETAKER_THREAD)) {
ast_channel_ref(masq);
found->type = LOCAL_CALL_ACTION_MASQUERADE;
found->action.masq = masq;
res = 0;
}
ao2_unlock(found);
ao2_ref(found, -1);
}
return res;
}
/*! \brief Initiate new call, part of PBX interface
* dest is the dial string */
static int local_call(struct ast_channel *ast, const char *dest, int timeout)
{
struct local_pvt *p = ast_channel_tech_pvt(ast);
int pvt_locked = 0;
struct ast_channel *owner = NULL;
struct ast_channel *chan = NULL;
int res;
char *reduced_dest = ast_strdupa(dest);
char *slash;
const char *chan_cid;
if (!p) {
return -1;
}
/* since we are letting go of channel locks that were locked coming into
* this function, then we need to give the tech pvt a ref */
ao2_ref(p, 1);
ast_channel_unlock(ast);
ast_unreal_lock_all(&p->base, &chan, &owner);
pvt_locked = 1;
if (owner != ast) {
res = -1;
goto return_cleanup;
}
if (!owner || !chan) {
res = -1;
goto return_cleanup;
}
ast_unreal_call_setup(owner, chan);
/*
* If the local channel has /n on the end of it, we need to lop
* that off for our argument to setting up the CC_INTERFACES
* variable.
*/
if ((slash = strrchr(reduced_dest, '/'))) {
*slash = '\0';
}
ast_set_cc_interfaces_chanvar(chan, reduced_dest);
ao2_unlock(p);
pvt_locked = 0;
ast_channel_unlock(owner);
chan_cid = S_COR(ast_channel_caller(chan)->id.number.valid,
ast_channel_caller(chan)->id.number.str, NULL);
if (chan_cid) {
chan_cid = ast_strdupa(chan_cid);
}
ast_channel_unlock(chan);
res = -1;
switch (p->type) {
case LOCAL_CALL_ACTION_DIALPLAN:
if (!ast_exists_extension(NULL, p->context, p->exten, 1, chan_cid)) {
ast_log(LOG_NOTICE, "No such extension/context %s@%s while calling Local channel\n",
p->exten, p->context);
} else {
publish_local_bridge_message(p);
/* Start switch on sub channel */
res = ast_pbx_start(chan);
}
break;
case LOCAL_CALL_ACTION_BRIDGE:
publish_local_bridge_message(p);
ast_answer(chan);
res = ast_bridge_impart(p->action.bridge.join, chan, p->action.bridge.swap,
p->action.bridge.features, AST_BRIDGE_IMPART_CHAN_INDEPENDENT);
ao2_ref(p->action.bridge.join, -1);
p->action.bridge.join = NULL;
ao2_cleanup(p->action.bridge.swap);
p->action.bridge.swap = NULL;
p->action.bridge.features = NULL;
break;
case LOCAL_CALL_ACTION_MASQUERADE:
publish_local_bridge_message(p);
ast_answer(chan);
res = ast_channel_move(p->action.masq, chan);
if (!res) {
/* Chan is now an orphaned zombie. Destroy it. */
ast_hangup(chan);
}
p->action.masq = ast_channel_unref(p->action.masq);
break;
}
if (!res) {
ao2_lock(p);
ast_set_flag(&p->base, AST_UNREAL_CARETAKER_THREAD);
ao2_unlock(p);
}
/* we already unlocked them, clear them here so the cleanup label won't touch them. */
owner = ast_channel_unref(owner);
chan = ast_channel_unref(chan);
return_cleanup:
if (p) {
if (pvt_locked) {
ao2_unlock(p);
}
ao2_ref(p, -1);
}
if (chan) {
ast_channel_unlock(chan);
ast_channel_unref(chan);
}
/*
* owner is supposed to be == to ast, if it is, don't unlock it
* because ast must exit locked
*/
if (owner) {
if (owner != ast) {
ast_channel_unlock(owner);
ast_channel_lock(ast);
}
ast_channel_unref(owner);
} else {
/* we have to exit with ast locked */
ast_channel_lock(ast);
}
return res;
}
/*! \brief Hangup a call through the local proxy channel */
static int local_hangup(struct ast_channel *ast)
{
struct local_pvt *p = ast_channel_tech_pvt(ast);
int res;
if (!p) {
return -1;
}
/* give the pvt a ref to fulfill calling requirements. */
ao2_ref(p, +1);
res = ast_unreal_hangup(&p->base, ast);
if (!res) {
int unlink;
ao2_lock(p);
unlink = !p->base.owner && !p->base.chan;
ao2_unlock(p);
if (unlink) {
ao2_unlink(locals, p);
}
}
ao2_ref(p, -1);
return res;
}
/*!
* \internal
* \brief struct local_pvt destructor.
*
* \param vdoomed Object to destroy.
*/
static void local_pvt_destructor(void *vdoomed)
{
struct local_pvt *doomed = vdoomed;
switch (doomed->type) {
case LOCAL_CALL_ACTION_DIALPLAN:
break;
case LOCAL_CALL_ACTION_BRIDGE:
ao2_cleanup(doomed->action.bridge.join);
ao2_cleanup(doomed->action.bridge.swap);
ast_bridge_features_destroy(doomed->action.bridge.features);
break;
case LOCAL_CALL_ACTION_MASQUERADE:
ao2_cleanup(doomed->action.masq);
break;
}
ast_unreal_destructor(&doomed->base);
}
/*! \brief Create a call structure */
static struct local_pvt *local_alloc(const char *data, struct ast_stream_topology *topology)
{
struct local_pvt *pvt;
char *parse;
char *context;
char *opts;
pvt = (struct local_pvt *) ast_unreal_alloc_stream_topology(sizeof(*pvt), local_pvt_destructor, topology);
if (!pvt) {
return NULL;
}
pvt->base.callbacks = &local_unreal_callbacks;
parse = ast_strdupa(data);
/*
* Local channels intercept MOH by default.
*
* This is a silly default because it represents state held by
* the local channels. Unless local channel optimization is
* disabled, the state will dissapear when the local channels
* optimize out.
*/
ast_set_flag(&pvt->base, AST_UNREAL_MOH_INTERCEPT);
/* Look for options */
if ((opts = strchr(parse, '/'))) {
*opts++ = '\0';
if (strchr(opts, 'n')) {
ast_set_flag(&pvt->base, AST_UNREAL_NO_OPTIMIZATION);
}
if (strchr(opts, 'j')) {
if (ast_test_flag(&pvt->base, AST_UNREAL_NO_OPTIMIZATION)) {
ast_set_flag(&pvt->base.jb_conf, AST_JB_ENABLED);
} else {
ast_log(LOG_ERROR, "You must use the 'n' option with the 'j' option to enable the jitter buffer\n");
}
}
if (strchr(opts, 'm')) {
ast_clear_flag(&pvt->base, AST_UNREAL_MOH_INTERCEPT);
}
}
/* Look for a context */
if ((context = strchr(parse, '@'))) {
*context++ = '\0';
}
ast_copy_string(pvt->context, S_OR(context, "default"), sizeof(pvt->context));
ast_copy_string(pvt->exten, parse, sizeof(pvt->exten));
snprintf(pvt->base.name, sizeof(pvt->base.name), "%s@%s", pvt->exten, pvt->context);
return pvt; /* this is returned with a ref */
}
/*! \brief Part of PBX interface */
static struct ast_channel *local_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
{
struct ast_stream_topology *topology;
struct ast_channel *chan;
topology = ast_stream_topology_create_from_format_cap(cap);
if (!topology) {
return NULL;
}
chan = local_request_with_stream_topology(type, topology, assignedids, requestor, data, cause);
ast_stream_topology_free(topology);
return chan;
}
/*! \brief Part of PBX interface */
static struct ast_channel *local_request_with_stream_topology(const char *type, struct ast_stream_topology *topology, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause)
{
struct ast_stream_topology *audio_filtered_topology;
int i;
struct local_pvt *p;
struct ast_channel *chan;
ast_callid callid;
/* Create a copy of the requested topology as we don't have ownership over
* the one that is passed in.
*/
audio_filtered_topology = ast_stream_topology_clone(topology);
if (!audio_filtered_topology) {
return NULL;
}
/* Some users of Local channels request every known format in the
* universe. The core itself automatically pruned this list down to a single
* "best" format for audio in non-multistream. We replicate the logic here to
* do the same thing.
*/
for (i = 0; i < ast_stream_topology_get_count(audio_filtered_topology); ++i) {
struct ast_stream *stream;
int res;
struct ast_format *tmp_fmt = NULL;
struct ast_format *best_audio_fmt = NULL;
struct ast_format_cap *caps;
stream = ast_stream_topology_get_stream(audio_filtered_topology, i);
if (ast_stream_get_type(stream) != AST_MEDIA_TYPE_AUDIO ||
ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
continue;
}
/* Respect the immutable state of formats on the stream and create a new
* format capabilities to replace the existing one.
*/
caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!caps) {
ao2_ref(audio_filtered_topology, -1);
return NULL;
}
/* The ast_translator_best_choice function treats both caps as const
* but does not declare it in the API.
*/
res = ast_translator_best_choice((struct ast_format_cap *)ast_stream_get_formats(stream), local_tech.capabilities,
&tmp_fmt, &best_audio_fmt);
if (res < 0) {
struct ast_str *tech_codecs = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
struct ast_str *request_codecs = ast_str_alloca(AST_FORMAT_CAP_NAMES_LEN);
ast_log(LOG_WARNING, "No translator path exists for channel type %s (native %s) to %s\n", type,
ast_format_cap_get_names(local_tech.capabilities, &tech_codecs),
ast_format_cap_get_names(ast_stream_get_formats(stream), &request_codecs));
/* If there are no formats then we abort */
ao2_ref(caps, -1);
ao2_ref(audio_filtered_topology, -1);
return NULL;
}
ast_format_cap_append(caps, best_audio_fmt, 0);
ast_stream_set_formats(stream, caps);
ao2_ref(caps, -1);
ao2_ref(tmp_fmt, -1);
ao2_ref(best_audio_fmt, -1);
}
/* Allocate a new private structure and then Asterisk channels */
p = local_alloc(data, audio_filtered_topology);
ao2_ref(audio_filtered_topology, -1);
if (!p) {
return NULL;
}
callid = ast_read_threadstorage_callid();
chan = ast_unreal_new_channels(&p->base, &local_tech, AST_STATE_DOWN, AST_STATE_RING,
p->exten, p->context, assignedids, requestor, callid);
if (chan) {
ao2_link(locals, p);
}
ao2_ref(p, -1); /* kill the ref from the alloc */
return chan;
}
/*! \brief CLI command "local show channels" */
static char *locals_show(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
struct local_pvt *p;
struct ao2_iterator it;
switch (cmd) {
case CLI_INIT:
e->command = "local show channels";
e->usage =
"Usage: local show channels\n"
" Provides summary information on active local proxy channels.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3) {
return CLI_SHOWUSAGE;
}
if (ao2_container_count(locals) == 0) {
ast_cli(a->fd, "No local channels in use\n");
return RESULT_SUCCESS;
}
it = ao2_iterator_init(locals, 0);
while ((p = ao2_iterator_next(&it))) {
ao2_lock(p);
ast_cli(a->fd, "%s -- %s\n",
p->base.owner ? ast_channel_name(p->base.owner) : "<unowned>",
p->base.name);
ao2_unlock(p);
ao2_ref(p, -1);
}
ao2_iterator_destroy(&it);
return CLI_SUCCESS;
}
static struct ast_cli_entry cli_local[] = {
AST_CLI_DEFINE(locals_show, "List status of local channels"),
};
static int manager_optimize_away(struct mansession *s, const struct message *m)
{
const char *channel;
struct local_pvt *p;
struct local_pvt *found;
struct ast_channel *chan;
channel = astman_get_header(m, "Channel");
if (ast_strlen_zero(channel)) {
astman_send_error(s, m, "'Channel' not specified.");
return 0;
}
chan = ast_channel_get_by_name(channel);
if (!chan) {
astman_send_error(s, m, "Channel does not exist.");
return 0;
}
p = ast_channel_tech_pvt(chan);
ast_channel_unref(chan);
found = p ? ao2_find(locals, p, 0) : NULL;
if (found) {
ao2_lock(found);
ast_clear_flag(&found->base, AST_UNREAL_NO_OPTIMIZATION);
ao2_unlock(found);
ao2_ref(found, -1);
astman_send_ack(s, m, "Queued channel to be optimized away");
} else {
astman_send_error(s, m, "Unable to find channel");
}
return 0;
}
static int locals_cmp_cb(void *obj, void *arg, int flags)
{
return (obj == arg) ? CMP_MATCH : 0;
}
/*!
* \internal
* \brief Shutdown the local proxy channel.
* \since 12.0.0
*/
static void local_shutdown(void)
{
/* First, take us out of the channel loop */
ast_cli_unregister_multiple(cli_local, ARRAY_LEN(cli_local));
ast_manager_unregister("LocalOptimizeAway");
ast_channel_unregister(&local_tech);
ao2_ref(locals, -1);
locals = NULL;
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
ao2_cleanup(local_tech.capabilities);
local_tech.capabilities = NULL;
STASIS_MESSAGE_TYPE_CLEANUP(ast_local_optimization_begin_type);
STASIS_MESSAGE_TYPE_CLEANUP(ast_local_optimization_end_type);
STASIS_MESSAGE_TYPE_CLEANUP(ast_local_bridge_type);
}
int ast_local_init(void)
{
if (STASIS_MESSAGE_TYPE_INIT(ast_local_optimization_begin_type)) {
return -1;
}
if (STASIS_MESSAGE_TYPE_INIT(ast_local_optimization_end_type)) {
return -1;
}
if (STASIS_MESSAGE_TYPE_INIT(ast_local_bridge_type)) {
return -1;
}
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
if (!(local_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
return -1;
}
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
ast_format_cap_append_by_type(local_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
locals = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0, NULL, locals_cmp_cb);
if (!locals) {
return -1;
}
/* Make sure we can register our channel type */
if (ast_channel_register(&local_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class 'Local'\n");
return -1;
}
ast_cli_register_multiple(cli_local, ARRAY_LEN(cli_local));
ast_manager_register_xml_core("LocalOptimizeAway", EVENT_FLAG_SYSTEM|EVENT_FLAG_CALL, manager_optimize_away);
ast_register_cleanup(local_shutdown);
return 0;
}
int local_setoption(struct ast_channel *ast, int option, void *data, int datalen)
{
switch (option) {
case AST_OPTION_SECURE_SIGNALING:
case AST_OPTION_SECURE_MEDIA:
return 0; /* local calls (like forwardings) are secure always */
default:
return ast_unreal_setoption(ast, option, data, datalen);
}
}