Files
asterisk/bridges/bridge_native_rtp.c
Mark Michelson 3422314cb2 Fix assumption in bridge_native_rtp.c regarding number of participants in a bridge.
When a party leaves a bridge, there may be more participants in the bridge than expected.
As such, it is important not to make assumptions regarding the list of channels in a
bridge.

This change makes it so that when a party leaves a native RTP bridge, we unbridge it and
the party it was bridged with. Previously, the first and last channels in the list were
unbridged since it was assumed that these were the two channels that had been bridged. As
previously stated, a new party had been inserted into the bridge, so this logic did not
work properly.

(closes issue ASTERISK-22615)
reported by Matt Jordan

(closes issue ASTERISK-22532)
reported by Matt Jordan

Review: https://reviewboard.asterisk.org/r/2899


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@400403 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03 19:11:22 +00:00

555 lines
20 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Joshua Colp <jcolp@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 Native RTP bridging technology module
*
* \author Joshua Colp <jcolp@digium.com>
*
* \ingroup bridges
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/types.h>
#include <sys/stat.h>
#include "asterisk/module.h"
#include "asterisk/channel.h"
#include "asterisk/bridge.h"
#include "asterisk/bridge_technology.h"
#include "asterisk/frame.h"
#include "asterisk/rtp_engine.h"
/*! \brief Internal structure which contains information about bridged RTP channels */
struct native_rtp_bridge_data {
/*! \brief Framehook used to intercept certain control frames */
int id;
};
/*! \brief Internal helper function which gets all RTP information (glue and instances) relating to the given channels */
static enum ast_rtp_glue_result native_rtp_bridge_get(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_glue **glue0,
struct ast_rtp_glue **glue1, struct ast_rtp_instance **instance0, struct ast_rtp_instance **instance1,
struct ast_rtp_instance **vinstance0, struct ast_rtp_instance **vinstance1)
{
enum ast_rtp_glue_result audio_glue0_res;
enum ast_rtp_glue_result video_glue0_res;
enum ast_rtp_glue_result audio_glue1_res;
enum ast_rtp_glue_result video_glue1_res;
if (!(*glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) ||
!(*glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
return AST_RTP_GLUE_RESULT_FORBID;
}
audio_glue0_res = (*glue0)->get_rtp_info(c0, instance0);
video_glue0_res = (*glue0)->get_vrtp_info ? (*glue0)->get_vrtp_info(c0, vinstance0) : AST_RTP_GLUE_RESULT_FORBID;
audio_glue1_res = (*glue1)->get_rtp_info(c1, instance1);
video_glue1_res = (*glue1)->get_vrtp_info ? (*glue1)->get_vrtp_info(c1, vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
/* Apply any limitations on direct media bridging that may be present */
if (audio_glue0_res == audio_glue1_res && audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
if ((*glue0)->allow_rtp_remote && !((*glue0)->allow_rtp_remote(c0, *instance1))) {
/* If the allow_rtp_remote indicates that remote isn't allowed, revert to local bridge */
audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
} else if ((*glue1)->allow_rtp_remote && !((*glue1)->allow_rtp_remote(c1, *instance0))) {
audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
}
}
if (video_glue0_res == video_glue1_res && video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
if ((*glue0)->allow_vrtp_remote && !((*glue0)->allow_vrtp_remote(c0, *instance1))) {
/* if the allow_vrtp_remote indicates that remote isn't allowed, revert to local bridge */
video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
} else if ((*glue1)->allow_vrtp_remote && !((*glue1)->allow_vrtp_remote(c1, *instance0))) {
video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
}
}
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID
&& (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE
|| video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
}
if (video_glue1_res != AST_RTP_GLUE_RESULT_FORBID
&& (audio_glue1_res != AST_RTP_GLUE_RESULT_REMOTE
|| video_glue1_res != AST_RTP_GLUE_RESULT_REMOTE)) {
audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
}
/* If any sort of bridge is forbidden just completely bail out and go back to generic bridging */
if (audio_glue0_res == AST_RTP_GLUE_RESULT_FORBID
|| audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID) {
return AST_RTP_GLUE_RESULT_FORBID;
}
return audio_glue0_res;
}
static int native_rtp_bridge_start(struct ast_bridge *bridge, struct ast_channel *target)
{
struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels);
struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
enum ast_rtp_glue_result native_type;
struct ast_rtp_glue *glue0, *glue1;
RAII_VAR(struct ast_rtp_instance *, instance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, vinstance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, vinstance1, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, tinstance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, tinstance1, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, cap0, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK), ast_format_cap_destroy);
RAII_VAR(struct ast_format_cap *, cap1, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK), ast_format_cap_destroy);
if (c0 == c1) {
return 0;
}
native_type = native_rtp_bridge_get(c0->chan, c1->chan, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
if (glue0->get_codec) {
glue0->get_codec(c0->chan, cap0);
}
if (glue1->get_codec) {
glue1->get_codec(c1->chan, cap1);
}
switch (native_type) {
case AST_RTP_GLUE_RESULT_LOCAL:
if (ast_rtp_instance_get_engine(instance0)->local_bridge) {
ast_rtp_instance_get_engine(instance0)->local_bridge(instance0, instance1);
}
if (ast_rtp_instance_get_engine(instance1)->local_bridge) {
ast_rtp_instance_get_engine(instance1)->local_bridge(instance1, instance0);
}
ast_rtp_instance_set_bridged(instance0, instance1);
ast_rtp_instance_set_bridged(instance1, instance0);
ast_debug(2, "Locally RTP bridged '%s' and '%s' in stack\n",
ast_channel_name(c0->chan), ast_channel_name(c1->chan));
break;
case AST_RTP_GLUE_RESULT_REMOTE:
/* If we have a target, it's the channel that received the UNHOLD or UPDATE_RTP_PEER frame and was told to resume */
if (!target) {
glue0->update_peer(c0->chan, instance1, vinstance1, tinstance1, cap1, 0);
glue1->update_peer(c1->chan, instance0, vinstance0, tinstance0, cap0, 0);
ast_debug(2, "Remotely bridged '%s' and '%s' - media will flow directly between them\n",
ast_channel_name(c0->chan), ast_channel_name(c1->chan));
} else {
/*
* If a target was provided, it is the recipient of an unhold or an update and needs to have
* its media redirected to fit the current remote bridging needs. The other channel is either
* already set up to handle the new media path or will have its own set of updates independent
* of this pass.
*/
if (c0->chan == target) {
glue0->update_peer(c0->chan, instance1, vinstance1, tinstance1, cap1, 0);
} else {
glue1->update_peer(c1->chan, instance0, vinstance0, tinstance0, cap0, 0);
}
}
break;
case AST_RTP_GLUE_RESULT_FORBID:
break;
}
return 0;
}
/*!
* \internal
* \brief Given a bridge channel, get its RTP instance
*
* The returned ast_rtp_instance has its refcount bumped.
*
* \param bridge_channel Take a guess
* \retval NULL No RTP instance on this bridge channel
* \retval non-NULL The RTP instance on this bridge channel
*/
static struct ast_rtp_instance *bridge_channel_get_rtp_instance(struct ast_bridge_channel *bridge_channel)
{
struct ast_rtp_glue *glue;
struct ast_rtp_instance *instance;
glue = ast_rtp_instance_get_glue(ast_channel_tech(bridge_channel->chan)->type);
if (!glue) {
return NULL;
}
glue->get_rtp_info(bridge_channel->chan, &instance);
return instance;
}
/*!
* \internal
* \brief Determine which two channels are bridged together
*
* Because of the nature of swapping, when the time comes for a channel to
* leave a native RTP bridge, it may be that there are more than two channels
* in the list of bridge channels. Therefore, it is important to correctly
* determine which two channels were bridged together.
*
* \param bridge The involved bridge
* \param leaving The bridge channel that is leaving the native RTP bridge
* \param[out] c0 The first bridged channel
* \param[out] c1 The second bridged channel
*/
static void find_bridged_channels(struct ast_bridge *bridge, struct ast_bridge_channel *leaving,
struct ast_bridge_channel **c0, struct ast_bridge_channel **c1)
{
RAII_VAR(struct ast_rtp_instance *, leaving_instance, bridge_channel_get_rtp_instance(leaving), ao2_cleanup);
struct ast_bridge_channel *iter;
if (!leaving_instance) {
return;
}
AST_LIST_TRAVERSE(&bridge->channels, iter, entry) {
RAII_VAR(struct ast_rtp_instance *, instance, NULL, ao2_cleanup);
if (iter == leaving) {
continue;
}
instance = bridge_channel_get_rtp_instance(iter);
if (!instance) {
continue;
}
if (instance == ast_rtp_instance_get_bridged(leaving_instance)) {
break;
}
}
*c0 = leaving;
*c1 = iter;
return;
}
/*!
* \internal
* \brief Stop native RTP bridging of two channels
*
* \param bridge The bridge that had native RTP bridging happening on it
* \param target If remote RTP bridging, the channel that is placed on hold.
* \param leaving If this is called because a channel is leaving, this is the
* bridge channel that is leaving the bridge
*/
static void native_rtp_bridge_stop(struct ast_bridge *bridge, struct ast_channel *target,
struct ast_bridge_channel *leaving)
{
struct ast_bridge_channel *c0 = NULL;
struct ast_bridge_channel *c1 = NULL;
enum ast_rtp_glue_result native_type;
struct ast_rtp_glue *glue0, *glue1 = NULL;
RAII_VAR(struct ast_rtp_instance *, instance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, vinstance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, vinstance1, NULL, ao2_cleanup);
if (bridge->num_channels == 2) {
c0 = AST_LIST_FIRST(&bridge->channels);
c1 = AST_LIST_LAST(&bridge->channels);
} else if (bridge->num_channels > 2) {
/* When a channel leaves a native RTP bridge, it is possible for
* more channels to exist in the bridge than when the RTP bridge
* was started. Thus we need to determine which two channels were
* bridged based on the leaving channel
*/
find_bridged_channels(bridge, leaving, &c0, &c1);
} else {
return;
}
if (!c0 || !c1) {
return;
}
native_type = native_rtp_bridge_get(c0->chan, c1->chan, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
switch (native_type) {
case AST_RTP_GLUE_RESULT_LOCAL:
if (ast_rtp_instance_get_engine(instance0)->local_bridge) {
ast_rtp_instance_get_engine(instance0)->local_bridge(instance0, NULL);
}
if (instance1 && ast_rtp_instance_get_engine(instance1)->local_bridge) {
ast_rtp_instance_get_engine(instance1)->local_bridge(instance1, NULL);
}
ast_rtp_instance_set_bridged(instance0, NULL);
if (instance1) {
ast_rtp_instance_set_bridged(instance1, NULL);
}
break;
case AST_RTP_GLUE_RESULT_REMOTE:
if (!target) {
glue0->update_peer(c0->chan, NULL, NULL, NULL, NULL, 0);
if (glue1) {
glue1->update_peer(c1->chan, NULL, NULL, NULL, NULL, 0);
}
} else {
/*
* If a target was provided, it is being put on hold and should expect to
* receive mediafrom sterisk instead of what it was previously connected to.
*/
if (c0->chan == target) {
glue0->update_peer(c0->chan, NULL, NULL, NULL, NULL, 0);
} else if (glue1) {
glue1->update_peer(c1->chan, NULL, NULL, NULL, NULL, 0);
}
}
break;
case AST_RTP_GLUE_RESULT_FORBID:
break;
}
ast_debug(2, "Discontinued RTP bridging of '%s' and '%s' - media will flow through Asterisk core\n",
ast_channel_name(c0->chan), ast_channel_name(c1->chan));
}
/*! \brief Frame hook that is called to intercept hold/unhold */
static struct ast_frame *native_rtp_framehook(struct ast_channel *chan, struct ast_frame *f, enum ast_framehook_event event, void *data)
{
RAII_VAR(struct ast_bridge *, bridge, NULL, ao2_cleanup);
if (!f || (event != AST_FRAMEHOOK_EVENT_WRITE)) {
return f;
}
ast_channel_lock(chan);
bridge = ast_channel_get_bridge(chan);
ast_channel_unlock(chan);
if (bridge) {
if (f->subclass.integer == AST_CONTROL_HOLD) {
native_rtp_bridge_stop(bridge, chan, NULL);
} else if ((f->subclass.integer == AST_CONTROL_UNHOLD) || (f->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
native_rtp_bridge_start(bridge, chan);
}
}
return f;
}
/*! \brief Internal helper function which checks whether the channels are compatible with our native bridging */
static int native_rtp_bridge_capable(struct ast_channel *chan)
{
return !ast_channel_has_audio_frame_or_monitor(chan);
}
static int native_rtp_bridge_compatible(struct ast_bridge *bridge)
{
struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels);
struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
enum ast_rtp_glue_result native_type;
struct ast_rtp_glue *glue0, *glue1;
RAII_VAR(struct ast_rtp_instance *, instance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, instance1, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, vinstance0, NULL, ao2_cleanup);
RAII_VAR(struct ast_rtp_instance *, vinstance1, NULL, ao2_cleanup);
RAII_VAR(struct ast_format_cap *, cap0, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK), ast_format_cap_destroy);
RAII_VAR(struct ast_format_cap *, cap1, ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_NOLOCK), ast_format_cap_destroy);
int read_ptime0, read_ptime1, write_ptime0, write_ptime1;
/* We require two channels before even considering native bridging */
if (bridge->num_channels != 2) {
ast_debug(1, "Bridge '%s' can not use native RTP bridge as two channels are required\n",
bridge->uniqueid);
return 0;
}
if (!native_rtp_bridge_capable(c0->chan)) {
ast_debug(1, "Bridge '%s' can not use native RTP bridge as channel '%s' has features which prevent it\n",
bridge->uniqueid, ast_channel_name(c0->chan));
return 0;
}
if (!native_rtp_bridge_capable(c1->chan)) {
ast_debug(1, "Bridge '%s' can not use native RTP bridge as channel '%s' has features which prevent it\n",
bridge->uniqueid, ast_channel_name(c1->chan));
return 0;
}
if ((native_type = native_rtp_bridge_get(c0->chan, c1->chan, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1))
== AST_RTP_GLUE_RESULT_FORBID) {
ast_debug(1, "Bridge '%s' can not use native RTP bridge as it was forbidden while getting details\n",
bridge->uniqueid);
return 0;
}
if (ao2_container_count(c0->features->dtmf_hooks) && ast_rtp_instance_dtmf_mode_get(instance0)) {
ast_debug(1, "Bridge '%s' can not use native RTP bridge as channel '%s' has DTMF hooks\n",
bridge->uniqueid, ast_channel_name(c0->chan));
return 0;
}
if (ao2_container_count(c1->features->dtmf_hooks) && ast_rtp_instance_dtmf_mode_get(instance1)) {
ast_debug(1, "Bridge '%s' can not use native RTP bridge as channel '%s' has DTMF hooks\n",
bridge->uniqueid, ast_channel_name(c1->chan));
return 0;
}
if ((native_type == AST_RTP_GLUE_RESULT_LOCAL) && ((ast_rtp_instance_get_engine(instance0)->local_bridge !=
ast_rtp_instance_get_engine(instance1)->local_bridge) ||
(ast_rtp_instance_get_engine(instance0)->dtmf_compatible &&
!ast_rtp_instance_get_engine(instance0)->dtmf_compatible(c0->chan, instance0, c1->chan, instance1)))) {
ast_debug(1, "Bridge '%s' can not use local native RTP bridge as local bridge or DTMF is not compatible\n",
bridge->uniqueid);
return 0;
}
/* Make sure that codecs match */
if (glue0->get_codec) {
glue0->get_codec(c0->chan, cap0);
}
if (glue1->get_codec) {
glue1->get_codec(c1->chan, cap1);
}
if (!ast_format_cap_is_empty(cap0) && !ast_format_cap_is_empty(cap1) && !ast_format_cap_has_joint(cap0, cap1)) {
char tmp0[256] = { 0, }, tmp1[256] = { 0, };
ast_debug(1, "Channel codec0 = %s is not codec1 = %s, cannot native bridge in RTP.\n",
ast_getformatname_multiple(tmp0, sizeof(tmp0), cap0),
ast_getformatname_multiple(tmp1, sizeof(tmp1), cap1));
return 0;
}
read_ptime0 = (ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance0)->pref, ast_channel_rawreadformat(c0->chan))).cur_ms;
read_ptime1 = (ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance1)->pref, ast_channel_rawreadformat(c1->chan))).cur_ms;
write_ptime0 = (ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance0)->pref, ast_channel_rawwriteformat(c0->chan))).cur_ms;
write_ptime1 = (ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance1)->pref, ast_channel_rawwriteformat(c1->chan))).cur_ms;
if (read_ptime0 != write_ptime1 || read_ptime1 != write_ptime0) {
ast_debug(1, "Packetization differs between RTP streams (%d != %d or %d != %d). Cannot native bridge in RTP\n",
read_ptime0, write_ptime1, read_ptime1, write_ptime0);
return 0;
}
return 1;
}
/*! \brief Helper function which adds frame hook to bridge channel */
static int native_rtp_bridge_framehook_attach(struct ast_bridge_channel *bridge_channel)
{
struct native_rtp_bridge_data *data = ao2_alloc(sizeof(*data), NULL);
static struct ast_framehook_interface hook = {
.version = AST_FRAMEHOOK_INTERFACE_VERSION,
.event_cb = native_rtp_framehook,
};
if (!data) {
return -1;
}
ast_channel_lock(bridge_channel->chan);
data->id = ast_framehook_attach(bridge_channel->chan, &hook);
ast_channel_unlock(bridge_channel->chan);
if (data->id < 0) {
ao2_cleanup(data);
return -1;
}
bridge_channel->tech_pvt = data;
return 0;
}
/*! \brief Helper function which removes frame hook from bridge channel */
static void native_rtp_bridge_framehook_detach(struct ast_bridge_channel *bridge_channel)
{
RAII_VAR(struct native_rtp_bridge_data *, data, bridge_channel->tech_pvt, ao2_cleanup);
if (!data) {
return;
}
ast_channel_lock(bridge_channel->chan);
ast_framehook_detach(bridge_channel->chan, data->id);
ast_channel_unlock(bridge_channel->chan);
bridge_channel->tech_pvt = NULL;
}
static int native_rtp_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
{
native_rtp_bridge_framehook_detach(bridge_channel);
if (native_rtp_bridge_framehook_attach(bridge_channel)) {
return -1;
}
return native_rtp_bridge_start(bridge, NULL);
}
static void native_rtp_bridge_unsuspend(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
{
native_rtp_bridge_join(bridge, bridge_channel);
}
static void native_rtp_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
{
native_rtp_bridge_framehook_detach(bridge_channel);
native_rtp_bridge_stop(bridge, NULL, bridge_channel);
}
static int native_rtp_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
{
return ast_bridge_queue_everyone_else(bridge, bridge_channel, frame);
}
static struct ast_bridge_technology native_rtp_bridge = {
.name = "native_rtp",
.capabilities = AST_BRIDGE_CAPABILITY_NATIVE,
.preference = AST_BRIDGE_PREFERENCE_BASE_NATIVE,
.join = native_rtp_bridge_join,
.unsuspend = native_rtp_bridge_unsuspend,
.leave = native_rtp_bridge_leave,
.suspend = native_rtp_bridge_leave,
.write = native_rtp_bridge_write,
.compatible = native_rtp_bridge_compatible,
};
static int unload_module(void)
{
ast_format_cap_destroy(native_rtp_bridge.format_capabilities);
return ast_bridge_technology_unregister(&native_rtp_bridge);
}
static int load_module(void)
{
if (!(native_rtp_bridge.format_capabilities = ast_format_cap_alloc(0))) {
return AST_MODULE_LOAD_DECLINE;
}
ast_format_cap_add_all_by_type(native_rtp_bridge.format_capabilities, AST_FORMAT_TYPE_AUDIO);
ast_format_cap_add_all_by_type(native_rtp_bridge.format_capabilities, AST_FORMAT_TYPE_VIDEO);
ast_format_cap_add_all_by_type(native_rtp_bridge.format_capabilities, AST_FORMAT_TYPE_TEXT);
return ast_bridge_technology_register(&native_rtp_bridge);
}
AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Native RTP bridging module");