[asterisk-commits] file: branch group/bridge_construction r388316 - in /team/group/bridge_constr...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri May 10 10:26:51 CDT 2013
Author: file
Date: Fri May 10 10:26:40 2013
New Revision: 388316
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=388316
Log:
Add support for native bridging RTP capable channels within the new bridging work.
Review: https://reviewboard.asterisk.org/r/2465/
(closes issue ASTERISK-20869)
Added:
team/group/bridge_construction/bridges/bridge_native_rtp.c (with props)
Modified:
team/group/bridge_construction/apps/app_chanspy.c
team/group/bridge_construction/apps/app_mixmonitor.c
team/group/bridge_construction/channels/chan_gulp.c
team/group/bridge_construction/channels/chan_h323.c
team/group/bridge_construction/channels/chan_jingle.c
team/group/bridge_construction/channels/chan_mgcp.c
team/group/bridge_construction/channels/chan_motif.c
team/group/bridge_construction/channels/chan_sip.c
team/group/bridge_construction/channels/chan_skinny.c
team/group/bridge_construction/channels/chan_unistim.c
team/group/bridge_construction/include/asterisk/rtp_engine.h
team/group/bridge_construction/main/bridging.c
team/group/bridge_construction/main/rtp_engine.c
Modified: team/group/bridge_construction/apps/app_chanspy.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/apps/app_chanspy.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/apps/app_chanspy.c (original)
+++ team/group/bridge_construction/apps/app_chanspy.c Fri May 10 10:26:40 2013
@@ -482,15 +482,18 @@
static int start_spying(struct ast_autochan *autochan, const char *spychan_name, struct ast_audiohook *audiohook)
{
int res = 0;
- struct ast_channel *peer = NULL;
ast_log(LOG_NOTICE, "Attaching %s to %s\n", spychan_name, ast_channel_name(autochan->chan));
ast_set_flag(audiohook, AST_AUDIOHOOK_TRIGGER_SYNC | AST_AUDIOHOOK_SMALL_QUEUE);
res = ast_audiohook_attach(autochan->chan, audiohook);
- if (!res && ast_test_flag(ast_channel_flags(autochan->chan), AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(autochan->chan))) {
- ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
+ if (!res) {
+ ast_channel_lock(autochan->chan);
+ if (ast_channel_is_bridged(autochan->chan)) {
+ ast_softhangup_nolock(autochan->chan, AST_SOFTHANGUP_UNBRIDGE);
+ }
+ ast_channel_unlock(autochan->chan);
}
return res;
}
Modified: team/group/bridge_construction/apps/app_mixmonitor.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/apps/app_mixmonitor.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/apps/app_mixmonitor.c (original)
+++ team/group/bridge_construction/apps/app_mixmonitor.c Fri May 10 10:26:40 2013
@@ -411,7 +411,6 @@
static int startmon(struct ast_channel *chan, struct ast_audiohook *audiohook)
{
- struct ast_channel *peer = NULL;
int res = 0;
if (!chan)
@@ -419,8 +418,13 @@
ast_audiohook_attach(chan, audiohook);
- if (!res && ast_test_flag(ast_channel_flags(chan), AST_FLAG_NBRIDGE) && (peer = ast_bridged_channel(chan)))
- ast_softhangup(peer, AST_SOFTHANGUP_UNBRIDGE);
+ if (!res) {
+ ast_channel_lock(chan);
+ if (ast_channel_is_bridged(chan)) {
+ ast_softhangup_nolock(chan, AST_SOFTHANGUP_UNBRIDGE);
+ }
+ ast_channel_unlock(chan);
+ }
return res;
}
Added: team/group/bridge_construction/bridges/bridge_native_rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/bridges/bridge_native_rtp.c?view=auto&rev=388316
==============================================================================
--- team/group/bridge_construction/bridges/bridge_native_rtp.c (added)
+++ team/group/bridge_construction/bridges/bridge_native_rtp.c Fri May 10 10:26:40 2013
@@ -1,0 +1,414 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2013, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at 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 module
+ *
+ * \author Joshua Colp <jcolp at 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/bridging.h"
+#include "asterisk/bridging_technology.h"
+#include "asterisk/frame.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/audiohook.h"
+
+/*! \brief Forward declarations for frame hook usage */
+static int native_rtp_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel);
+static void native_rtp_bridge_leave(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel);
+
+/*! \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 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);
+
+ /* It's safe for NULL to be passed to both of these, bridge_channel isn't used at all */
+ if (bridge) {
+ if (f->subclass.integer == AST_CONTROL_HOLD) {
+ native_rtp_bridge_leave(ast_channel_internal_bridge(chan), NULL);
+ } else if ((f->subclass.integer == AST_CONTROL_UNHOLD) || (f->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
+ native_rtp_bridge_join(ast_channel_internal_bridge(chan), NULL);
+ }
+ }
+
+ 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)
+{
+ if (ast_channel_monitor(chan) || (ast_channel_audiohooks(chan) &&
+ !ast_audiohook_write_list_empty(ast_channel_audiohooks(chan))) ||
+ !ast_framehook_list_is_empty(ast_channel_framehooks(chan))) {
+ return 0;
+ } else {
+ return 1;
+ }
+}
+
+/*! \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 = AST_RTP_GLUE_RESULT_FORBID, video_glue0_res = AST_RTP_GLUE_RESULT_FORBID;
+ enum ast_rtp_glue_result audio_glue1_res = AST_RTP_GLUE_RESULT_FORBID, video_glue1_res = AST_RTP_GLUE_RESULT_FORBID;
+
+ if (!(*glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) ||
+ (c1 && !(*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;
+
+ if (c1) {
+ 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 (c1 && 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 (c1 && 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 || (c1 && audio_glue1_res == AST_RTP_GLUE_RESULT_FORBID)) {
+ return AST_RTP_GLUE_RESULT_FORBID;
+ }
+
+ return audio_glue0_res;
+}
+
+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;
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, *vinstance0 = NULL, *vinstance1 = NULL;
+ RAII_VAR(struct ast_format_cap *, cap0, ast_format_cap_alloc_nolock(), ast_format_cap_destroy);
+ RAII_VAR(struct ast_format_cap *, cap1, ast_format_cap_alloc_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);
+
+ if (!(data->id = ast_framehook_attach(bridge_channel->chan, &hook)) < 0) {
+ ast_channel_unlock(bridge_channel->chan);
+ ao2_cleanup(data);
+ return -1;
+ }
+
+ ast_channel_unlock(bridge_channel->chan);
+
+ bridge_channel->bridge_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->bridge_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->bridge_pvt = NULL;
+}
+
+static int native_rtp_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
+{
+ 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;
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, *vinstance0 = NULL;
+ struct ast_rtp_instance *vinstance1 = NULL, *tinstance0 = NULL, *tinstance1 = NULL;
+ RAII_VAR(struct ast_format_cap *, cap0, ast_format_cap_alloc_nolock(), ast_format_cap_destroy);
+ RAII_VAR(struct ast_format_cap *, cap1, ast_format_cap_alloc_nolock(), ast_format_cap_destroy);
+
+ native_rtp_bridge_framehook_detach(c0);
+ if (native_rtp_bridge_framehook_attach(c0)) {
+ return -1;
+ }
+
+ native_rtp_bridge_framehook_detach(c1);
+ if (native_rtp_bridge_framehook_attach(c1)) {
+ native_rtp_bridge_framehook_detach(c0);
+ return -1;
+ }
+
+ 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);
+ }
+
+ if (native_type == 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);
+ } else {
+ glue0->update_peer(c0->chan, instance1, vinstance1, tinstance1, cap1, 0);
+ glue1->update_peer(c1->chan, instance0, vinstance0, tinstance0, cap0, 0);
+ }
+
+ return 0;
+}
+
+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)
+{
+ struct ast_bridge_channel *c0 = AST_LIST_FIRST(&bridge->channels) ? AST_LIST_FIRST(&bridge->channels) : bridge_channel;
+ struct ast_bridge_channel *c1 = AST_LIST_LAST(&bridge->channels);
+ enum ast_rtp_glue_result native_type;
+ struct ast_rtp_glue *glue0, *glue1 = NULL;
+ struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, *vinstance0 = NULL, *vinstance1 = NULL;
+
+ native_rtp_bridge_framehook_detach(c0);
+ if (c1) {
+ native_rtp_bridge_framehook_detach(c1);
+ }
+
+ native_type = native_rtp_bridge_get(c0->chan, c1 ? c1->chan : NULL, &glue0, &glue1, &instance0, &instance1, &vinstance0, &vinstance1);
+
+ if (native_type == 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, instance1);
+ if (instance1) {
+ ast_rtp_instance_set_bridged(instance1, instance0);
+ }
+ } else {
+ glue0->update_peer(c0->chan, NULL, NULL, NULL, NULL, 0);
+ if (glue1) {
+ glue1->update_peer(c1->chan, NULL, NULL, NULL, NULL, 0);
+ }
+ }
+}
+
+static int native_rtp_bridge_write(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel, struct ast_frame *frame)
+{
+ struct ast_bridge_channel *other = ast_bridge_channel_peer(bridge_channel);
+
+ if (!other) {
+ return -1;
+ }
+
+ /* The bridging core takes care of freeing the passed in frame. */
+ ast_bridge_channel_queue_frame(other, frame);
+
+ return 0;
+}
+
+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())) {
+ 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");
Propchange: team/group/bridge_construction/bridges/bridge_native_rtp.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/group/bridge_construction/bridges/bridge_native_rtp.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/group/bridge_construction/bridges/bridge_native_rtp.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/group/bridge_construction/channels/chan_gulp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_gulp.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_gulp.c (original)
+++ team/group/bridge_construction/channels/chan_gulp.c Fri May 10 10:26:40 2013
@@ -132,7 +132,6 @@
.send_text = gulp_sendtext,
.send_digit_begin = gulp_digit_begin,
.send_digit_end = gulp_digit_end,
- .bridge = ast_rtp_instance_bridge,
.call = gulp_call,
.hangup = gulp_hangup,
.answer = gulp_answer,
Modified: team/group/bridge_construction/channels/chan_h323.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_h323.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_h323.c (original)
+++ team/group/bridge_construction/channels/chan_h323.c Fri May 10 10:26:40 2013
@@ -275,7 +275,6 @@
.write = oh323_write,
.indicate = oh323_indicate,
.fixup = oh323_fixup,
- .bridge = ast_rtp_instance_bridge,
};
static const char* redirectingreason2str(int redirectingreason)
Modified: team/group/bridge_construction/channels/chan_jingle.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_jingle.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_jingle.c (original)
+++ team/group/bridge_construction/channels/chan_jingle.c Fri May 10 10:26:40 2013
@@ -205,7 +205,6 @@
.send_text = jingle_sendtext,
.send_digit_begin = jingle_digit_begin,
.send_digit_end = jingle_digit_end,
- .bridge = ast_rtp_instance_bridge,
.call = jingle_call,
.hangup = jingle_hangup,
.answer = jingle_answer,
Modified: team/group/bridge_construction/channels/chan_mgcp.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_mgcp.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_mgcp.c (original)
+++ team/group/bridge_construction/channels/chan_mgcp.c Fri May 10 10:26:40 2013
@@ -480,7 +480,6 @@
.fixup = mgcp_fixup,
.send_digit_begin = mgcp_senddigit_begin,
.send_digit_end = mgcp_senddigit_end,
- .bridge = ast_rtp_instance_bridge,
.func_channel_read = acf_channel_read,
};
Modified: team/group/bridge_construction/channels/chan_motif.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_motif.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_motif.c (original)
+++ team/group/bridge_construction/channels/chan_motif.c Fri May 10 10:26:40 2013
@@ -360,7 +360,6 @@
.send_text = jingle_sendtext,
.send_digit_begin = jingle_digit_begin,
.send_digit_end = jingle_digit_end,
- .bridge = ast_rtp_instance_bridge,
.call = jingle_call,
.hangup = jingle_hangup,
.answer = jingle_answer,
Modified: team/group/bridge_construction/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_sip.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_sip.c (original)
+++ team/group/bridge_construction/channels/chan_sip.c Fri May 10 10:26:40 2013
@@ -1543,7 +1543,6 @@
.fixup = sip_fixup, /* called with chan locked */
.send_digit_begin = sip_senddigit_begin, /* called with chan unlocked */
.send_digit_end = sip_senddigit_end,
- .bridge = ast_rtp_instance_bridge, /* XXX chan unlocked ? */
.early_bridge = ast_rtp_instance_early_bridge,
.send_text = sip_sendtext, /* called with chan locked */
.func_channel_read = sip_acf_channel_read,
@@ -32609,7 +32608,7 @@
/* Disable early RTP bridge */
if ((instance || vinstance || tinstance) &&
- !ast_bridged_channel(chan) &&
+ !ast_channel_is_bridged(chan) &&
!sip_cfg.directrtpsetup) {
sip_pvt_unlock(p);
ast_channel_unlock(chan);
Modified: team/group/bridge_construction/channels/chan_skinny.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_skinny.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_skinny.c (original)
+++ team/group/bridge_construction/channels/chan_skinny.c Fri May 10 10:26:40 2013
@@ -1635,7 +1635,6 @@
.fixup = skinny_fixup,
.send_digit_begin = skinny_senddigit_begin,
.send_digit_end = skinny_senddigit_end,
- .bridge = ast_rtp_instance_bridge,
};
static int skinny_extensionstate_cb(char *context, char *id, struct ast_state_cb_info *info, void *data);
Modified: team/group/bridge_construction/channels/chan_unistim.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/channels/chan_unistim.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/channels/chan_unistim.c (original)
+++ team/group/bridge_construction/channels/chan_unistim.c Fri May 10 10:26:40 2013
@@ -708,7 +708,6 @@
.send_digit_begin = unistim_senddigit_begin,
.send_digit_end = unistim_senddigit_end,
.send_text = unistim_sendtext,
- .bridge = ast_rtp_instance_bridge,
};
static void send_start_rtp(struct unistim_subchannel *);
Modified: team/group/bridge_construction/include/asterisk/rtp_engine.h
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/include/asterisk/rtp_engine.h?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/include/asterisk/rtp_engine.h (original)
+++ team/group/bridge_construction/include/asterisk/rtp_engine.h Fri May 10 10:26:40 2013
@@ -1541,24 +1541,6 @@
struct ast_rtp_glue *ast_rtp_instance_get_glue(const char *type);
/*!
- * \brief Bridge two channels that use RTP instances
- *
- * \param c0 First channel part of the bridge
- * \param c1 Second channel part of the bridge
- * \param flags Bridging flags
- * \param fo If a frame needs to be passed up it is stored here
- * \param rc Channel that passed the above frame up
- * \param timeoutms How long the channels should be bridged for
- *
- * \retval Bridge result
- *
- * \note This should only be used by channel drivers in their technology declaration.
- *
- * \since 1.8
- */
-enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms);
-
-/*!
* \brief Get the other RTP instance that an instance is bridged to
*
* \param instance The RTP instance that we want
@@ -1577,6 +1559,16 @@
* \since 1.8
*/
struct ast_rtp_instance *ast_rtp_instance_get_bridged(struct ast_rtp_instance *instance);
+
+/*!
+ * \brief Set the other RTP instance that an instance is bridged to
+ *
+ * \param instance The RTP instance that we want to set the bridged value on
+ * \param bridged The RTP instance they are bridged to
+ *
+ * \since 12
+ */
+void ast_rtp_instance_set_bridged(struct ast_rtp_instance *instance, struct ast_rtp_instance *bridged);
/*!
* \brief Make two channels compatible for early bridging
Modified: team/group/bridge_construction/main/bridging.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/main/bridging.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/main/bridging.c (original)
+++ team/group/bridge_construction/main/bridging.c Fri May 10 10:26:40 2013
@@ -2274,6 +2274,13 @@
chan = ast_waitfor_nandfds(&bridge_channel->chan, 1,
&bridge_channel->alert_pipe[0], 1, NULL, &outfd, &ms);
bridge_channel->waiting = 0;
+ if (ast_channel_softhangup_internal_flag(bridge_channel->chan) & AST_SOFTHANGUP_UNBRIDGE) {
+ ast_channel_clear_softhangup(bridge_channel->chan, AST_SOFTHANGUP_UNBRIDGE);
+ ast_bridge_channel_lock_bridge(bridge_channel);
+ bridge_channel->bridge->reconfigured = 1;
+ bridge_reconfigured(bridge_channel->bridge);
+ ast_bridge_unlock(bridge_channel->bridge);
+ }
ast_bridge_channel_lock(bridge_channel);
bridge_channel->activity = AST_BRIDGE_CHANNEL_THREAD_FRAME;
ast_bridge_channel_unlock(bridge_channel);
Modified: team/group/bridge_construction/main/rtp_engine.c
URL: http://svnview.digium.com/svn/asterisk/team/group/bridge_construction/main/rtp_engine.c?view=diff&rev=388316&r1=388315&r2=388316
==============================================================================
--- team/group/bridge_construction/main/rtp_engine.c (original)
+++ team/group/bridge_construction/main/rtp_engine.c Fri May 10 10:26:40 2013
@@ -928,497 +928,6 @@
return glue;
}
-static enum ast_bridge_result local_bridge_loop(struct ast_channel *c0, struct ast_channel *c1, struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1, int timeoutms, int flags, struct ast_frame **fo, struct ast_channel **rc, void *pvt0, void *pvt1)
-{
- enum ast_bridge_result res = AST_BRIDGE_FAILED;
- struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
- struct ast_frame *fr = NULL;
- struct timeval start;
-
- /* Start locally bridging both instances */
- if (instance0->engine->local_bridge && instance0->engine->local_bridge(instance0, instance1)) {
- ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c0), ast_channel_name(c1));
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
- if (instance1->engine->local_bridge && instance1->engine->local_bridge(instance1, instance0)) {
- ast_debug(1, "Failed to locally bridge %s to %s, backing out.\n", ast_channel_name(c1), ast_channel_name(c0));
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, NULL);
- }
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- return AST_BRIDGE_FAILED_NOWARN;
- }
-
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
-
- instance0->bridged = instance1;
- instance1->bridged = instance0;
-
- ast_poll_channel_add(c0, c1);
-
- /* Hop into a loop waiting for a frame from either channel */
- cs[0] = c0;
- cs[1] = c1;
- cs[2] = NULL;
- start = ast_tvnow();
- for (;;) {
- int ms;
- /* If the underlying formats have changed force this bridge to break */
- if ((ast_format_cmp(ast_channel_rawreadformat(c0), ast_channel_rawwriteformat(c1)) == AST_FORMAT_CMP_NOT_EQUAL) ||
- (ast_format_cmp(ast_channel_rawreadformat(c1), ast_channel_rawwriteformat(c0)) == AST_FORMAT_CMP_NOT_EQUAL)) {
- ast_debug(1, "rtp-engine-local-bridge: Oooh, formats changed, backing out\n");
- res = AST_BRIDGE_FAILED_NOWARN;
- break;
- }
- /* Check if anything changed */
- if ((ast_channel_tech_pvt(c0) != pvt0) ||
- (ast_channel_tech_pvt(c1) != pvt1) ||
- (ast_channel_masq(c0) || ast_channel_masqr(c0) || ast_channel_masq(c1) || ast_channel_masqr(c1)) ||
- (ast_channel_monitor(c0) || ast_channel_audiohooks(c0) || ast_channel_monitor(c1) || ast_channel_audiohooks(c1)) ||
- (!ast_framehook_list_is_empty(ast_channel_framehooks(c0)) || !ast_framehook_list_is_empty(ast_channel_framehooks(c1)))) {
- ast_debug(1, "rtp-engine-local-bridge: Oooh, something is weird, backing out\n");
- /* If a masquerade needs to happen we have to try to read in a frame so that it actually happens. Without this we risk being called again and going into a loop */
- if ((ast_channel_masq(c0) || ast_channel_masqr(c0)) && (fr = ast_read(c0))) {
- ast_frfree(fr);
- }
- if ((ast_channel_masq(c1) || ast_channel_masqr(c1)) && (fr = ast_read(c1))) {
- ast_frfree(fr);
- }
- res = AST_BRIDGE_RETRY;
- break;
- }
- /* Wait on a channel to feed us a frame */
- ms = ast_remaining_ms(start, timeoutms);
- if (!(who = ast_waitfor_n(cs, 2, &ms))) {
- if (!ms) {
- res = AST_BRIDGE_RETRY;
- break;
- }
- ast_debug(2, "rtp-engine-local-bridge: Ooh, empty read...\n");
- if (ast_check_hangup(c0) || ast_check_hangup(c1)) {
- break;
- }
- continue;
- }
- /* Read in frame from channel */
- fr = ast_read(who);
- other = (who == c0) ? c1 : c0;
- /* Depending on the frame we may need to break out of our bridge */
- if (!fr || ((fr->frametype == AST_FRAME_DTMF_BEGIN || fr->frametype == AST_FRAME_DTMF_END) &&
- ((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) |
- ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)))) {
- /* Record received frame and who */
- *fo = fr;
- *rc = who;
- ast_debug(1, "rtp-engine-local-bridge: Ooh, got a %s\n", fr ? "digit" : "hangup");
- res = AST_BRIDGE_COMPLETE;
- break;
- } else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
- if ((fr->subclass.integer == AST_CONTROL_HOLD) ||
- (fr->subclass.integer == AST_CONTROL_UNHOLD) ||
- (fr->subclass.integer == AST_CONTROL_VIDUPDATE) ||
- (fr->subclass.integer == AST_CONTROL_SRCUPDATE) ||
- (fr->subclass.integer == AST_CONTROL_T38_PARAMETERS) ||
- (fr->subclass.integer == AST_CONTROL_UPDATE_RTP_PEER)) {
- /* If we are going on hold, then break callback mode and P2P bridging */
- if (fr->subclass.integer == AST_CONTROL_HOLD) {
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, NULL);
- }
- if (instance1->engine->local_bridge) {
- instance1->engine->local_bridge(instance1, NULL);
- }
- instance0->bridged = NULL;
- instance1->bridged = NULL;
- } else if (fr->subclass.integer == AST_CONTROL_UNHOLD) {
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, instance1);
- }
- if (instance1->engine->local_bridge) {
- instance1->engine->local_bridge(instance1, instance0);
- }
- instance0->bridged = instance1;
- instance1->bridged = instance0;
- }
- /* Since UPDATE_BRIDGE_PEER is only used by the bridging code, don't forward it */
- if (fr->subclass.integer != AST_CONTROL_UPDATE_RTP_PEER) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_CONNECTED_LINE) {
- if (ast_channel_connected_line_sub(who, other, fr, 1) &&
- ast_channel_connected_line_macro(who, other, fr, other == c0, 1)) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_REDIRECTING) {
- if (ast_channel_redirecting_sub(who, other, fr, 1) &&
- ast_channel_redirecting_macro(who, other, fr, other == c0, 1)) {
- ast_indicate_data(other, fr->subclass.integer, fr->data.ptr, fr->datalen);
- }
- ast_frfree(fr);
- } else if (fr->subclass.integer == AST_CONTROL_PVT_CAUSE_CODE) {
- ast_channel_hangupcause_hash_set(other, fr->data.ptr, fr->datalen);
- ast_frfree(fr);
- } else {
- *fo = fr;
- *rc = who;
- ast_debug(1, "rtp-engine-local-bridge: Got a FRAME_CONTROL (%d) frame on channel %s\n", fr->subclass.integer, ast_channel_name(who));
- res = AST_BRIDGE_COMPLETE;
- break;
- }
- } else {
- if ((fr->frametype == AST_FRAME_DTMF_BEGIN) ||
- (fr->frametype == AST_FRAME_DTMF_END) ||
- (fr->frametype == AST_FRAME_VOICE) ||
- (fr->frametype == AST_FRAME_VIDEO) ||
- (fr->frametype == AST_FRAME_IMAGE) ||
- (fr->frametype == AST_FRAME_HTML) ||
- (fr->frametype == AST_FRAME_MODEM) ||
- (fr->frametype == AST_FRAME_TEXT)) {
- ast_write(other, fr);
- }
-
- ast_frfree(fr);
- }
- /* Swap priority */
- cs[2] = cs[0];
- cs[0] = cs[1];
- cs[1] = cs[2];
- }
-
- /* Stop locally bridging both instances */
- if (instance0->engine->local_bridge) {
- instance0->engine->local_bridge(instance0, NULL);
- }
- if (instance1->engine->local_bridge) {
- instance1->engine->local_bridge(instance1, NULL);
- }
-
- instance0->bridged = NULL;
- instance1->bridged = NULL;
-
- ast_poll_channel_del(c0, c1);
-
- return res;
-}
-
-static enum ast_bridge_result remote_bridge_loop(struct ast_channel *c0,
- struct ast_channel *c1,
- struct ast_rtp_instance *instance0,
- struct ast_rtp_instance *instance1,
- struct ast_rtp_instance *vinstance0,
- struct ast_rtp_instance *vinstance1,
- struct ast_rtp_instance *tinstance0,
- struct ast_rtp_instance *tinstance1,
- struct ast_rtp_glue *glue0,
- struct ast_rtp_glue *glue1,
- struct ast_format_cap *cap0,
- struct ast_format_cap *cap1,
- int timeoutms,
- int flags,
- struct ast_frame **fo,
- struct ast_channel **rc,
- void *pvt0,
- void *pvt1)
-{
- enum ast_bridge_result res = AST_BRIDGE_FAILED;
- struct ast_channel *who = NULL, *other = NULL, *cs[3] = { NULL, };
- struct ast_format_cap *oldcap0 = ast_format_cap_dup(cap0);
- struct ast_format_cap *oldcap1 = ast_format_cap_dup(cap1);
- struct ast_sockaddr ac1 = {{0,}}, vac1 = {{0,}}, tac1 = {{0,}}, ac0 = {{0,}}, vac0 = {{0,}}, tac0 = {{0,}};
- struct ast_sockaddr t1 = {{0,}}, vt1 = {{0,}}, tt1 = {{0,}}, t0 = {{0,}}, vt0 = {{0,}}, tt0 = {{0,}};
- struct ast_frame *fr = NULL;
- struct timeval start;
-
- if (!oldcap0 || !oldcap1) {
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
- goto remote_bridge_cleanup;
- }
- /* Test the first channel */
- if (!(glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0))) {
- ast_rtp_instance_get_remote_address(instance1, &ac1);
- if (vinstance1) {
- ast_rtp_instance_get_remote_address(vinstance1, &vac1);
- }
- if (tinstance1) {
- ast_rtp_instance_get_remote_address(tinstance1, &tac1);
- }
- } else {
- ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c0), ast_channel_name(c1));
- }
-
- /* Test the second channel */
- if (!(glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0))) {
- ast_rtp_instance_get_remote_address(instance0, &ac0);
- if (vinstance0) {
- ast_rtp_instance_get_remote_address(instance0, &vac0);
- }
- if (tinstance0) {
- ast_rtp_instance_get_remote_address(instance0, &tac0);
- }
- } else {
- ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", ast_channel_name(c1), ast_channel_name(c0));
- }
-
- ast_channel_unlock(c0);
- ast_channel_unlock(c1);
-
- instance0->bridged = instance1;
- instance1->bridged = instance0;
-
- ast_poll_channel_add(c0, c1);
-
- /* Go into a loop handling any stray frames that may come in */
- cs[0] = c0;
- cs[1] = c1;
- cs[2] = NULL;
- start = ast_tvnow();
- for (;;) {
- int ms;
- /* Check if anything changed */
- if ((ast_channel_tech_pvt(c0) != pvt0) ||
- (ast_channel_tech_pvt(c1) != pvt1) ||
[... 429 lines stripped ...]
More information about the asterisk-commits
mailing list