[asterisk-commits] file: branch file/bridge_native r385908 - in /team/file/bridge_native: bridge...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Apr 16 13:55:51 CDT 2013
Author: file
Date: Tue Apr 16 13:55:47 2013
New Revision: 385908
URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=385908
Log:
Add a module which does native RTP bridging. Right now it's hardcoded for remote and doesn't do DTMF checks. One step at a time!
Added:
team/file/bridge_native/bridges/bridge_native_rtp.c (with props)
Modified:
team/file/bridge_native/channels/chan_sip.c
Added: team/file/bridge_native/bridges/bridge_native_rtp.c
URL: http://svnview.digium.com/svn/asterisk/team/file/bridge_native/bridges/bridge_native_rtp.c?view=auto&rev=385908
==============================================================================
--- team/file/bridge_native/bridges/bridge_native_rtp.c (added)
+++ team/file/bridge_native/bridges/bridge_native_rtp.c Tue Apr 16 13:55:47 2013
@@ -1,0 +1,238 @@
+/*
+ * 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"
+
+/*! \brief Structure which contains information about native bridged RTP capable channel */
+struct native_rtp_bridge {
+ /*! \brief What type of bridge is in progress */
+ enum ast_rtp_glue_result type;
+};
+
+static int native_rtp_bridge_compatible(struct ast_bridge *bridge)
+{
+ struct ast_channel *c0 = AST_LIST_FIRST(&bridge->channels)->chan;
+ struct ast_channel *c1 = AST_LIST_LAST(&bridge->channels)->chan;
+ struct ast_rtp_glue *glue0, *glue1;
+ 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;
+ 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 (c0 == c1) {
+ return 0;
+ }
+
+ if (!(glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type)) ||
+ !(glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type))) {
+ return 0;
+ }
+
+ 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 0;
+ }
+
+ /* Make sure that codecs match */
+ if (glue0->get_codec) {
+ glue0->get_codec(c0, cap0);
+ }
+ if (glue1->get_codec) {
+ glue1->get_codec(c1, 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))).cur_ms;
+ read_ptime1 = (ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance1)->pref, ast_channel_rawreadformat(c1))).cur_ms;
+ write_ptime0 = (ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance0)->pref, ast_channel_rawwriteformat(c0))).cur_ms;
+ write_ptime1 = (ast_codec_pref_getsize(&ast_rtp_instance_get_codecs(instance1)->pref, ast_channel_rawwriteformat(c1))).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;
+}
+
+static int native_rtp_bridge_join(struct ast_bridge *bridge, struct ast_bridge_channel *bridge_channel)
+{
+ struct ast_channel *c0 = AST_LIST_FIRST(&bridge->channels)->chan;
+ struct ast_channel *c1 = AST_LIST_LAST(&bridge->channels)->chan;
+ struct ast_rtp_glue *glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type);
+ struct ast_rtp_glue *glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type);
+ 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;
+ 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);
+
+ 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;
+
+ if (glue0->get_codec) {
+ glue0->get_codec(c0, cap0);
+ }
+ if (glue1->get_codec) {
+ glue1->get_codec(c1, cap1);
+ }
+
+ glue0->update_peer(c0, instance1, vinstance1, tinstance1, cap1, 0);
+ glue1->update_peer(c1, instance0, vinstance0, tinstance0, cap0, 0);
+
+ ast_log(LOG_NOTICE, "Res = %d %d %d %d\n", audio_glue0_res, audio_glue1_res, video_glue0_res, video_glue1_res);
+
+ 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_channel *c0 = AST_LIST_FIRST(&bridge->channels)->chan;
+ struct ast_channel *c1 = AST_LIST_LAST(&bridge->channels)->chan;
+ struct ast_rtp_glue *glue0 = ast_rtp_instance_get_glue(ast_channel_tech(c0)->type);
+ struct ast_rtp_glue *glue1 = ast_rtp_instance_get_glue(ast_channel_tech(c1)->type);
+
+ glue0->update_peer(c0, NULL, NULL, NULL, 0, 0);
+ glue1->update_peer(c1, NULL, NULL, NULL, 0, 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_LIST_FIRST(&bridge->channels);
+
+ /* Find the channel we actually want to write to */
+ if (other == bridge_channel) {
+ other = AST_LIST_LAST(&bridge->channels);
+ }
+
+ /* 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_1TO1MIX | AST_BRIDGE_CAPABILITY_NATIVE,
+ .preference = AST_BRIDGE_PREFERENCE_HIGH,
+ .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/file/bridge_native/bridges/bridge_native_rtp.c
------------------------------------------------------------------------------
svn:eol-style = native
Propchange: team/file/bridge_native/bridges/bridge_native_rtp.c
------------------------------------------------------------------------------
svn:keywords = Author Date Id Revision
Propchange: team/file/bridge_native/bridges/bridge_native_rtp.c
------------------------------------------------------------------------------
svn:mime-type = text/plain
Modified: team/file/bridge_native/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/file/bridge_native/channels/chan_sip.c?view=diff&rev=385908&r1=385907&r2=385908
==============================================================================
--- team/file/bridge_native/channels/chan_sip.c (original)
+++ team/file/bridge_native/channels/chan_sip.c Tue Apr 16 13:55:47 2013
@@ -32853,6 +32853,7 @@
/* Disable early RTP bridge */
if ((instance || vinstance || tinstance) &&
!ast_bridged_channel(chan) &&
+ !ast_channel_internal_bridge(chan) &&
!sip_cfg.directrtpsetup) {
sip_pvt_unlock(p);
ast_channel_unlock(chan);
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