/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2013, Digium, Inc. * * Joshua Colp * * 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 * * \ingroup bridges */ /*** MODULEINFO core ***/ #include "asterisk.h" ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include #include #include #include #include #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; 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); 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; } static void native_rtp_bridge_stop(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 = NULL; struct ast_rtp_instance *instance0 = NULL, *instance1 = NULL, *vinstance0 = NULL, *vinstance1 = NULL; 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); } 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_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); 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); } 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())) { 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");