/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2017, Digium, Inc. * * Mark Michelson * * 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. */ #include "asterisk.h" #include "asterisk/sdp_state.h" #include "asterisk/sdp_options.h" #include "asterisk/sdp_translator.h" #include "asterisk/vector.h" #include "asterisk/utils.h" #include "asterisk/netsock2.h" #include "asterisk/rtp_engine.h" #include "asterisk/format.h" #include "asterisk/format_cap.h" #include "asterisk/config.h" #include "asterisk/codec.h" #include "../include/asterisk/sdp.h" #include "asterisk/stream.h" #include "sdp_private.h" enum ast_sdp_role { /*! * \brief The role has not yet been determined. * * When the SDP state is allocated, this is the starting role. * Similarly, when the SDP state is reset, the role is reverted * to this. */ SDP_ROLE_NOT_SET, /*! * \brief We are the offerer. * * If a local SDP is requested before a remote SDP has been set, then * we assume the role of offerer. This means that we will generate an * SDP from the local capabilities and configured options. */ SDP_ROLE_OFFERER, /*! * \brief We are the answerer. * * If a remote SDP is set before a local SDP is requested, then we * assume the role of answerer. This means that we will generate an * SDP based on a merge of the remote capabilities and our local capabilities. */ SDP_ROLE_ANSWERER, }; typedef int (*state_fn)(struct ast_sdp_state *state); struct sdp_state_stream { union { /*! The underlying RTP instance */ struct ast_rtp_instance *instance; }; /*! An explicit connection address for this stream */ struct ast_sockaddr connection_address; /*! Whether this stream is held or not */ unsigned int locally_held; }; static void sdp_state_stream_free(struct sdp_state_stream *state_stream) { if (state_stream->instance) { ast_rtp_instance_destroy(state_stream->instance); } ast_free(state_stream); } AST_VECTOR(sdp_state_streams, struct sdp_state_stream *); struct sdp_state_capabilities { /*! Stream topology */ struct ast_stream_topology *topology; /*! Additional information about the streams */ struct sdp_state_streams streams; /*! An explicit global connection address */ struct ast_sockaddr connection_address; }; static void sdp_state_capabilities_free(struct sdp_state_capabilities *capabilities) { if (!capabilities) { return; } ast_stream_topology_free(capabilities->topology); AST_VECTOR_CALLBACK_VOID(&capabilities->streams, sdp_state_stream_free); AST_VECTOR_FREE(&capabilities->streams); ast_free(capabilities); } /* TODO * This isn't set anywhere yet. */ /*! \brief Scheduler for RTCP purposes */ static struct ast_sched_context *sched; /*! \brief Internal function which creates an RTP instance */ static struct ast_rtp_instance *create_rtp(const struct ast_sdp_options *options, enum ast_media_type media_type) { struct ast_rtp_instance *rtp; struct ast_rtp_engine_ice *ice; struct ast_sockaddr temp_media_address; static struct ast_sockaddr address_rtp; struct ast_sockaddr *media_address = &address_rtp; if (options->bind_rtp_to_media_address && !ast_strlen_zero(options->media_address)) { ast_sockaddr_parse(&temp_media_address, options->media_address, 0); media_address = &temp_media_address; } else { if (ast_check_ipv6()) { ast_sockaddr_parse(&address_rtp, "::", 0); } else { ast_sockaddr_parse(&address_rtp, "0.0.0.0", 0); } } if (!(rtp = ast_rtp_instance_new(options->rtp_engine, sched, media_address, NULL))) { ast_log(LOG_ERROR, "Unable to create RTP instance using RTP engine '%s'\n", options->rtp_engine); return NULL; } ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_RTCP, 1); ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_NAT, options->rtp_symmetric); if (options->ice == AST_SDP_ICE_DISABLED && (ice = ast_rtp_instance_get_ice(rtp))) { ice->stop(rtp); } if (options->telephone_event) { ast_rtp_instance_dtmf_mode_set(rtp, AST_RTP_DTMF_MODE_RFC2833); ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_DTMF, 1); } if (media_type == AST_MEDIA_TYPE_AUDIO && (options->tos_audio || options->cos_audio)) { ast_rtp_instance_set_qos(rtp, options->tos_audio, options->cos_audio, "SIP RTP Audio"); } else if (media_type == AST_MEDIA_TYPE_VIDEO && (options->tos_video || options->cos_video)) { ast_rtp_instance_set_qos(rtp, options->tos_video, options->cos_video, "SIP RTP Video"); } ast_rtp_instance_set_last_rx(rtp, time(NULL)); return rtp; } static struct sdp_state_capabilities *sdp_initialize_state_capabilities(const struct ast_stream_topology *topology, const struct ast_sdp_options *options) { struct sdp_state_capabilities *capabilities; int i; capabilities = ast_calloc(1, sizeof(*capabilities)); if (!capabilities) { return NULL; } capabilities->topology = ast_stream_topology_clone(topology); if (!capabilities->topology) { sdp_state_capabilities_free(capabilities); return NULL; } if (AST_VECTOR_INIT(&capabilities->streams, ast_stream_topology_get_count(topology))) { sdp_state_capabilities_free(capabilities); return NULL; } ast_sockaddr_setnull(&capabilities->connection_address); for (i = 0; i < ast_stream_topology_get_count(topology); ++i) { struct sdp_state_stream *state_stream; enum ast_media_type stream_type; state_stream = ast_calloc(1, sizeof(*state_stream)); if (!state_stream) { return NULL; } stream_type = ast_stream_get_type(ast_stream_topology_get_stream(topology, i)); if (stream_type == AST_MEDIA_TYPE_AUDIO || stream_type == AST_MEDIA_TYPE_VIDEO) { state_stream->instance = create_rtp(options, stream_type); } if (!state_stream->instance) { sdp_state_stream_free(state_stream); return NULL; } AST_VECTOR_APPEND(&capabilities->streams, state_stream); } return capabilities; } /*! * \brief SDP state, the main structure used to keep track of SDP negotiation * and settings. * * Most fields are pretty self-explanatory, but negotiated_capabilities and * proposed_capabilities could use some further explanation. When an SDP * state is allocated, a stream topology is provided that dictates the * types of streams to offer in the resultant SDP. At the time the SDP * is allocated, this topology is used to create the proposed_capabilities. * * If we are the SDP offerer, then the proposed_capabilities are what are used * to generate the SDP offer. When the SDP answer arrives, the proposed capabilities * are merged with the SDP answer to create the negotiated capabilities. * * If we are the SDP answerer, then the incoming SDP offer is merged with our * proposed capabilities to to create the negotiated capabilities. These negotiated * capabilities are what we send in our SDP answer. * * Any changes that a user of the API performs will occur on the proposed capabilities. * The negotiated capabilities are only altered based on actual SDP negotiation. This is * done so that the negotiated capabilities can be fallen back on if the proposed * capabilities run into some sort of issue. */ struct ast_sdp_state { /*! Current capabilities */ struct sdp_state_capabilities *negotiated_capabilities; /*! Proposed capabilities */ struct sdp_state_capabilities *proposed_capabilities; /*! Remote capabilities, learned through remote SDP */ struct ast_stream_topology *remote_capabilities; /*! Local SDP. Generated via the options and local capabilities. */ struct ast_sdp *local_sdp; /*! SDP options. Configured options beyond media capabilities. */ struct ast_sdp_options *options; /*! Translator that puts SDPs into the expected representation */ struct ast_sdp_translator *translator; /*! The role that we occupy in SDP negotiation */ enum ast_sdp_role role; }; struct ast_sdp_state *ast_sdp_state_alloc(struct ast_stream_topology *streams, struct ast_sdp_options *options) { struct ast_sdp_state *sdp_state; sdp_state = ast_calloc(1, sizeof(*sdp_state)); if (!sdp_state) { return NULL; } sdp_state->options = options; sdp_state->translator = ast_sdp_translator_new(ast_sdp_options_get_impl(sdp_state->options)); if (!sdp_state->translator) { ast_sdp_state_free(sdp_state); return NULL; } sdp_state->proposed_capabilities = sdp_initialize_state_capabilities(streams, options); if (!sdp_state->proposed_capabilities) { ast_sdp_state_free(sdp_state); return NULL; } sdp_state->role = SDP_ROLE_NOT_SET; return sdp_state; } void ast_sdp_state_free(struct ast_sdp_state *sdp_state) { if (!sdp_state) { return; } sdp_state_capabilities_free(sdp_state->negotiated_capabilities); sdp_state_capabilities_free(sdp_state->proposed_capabilities); ast_stream_topology_free(sdp_state->remote_capabilities); ast_sdp_free(sdp_state->local_sdp); ast_sdp_options_free(sdp_state->options); ast_sdp_translator_free(sdp_state->translator); ast_free(sdp_state); } static struct sdp_state_stream *sdp_state_get_stream(const struct ast_sdp_state *sdp_state, int stream_index) { if (stream_index >= AST_VECTOR_SIZE(&sdp_state->proposed_capabilities->streams)) { return NULL; } return AST_VECTOR_GET(&sdp_state->proposed_capabilities->streams, stream_index); } struct ast_rtp_instance *ast_sdp_state_get_rtp_instance( const struct ast_sdp_state *sdp_state, int stream_index) { struct sdp_state_stream *stream_state; ast_assert(sdp_state != NULL); stream_state = sdp_state_get_stream(sdp_state, stream_index); if (!stream_state) { return NULL; } return stream_state->instance; } const struct ast_sockaddr *ast_sdp_state_get_connection_address(const struct ast_sdp_state *sdp_state) { ast_assert(sdp_state != NULL); return &sdp_state->proposed_capabilities->connection_address; } int ast_sdp_state_get_stream_connection_address(const struct ast_sdp_state *sdp_state, int stream_index, struct ast_sockaddr *address) { struct sdp_state_stream *stream_state; enum ast_media_type type; ast_assert(sdp_state != NULL); ast_assert(address != NULL); stream_state = sdp_state_get_stream(sdp_state, stream_index); if (!stream_state) { return -1; } /* If an explicit connection address has been provided for the stream return it */ if (!ast_sockaddr_isnull(&stream_state->connection_address)) { ast_sockaddr_copy(address, &stream_state->connection_address); return 0; } type = ast_stream_get_type(ast_stream_topology_get_stream(sdp_state->proposed_capabilities->topology, stream_index)); if (type == AST_MEDIA_TYPE_AUDIO || type == AST_MEDIA_TYPE_VIDEO) { ast_rtp_instance_get_local_address(stream_state->instance, address); } else { return -1; } /* If an explicit global connection address is set use it here for the IP part */ if (!ast_sockaddr_isnull(&sdp_state->proposed_capabilities->connection_address)) { int port = ast_sockaddr_port(address); ast_sockaddr_copy(address, &sdp_state->proposed_capabilities->connection_address); ast_sockaddr_set_port(address, port); } return 0; } const struct ast_stream_topology *ast_sdp_state_get_joint_topology( const struct ast_sdp_state *sdp_state) { ast_assert(sdp_state != NULL); if (sdp_state->negotiated_capabilities) { return sdp_state->negotiated_capabilities->topology; } return sdp_state->proposed_capabilities->topology; } const struct ast_stream_topology *ast_sdp_state_get_local_topology( const struct ast_sdp_state *sdp_state) { ast_assert(sdp_state != NULL); return sdp_state->proposed_capabilities->topology; } const struct ast_sdp_options *ast_sdp_state_get_options( const struct ast_sdp_state *sdp_state) { ast_assert(sdp_state != NULL); return sdp_state->options; } /*! * \brief Merge two streams into a joint stream. * * \param local Our local stream * \param remote A remote stream * \retval NULL An error occurred * \retval non-NULL The joint stream created */ static struct ast_stream *merge_streams(const struct ast_stream *local, const struct ast_stream *remote) { struct ast_stream *joint_stream; struct ast_format_cap *joint_cap; struct ast_format_cap *local_cap; struct ast_format_cap *remote_cap; struct ast_str *local_buf = ast_str_alloca(128); struct ast_str *remote_buf = ast_str_alloca(128); struct ast_str *joint_buf = ast_str_alloca(128); joint_stream = ast_stream_alloc(ast_codec_media_type2str(ast_stream_get_type(remote)), ast_stream_get_type(remote)); if (!joint_stream) { return NULL; } if (!local) { return joint_stream; } joint_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); if (!joint_cap) { ast_stream_free(joint_stream); return NULL; } local_cap = ast_stream_get_formats(local); remote_cap = ast_stream_get_formats(remote); ast_format_cap_get_compatible(local_cap, remote_cap, joint_cap); ast_debug(3, "Combined local '%s' with remote '%s' to get joint '%s'. Joint has %zu formats\n", ast_format_cap_get_names(local_cap, &local_buf), ast_format_cap_get_names(remote_cap, &remote_buf), ast_format_cap_get_names(joint_cap, &joint_buf), ast_format_cap_count(joint_cap)); ast_stream_set_formats(joint_stream, joint_cap); ao2_ref(joint_cap, -1); return joint_stream; } /*! * \brief Get a local stream that corresponds with a remote stream. * * \param local The local topology * \param media_type The type of stream we are looking for * \param[in,out] media_indices Keeps track of where to start searching in the topology * \retval NULL No corresponding stream found * \retval non-NULL The corresponding stream */ static int get_corresponding_index(const struct ast_stream_topology *local, enum ast_media_type media_type, int *media_indices) { int i; int winner = -1; for (i = media_indices[media_type]; i < ast_stream_topology_get_count(local); ++i) { struct ast_stream *candidate; candidate = ast_stream_topology_get_stream(local, i); if (ast_stream_get_type(candidate) == media_type) { winner = i; break; } } media_indices[media_type] = i + 1; return winner; } /*! * \brief Merge existing stream capabilities and a new topology into joint capabilities. * * This is a bit complicated. The idea is that we already have some capabilities set, and * we've now been confronted with a new stream topology. We want to take what's been * presented to us and merge those new capabilities with our own. * * For each of the new streams, we try to find a corresponding stream in our current * capabilities. If we find one, then we get the compatible formats of the two streams * and create a new stream with those formats set. We then will re-use the underlying * media instance (such as an RTP instance) on this merged stream. * * The create_new parameter determines whether we should attempt to create new media * instances. * If we do not find a corresponding stream, then we create a new one. If the * create_new parameter is true, this created stream is made a clone of the new stream, * and a media instance is created. If the create_new parameter is not true, then the * created stream has no formats set and no media instance is created for it. * * \param current Current capabilities of the SDP state (may be NULL) * \param new_topology The new topology to base merged capabilities on * \param options The options set on the SDP state * \retval NULL An error occurred * \retval non-NULL The merged capabilities */ static struct sdp_state_capabilities *merge_capabilities(const struct sdp_state_capabilities *current, const struct ast_stream_topology *new_topology, const struct ast_sdp_options *options, int create_missing) { struct sdp_state_capabilities *joint_capabilities; struct ast_stream_topology *topology; int media_indices[AST_MEDIA_TYPE_END] = {0}; int i; ast_assert(current != NULL); joint_capabilities = ast_calloc(1, sizeof(*joint_capabilities)); if (!joint_capabilities) { return NULL; } joint_capabilities->topology = ast_stream_topology_alloc(); if (!joint_capabilities->topology) { goto fail; } AST_VECTOR_INIT(&joint_capabilities->streams, AST_VECTOR_SIZE(¤t->streams)); ast_sockaddr_copy(&joint_capabilities->connection_address, ¤t->connection_address); topology = current->topology; for (i = 0; i < ast_stream_topology_get_count(new_topology); ++i) { enum ast_media_type new_stream_type; struct ast_stream *new_stream; struct ast_stream *current_stream; struct ast_stream *joint_stream; struct sdp_state_stream *current_state_stream; struct sdp_state_stream *joint_state_stream; int current_index; joint_state_stream = ast_calloc(1, sizeof(*joint_state_stream)); if (!joint_state_stream) { goto fail; } new_stream = ast_stream_topology_get_stream(new_topology, i); new_stream_type = ast_stream_get_type(new_stream); current_index = get_corresponding_index(topology, new_stream_type, media_indices); if (current_index >= 0) { current_stream = ast_stream_topology_get_stream(topology, current_index); joint_stream = merge_streams(current_stream, new_stream); if (!joint_stream) { goto fail; } current_state_stream = AST_VECTOR_GET(¤t->streams, current_index); joint_state_stream->instance = ao2_bump(current_state_stream->instance); if (!ast_sockaddr_isnull(¤t_state_stream->connection_address)) { ast_sockaddr_copy(&joint_state_stream->connection_address, ¤t_state_stream->connection_address); } else { ast_sockaddr_setnull(&joint_state_stream->connection_address); } joint_state_stream->locally_held = current_state_stream->locally_held; } else if (create_missing) { /* We don't have a stream state that corresponds to the stream in the new topology, so * create a stream state as appropriate. */ joint_stream = ast_stream_clone(new_stream); if (!joint_stream) { goto fail; } if (new_stream_type == AST_MEDIA_TYPE_AUDIO || new_stream_type == AST_MEDIA_TYPE_VIDEO) { joint_state_stream->instance = create_rtp(options, new_stream_type); if (!joint_state_stream->instance) { goto fail; } } ast_sockaddr_setnull(&joint_state_stream->connection_address); joint_state_stream->locally_held = 0; } else { /* We don't have a stream that corresponds to the stream in the new topology. Create a * dummy stream to go in its place so that the resulting SDP created will contain * the stream but will have no port or codecs set */ joint_stream = ast_stream_alloc("dummy", new_stream_type); if (!joint_stream) { goto fail; } } ast_stream_topology_append_stream(joint_capabilities->topology, joint_stream); AST_VECTOR_APPEND(&joint_capabilities->streams, joint_state_stream); } return joint_capabilities; fail: sdp_state_capabilities_free(joint_capabilities); return NULL; } /*! * \brief Apply remote SDP's ICE information to our RTP session * * \param state The SDP state on which negotiation has taken place * \param options The SDP options we support * \param remote_sdp The SDP we most recently received * \param remote_m_line The stream on which we are examining ICE candidates */ static void update_ice(const struct ast_sdp_state *state, struct ast_rtp_instance *rtp, const struct ast_sdp_options *options, const struct ast_sdp *remote_sdp, const struct ast_sdp_m_line *remote_m_line) { struct ast_rtp_engine_ice *ice; const struct ast_sdp_a_line *attr; unsigned int attr_i; /* If ICE support is not enabled or available exit early */ if (ast_sdp_options_get_ice(options) != AST_SDP_ICE_ENABLED_STANDARD || !(ice = ast_rtp_instance_get_ice(rtp))) { return; } attr = ast_sdp_m_find_attribute(remote_m_line, "ice-ufrag", -1); if (!attr) { attr = ast_sdp_find_attribute(remote_sdp, "ice-ufrag", -1); } if (attr) { ice->set_authentication(rtp, attr->value, NULL); } else { return; } attr = ast_sdp_m_find_attribute(remote_m_line, "ice-pwd", -1); if (!attr) { attr = ast_sdp_find_attribute(remote_sdp, "ice-pwd", -1); } if (attr) { ice->set_authentication(rtp, NULL, attr->value); } else { return; } if (ast_sdp_m_find_attribute(remote_m_line, "ice-lite", -1)) { ice->ice_lite(rtp); } /* Find all of the candidates */ for (attr_i = 0; attr_i < ast_sdp_m_get_a_count(remote_m_line); ++attr_i) { char foundation[32]; char transport[32]; char address[INET6_ADDRSTRLEN + 1]; char cand_type[6]; char relay_address[INET6_ADDRSTRLEN + 1] = ""; unsigned int port; unsigned int relay_port = 0; struct ast_rtp_engine_ice_candidate candidate = { 0, }; attr = ast_sdp_m_get_a(remote_m_line, attr_i); /* If this is not a candidate line skip it */ if (strcmp(attr->name, "candidate")) { continue; } if (sscanf(attr->value, "%31s %30u %31s %30u %46s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport, (unsigned *)&candidate.priority, address, &port, cand_type, relay_address, &relay_port) < 7) { /* Candidate did not parse properly */ continue; } if (ast_sdp_options_get_rtcp_mux(options) && ast_sdp_m_find_attribute(remote_m_line, "rtcp-mux", -1) && candidate.id > 1) { /* Remote side may have offered RTP and RTCP candidates. However, if we're using RTCP MUX, * then we should ignore RTCP candidates. */ continue; } candidate.foundation = foundation; candidate.transport = transport; ast_sockaddr_parse(&candidate.address, address, PARSE_PORT_FORBID); ast_sockaddr_set_port(&candidate.address, port); if (!strcasecmp(cand_type, "host")) { candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_HOST; } else if (!strcasecmp(cand_type, "srflx")) { candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX; } else if (!strcasecmp(cand_type, "relay")) { candidate.type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED; } else { continue; } if (!ast_strlen_zero(relay_address)) { ast_sockaddr_parse(&candidate.relay_address, relay_address, PARSE_PORT_FORBID); } if (relay_port) { ast_sockaddr_set_port(&candidate.relay_address, relay_port); } ice->add_remote_candidate(rtp, &candidate); } if (state->role == SDP_ROLE_OFFERER) { ice->set_role(rtp, AST_RTP_ICE_ROLE_CONTROLLING); } else { ice->set_role(rtp, AST_RTP_ICE_ROLE_CONTROLLED); } ice->start(rtp); } /*! * \brief Update RTP instances based on merged SDPs * * RTP instances, when first allocated, cannot make assumptions about what the other * side supports and thus has to go with some default behaviors. This function gets * called after we know both what we support and what the remote endpoint supports. * This way, we can update the RTP instance to reflect what is supported by both * sides. * * \param state The SDP state in which SDPs have been negotiated * \param rtp The RTP instance that is being updated * \param options Our locally-supported SDP options * \param remote_sdp The SDP we most recently received * \param remote_m_line The remote SDP stream that corresponds to the RTP instance we are modifying */ static void update_rtp_after_merge(const struct ast_sdp_state *state, struct ast_rtp_instance *rtp, const struct ast_sdp_options *options, const struct ast_sdp *remote_sdp, const struct ast_sdp_m_line *remote_m_line) { if (ast_sdp_options_get_rtcp_mux(options) && ast_sdp_m_find_attribute(remote_m_line, "rtcp-mux", -1)) { ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_MUX); } else { ast_rtp_instance_set_prop(rtp, AST_RTP_PROPERTY_RTCP, AST_RTP_INSTANCE_RTCP_STANDARD); } if (ast_sdp_options_get_ice(options) == AST_SDP_ICE_ENABLED_STANDARD) { update_ice(state, rtp, options, remote_sdp, remote_m_line); } } static void set_negotiated_capabilities(struct ast_sdp_state *sdp_state, struct sdp_state_capabilities *new_capabilities) { struct sdp_state_capabilities *old_capabilities = sdp_state->negotiated_capabilities; sdp_state->negotiated_capabilities = new_capabilities; sdp_state_capabilities_free(old_capabilities); } static void set_proposed_capabilities(struct ast_sdp_state *sdp_state, struct sdp_state_capabilities *new_capabilities) { struct sdp_state_capabilities *old_capabilities = sdp_state->proposed_capabilities; sdp_state->proposed_capabilities = new_capabilities; sdp_state_capabilities_free(old_capabilities); } static struct ast_sdp *sdp_create_from_state(const struct ast_sdp_state *sdp_state, const struct sdp_state_capabilities *capabilities); /*! * \brief Merge SDPs into a joint SDP. * * This function is used to take a remote SDP and merge it with our local * capabilities to produce a new local SDP. After creating the new local SDP, * it then iterates through media instances and updates them as necessary. For * instance, if a specific RTP feature is supported by both us and the far end, * then we can ensure that the feature is enabled. * * \param sdp_state The current SDP state * \retval -1 Failure * \retval 0 Success */ static int merge_sdps(struct ast_sdp_state *sdp_state, const struct ast_sdp *remote_sdp) { struct sdp_state_capabilities *joint_capabilities; int i; sdp_state->remote_capabilities = ast_get_topology_from_sdp(remote_sdp); if (!sdp_state->remote_capabilities) { return -1; } joint_capabilities = merge_capabilities(sdp_state->proposed_capabilities, sdp_state->remote_capabilities, sdp_state->options, 0); if (!joint_capabilities) { return -1; } set_negotiated_capabilities(sdp_state, joint_capabilities); if (sdp_state->local_sdp) { ast_sdp_free(sdp_state->local_sdp); sdp_state->local_sdp = NULL; } sdp_state->local_sdp = sdp_create_from_state(sdp_state, joint_capabilities); if (!sdp_state->local_sdp) { return -1; } for (i = 0; i < AST_VECTOR_SIZE(&joint_capabilities->streams); ++i) { struct sdp_state_stream *state_stream; enum ast_media_type stream_type; stream_type = ast_stream_get_type(ast_stream_topology_get_stream(joint_capabilities->topology, i)); state_stream = AST_VECTOR_GET(&joint_capabilities->streams, i); if ((stream_type == AST_MEDIA_TYPE_AUDIO || stream_type == AST_MEDIA_TYPE_VIDEO) && state_stream->instance) { update_rtp_after_merge(sdp_state, state_stream->instance, sdp_state->options, remote_sdp, ast_sdp_get_m(remote_sdp, i)); } } return 0; } const struct ast_sdp *ast_sdp_state_get_local_sdp(struct ast_sdp_state *sdp_state) { ast_assert(sdp_state != NULL); if (sdp_state->role == SDP_ROLE_NOT_SET) { ast_assert(sdp_state->local_sdp == NULL); sdp_state->role = SDP_ROLE_OFFERER; sdp_state->local_sdp = sdp_create_from_state(sdp_state, sdp_state->proposed_capabilities); } return sdp_state->local_sdp; } const void *ast_sdp_state_get_local_sdp_impl(struct ast_sdp_state *sdp_state) { const struct ast_sdp *sdp = ast_sdp_state_get_local_sdp(sdp_state); if (!sdp) { return NULL; } return ast_sdp_translator_from_sdp(sdp_state->translator, sdp); } void ast_sdp_state_set_remote_sdp(struct ast_sdp_state *sdp_state, const struct ast_sdp *sdp) { ast_assert(sdp_state != NULL); if (sdp_state->role == SDP_ROLE_NOT_SET) { sdp_state->role = SDP_ROLE_ANSWERER; } merge_sdps(sdp_state, sdp); } int ast_sdp_state_set_remote_sdp_from_impl(struct ast_sdp_state *sdp_state, void *remote) { struct ast_sdp *sdp; ast_assert(sdp_state != NULL); sdp = ast_sdp_translator_to_sdp(sdp_state->translator, remote); if (!sdp) { return -1; } ast_sdp_state_set_remote_sdp(sdp_state, sdp); return 0; } int ast_sdp_state_reset(struct ast_sdp_state *sdp_state) { ast_assert(sdp_state != NULL); ast_sdp_free(sdp_state->local_sdp); sdp_state->local_sdp = NULL; ast_stream_topology_free(sdp_state->remote_capabilities); sdp_state->remote_capabilities = NULL; set_proposed_capabilities(sdp_state, NULL); sdp_state->role = SDP_ROLE_NOT_SET; return 0; } int ast_sdp_state_update_local_topology(struct ast_sdp_state *sdp_state, struct ast_stream_topology *streams) { struct sdp_state_capabilities *capabilities; ast_assert(sdp_state != NULL); ast_assert(streams != NULL); capabilities = merge_capabilities(sdp_state->proposed_capabilities, streams, sdp_state->options, 1); if (!capabilities) { return -1; } set_proposed_capabilities(sdp_state, capabilities); return 0; } void ast_sdp_state_set_local_address(struct ast_sdp_state *sdp_state, struct ast_sockaddr *address) { ast_assert(sdp_state != NULL); if (!address) { ast_sockaddr_setnull(&sdp_state->proposed_capabilities->connection_address); } else { ast_sockaddr_copy(&sdp_state->proposed_capabilities->connection_address, address); } } int ast_sdp_state_set_connection_address(struct ast_sdp_state *sdp_state, int stream_index, struct ast_sockaddr *address) { struct sdp_state_stream *stream_state; ast_assert(sdp_state != NULL); stream_state = sdp_state_get_stream(sdp_state, stream_index); if (!stream_state) { return -1; } if (!address) { ast_sockaddr_setnull(&stream_state->connection_address); } else { ast_sockaddr_copy(&stream_state->connection_address, address); } return 0; } void ast_sdp_state_set_locally_held(struct ast_sdp_state *sdp_state, int stream_index, unsigned int locally_held) { struct sdp_state_stream *stream_state; ast_assert(sdp_state != NULL); stream_state = sdp_state_get_stream(sdp_state, stream_index); if (!stream_state) { return; } stream_state->locally_held = locally_held; } unsigned int ast_sdp_state_get_locally_held(const struct ast_sdp_state *sdp_state, int stream_index) { struct sdp_state_stream *stream_state; ast_assert(sdp_state != NULL); stream_state = sdp_state_get_stream(sdp_state, stream_index); if (!stream_state) { return 0; } return stream_state->locally_held; } static int sdp_add_m_from_rtp_stream(struct ast_sdp *sdp, const struct ast_sdp_state *sdp_state, const struct ast_sdp_options *options, const struct sdp_state_capabilities *capabilities, int stream_index) { struct ast_stream *stream; struct ast_sdp_m_line *m_line; struct ast_format_cap *caps; int i; int rtp_code; int min_packet_size = 0; int max_packet_size = 0; enum ast_media_type media_type; char tmp[64]; struct ast_sockaddr address_rtp; struct ast_rtp_instance *rtp; struct ast_sdp_a_line *a_line; stream = ast_stream_topology_get_stream(capabilities->topology, stream_index); rtp = AST_VECTOR_GET(&capabilities->streams, stream_index)->instance; ast_assert(sdp && options && stream); media_type = ast_stream_get_type(stream); if (rtp) { if (ast_sdp_state_get_stream_connection_address(sdp_state, 0, &address_rtp)) { return -1; } } else { ast_sockaddr_setnull(&address_rtp); } m_line = ast_sdp_m_alloc( ast_codec_media_type2str(ast_stream_get_type(stream)), ast_sockaddr_port(&address_rtp), 1, options->encryption != AST_SDP_ENCRYPTION_DISABLED ? "RTP/SAVP" : "RTP/AVP", NULL); if (!m_line) { return -1; } caps = ast_stream_get_formats(stream); for (i = 0; i < ast_format_cap_count(caps); i++) { struct ast_format *format = ast_format_cap_get_format(caps, i); if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(rtp), 1, format, 0)) == -1) { ast_log(LOG_WARNING,"Unable to get rtp codec payload code for %s\n", ast_format_get_name(format)); ao2_ref(format, -1); continue; } if (ast_sdp_m_add_format(m_line, options, rtp_code, 1, format, 0)) { ast_sdp_m_free(m_line); ao2_ref(format, -1); return -1; } if (ast_format_get_maximum_ms(format) && ((ast_format_get_maximum_ms(format) < max_packet_size) || !max_packet_size)) { max_packet_size = ast_format_get_maximum_ms(format); } ao2_ref(format, -1); } if (rtp && media_type != AST_MEDIA_TYPE_VIDEO) { for (i = 1LL; i <= AST_RTP_MAX; i <<= 1) { if (!(options->telephone_event & i)) { continue; } rtp_code = ast_rtp_codecs_payload_code( ast_rtp_instance_get_codecs(rtp), 0, NULL, i); if (rtp_code == -1) { continue; } if (ast_sdp_m_add_format(m_line, options, rtp_code, 0, NULL, i)) { continue; } if (i == AST_RTP_DTMF) { snprintf(tmp, sizeof(tmp), "%d 0-16", rtp_code); a_line = ast_sdp_a_alloc("fmtp", tmp); if (!a_line || ast_sdp_m_add_a(m_line, a_line)) { ast_sdp_a_free(a_line); ast_sdp_m_free(m_line); return -1; } } } } if (ast_sdp_m_get_a_count(m_line) == 0) { return 0; } /* If ptime is set add it as an attribute */ min_packet_size = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(rtp)); if (!min_packet_size) { min_packet_size = ast_format_cap_get_framing(caps); } if (min_packet_size) { snprintf(tmp, sizeof(tmp), "%d", min_packet_size); a_line = ast_sdp_a_alloc("ptime", tmp); if (!a_line || ast_sdp_m_add_a(m_line, a_line)) { ast_sdp_a_free(a_line); ast_sdp_m_free(m_line); return -1; } } if (max_packet_size) { snprintf(tmp, sizeof(tmp), "%d", max_packet_size); a_line = ast_sdp_a_alloc("maxptime", tmp); if (!a_line || ast_sdp_m_add_a(m_line, a_line)) { ast_sdp_a_free(a_line); ast_sdp_m_free(m_line); return -1; } } a_line = ast_sdp_a_alloc(ast_sdp_state_get_locally_held(sdp_state, stream_index) ? "sendonly" : "sendrecv", ""); if (!a_line || ast_sdp_m_add_a(m_line, a_line)) { ast_sdp_a_free(a_line); ast_sdp_m_free(m_line); return -1; } if (ast_sdp_add_m(sdp, m_line)) { ast_sdp_m_free(m_line); return -1; } return 0; } /*! * \brief Create an SDP based on current SDP state * * \param sdp_state The current SDP state * \retval NULL Failed to create SDP * \retval non-NULL Newly-created SDP */ static struct ast_sdp *sdp_create_from_state(const struct ast_sdp_state *sdp_state, const struct sdp_state_capabilities *capabilities) { struct ast_sdp *sdp = NULL; struct ast_stream_topology *topology; const struct ast_sdp_options *options; int stream_num; struct ast_sdp_o_line *o_line = NULL; struct ast_sdp_c_line *c_line = NULL; struct ast_sdp_s_line *s_line = NULL; struct ast_sdp_t_line *t_line = NULL; char *address_type; struct timeval tv = ast_tvnow(); uint32_t t; int stream_count; options = ast_sdp_state_get_options(sdp_state); topology = capabilities->topology; t = tv.tv_sec + 2208988800UL; address_type = (strchr(options->media_address, ':') ? "IP6" : "IP4"); o_line = ast_sdp_o_alloc(options->sdpowner, t, t, address_type, options->media_address); if (!o_line) { goto error; } c_line = ast_sdp_c_alloc(address_type, options->media_address); if (!c_line) { goto error; } s_line = ast_sdp_s_alloc(options->sdpsession); if (!s_line) { goto error; } sdp = ast_sdp_alloc(o_line, c_line, s_line, NULL); if (!sdp) { goto error; } stream_count = ast_stream_topology_get_count(topology); for (stream_num = 0; stream_num < stream_count; stream_num++) { enum ast_media_type type = ast_stream_get_type(ast_stream_topology_get_stream(topology, stream_num)); if (type == AST_MEDIA_TYPE_AUDIO || type == AST_MEDIA_TYPE_VIDEO) { if (sdp_add_m_from_rtp_stream(sdp, sdp_state, options, capabilities, stream_num)) { goto error; } } } return sdp; error: if (sdp) { ast_sdp_free(sdp); } else { ast_sdp_t_free(t_line); ast_sdp_s_free(s_line); ast_sdp_c_free(c_line); ast_sdp_o_free(o_line); } return NULL; }