Per the RFC when an outgoing re-INVITE is done we should only terminate the dialog if a 481 or 408 is received. ASTERISK-29033 Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503
4760 lines
158 KiB
C
4760 lines
158 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2013, Digium, Inc.
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*
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* Mark Michelson <mmichelson@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*** MODULEINFO
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<depend>pjproject</depend>
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<depend>res_pjsip</depend>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <pjsip.h>
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#include <pjsip_ua.h>
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#include <pjlib.h>
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#include <pjmedia.h>
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#include "asterisk/res_pjsip.h"
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#include "asterisk/res_pjsip_session.h"
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#include "asterisk/res_pjsip_session_caps.h"
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#include "asterisk/callerid.h"
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#include "asterisk/datastore.h"
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#include "asterisk/module.h"
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#include "asterisk/logger.h"
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#include "asterisk/res_pjsip.h"
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#include "asterisk/astobj2.h"
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#include "asterisk/lock.h"
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#include "asterisk/uuid.h"
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#include "asterisk/pbx.h"
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#include "asterisk/taskprocessor.h"
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#include "asterisk/causes.h"
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#include "asterisk/sdp_srtp.h"
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#include "asterisk/dsp.h"
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#include "asterisk/acl.h"
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#include "asterisk/features_config.h"
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#include "asterisk/pickup.h"
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#include "asterisk/test.h"
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#include "asterisk/stream.h"
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#include "asterisk/vector.h"
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#define SDP_HANDLER_BUCKETS 11
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#define MOD_DATA_ON_RESPONSE "on_response"
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#define MOD_DATA_NAT_HOOK "nat_hook"
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/* Most common case is one audio and one video stream */
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#define DEFAULT_NUM_SESSION_MEDIA 2
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/* Some forward declarations */
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static void handle_session_begin(struct ast_sip_session *session);
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static void handle_session_end(struct ast_sip_session *session);
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static void handle_session_destroy(struct ast_sip_session *session);
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static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata);
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static void handle_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata,
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enum ast_sip_session_response_priority response_priority);
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static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata,
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enum ast_sip_session_response_priority response_priority);
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static void handle_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata);
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static void handle_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata);
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static int sip_session_refresh(struct ast_sip_session *session,
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ast_sip_session_request_creation_cb on_request_creation,
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ast_sip_session_sdp_creation_cb on_sdp_creation,
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ast_sip_session_response_cb on_response,
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enum ast_sip_session_refresh_method method, int generate_new_sdp,
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struct ast_sip_session_media_state *media_state,
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int queued);
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/*! \brief NAT hook for modifying outgoing messages with SDP */
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static struct ast_sip_nat_hook *nat_hook;
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/*!
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* \brief Registered SDP stream handlers
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*
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* This container is keyed on stream types. Each
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* object in the container is a linked list of
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* handlers for the stream type.
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*/
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static struct ao2_container *sdp_handlers;
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/*!
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* These are the objects in the sdp_handlers container
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*/
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struct sdp_handler_list {
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/* The list of handlers to visit */
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AST_LIST_HEAD_NOLOCK(, ast_sip_session_sdp_handler) list;
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/* The handlers in this list handle streams of this type */
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char stream_type[1];
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};
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static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer);
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static int sdp_handler_list_hash(const void *obj, int flags)
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{
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const struct sdp_handler_list *handler_list = obj;
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const char *stream_type = flags & OBJ_KEY ? obj : handler_list->stream_type;
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return ast_str_hash(stream_type);
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}
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const char *ast_sip_session_get_name(const struct ast_sip_session *session)
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{
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if (!session) {
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return "(null session)";
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}
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if (session->channel) {
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return ast_channel_name(session->channel);
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} else if (session->endpoint) {
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return ast_sorcery_object_get_id(session->endpoint);
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} else {
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return "unknown";
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}
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}
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static int sdp_handler_list_cmp(void *obj, void *arg, int flags)
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{
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struct sdp_handler_list *handler_list1 = obj;
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struct sdp_handler_list *handler_list2 = arg;
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const char *stream_type2 = flags & OBJ_KEY ? arg : handler_list2->stream_type;
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return strcmp(handler_list1->stream_type, stream_type2) ? 0 : CMP_MATCH | CMP_STOP;
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}
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int ast_sip_session_register_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type)
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{
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RAII_VAR(struct sdp_handler_list *, handler_list,
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ao2_find(sdp_handlers, stream_type, OBJ_KEY), ao2_cleanup);
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SCOPED_AO2LOCK(lock, sdp_handlers);
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if (handler_list) {
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struct ast_sip_session_sdp_handler *iter;
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/* Check if this handler is already registered for this stream type */
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AST_LIST_TRAVERSE(&handler_list->list, iter, next) {
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if (!strcmp(iter->id, handler->id)) {
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ast_log(LOG_WARNING, "Handler '%s' already registered for stream type '%s'.\n", handler->id, stream_type);
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return -1;
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}
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}
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AST_LIST_INSERT_TAIL(&handler_list->list, handler, next);
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ast_debug(1, "Registered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
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return 0;
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}
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/* No stream of this type has been registered yet, so we need to create a new list */
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handler_list = ao2_alloc(sizeof(*handler_list) + strlen(stream_type), NULL);
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if (!handler_list) {
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return -1;
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}
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/* Safe use of strcpy */
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strcpy(handler_list->stream_type, stream_type);
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AST_LIST_HEAD_INIT_NOLOCK(&handler_list->list);
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AST_LIST_INSERT_TAIL(&handler_list->list, handler, next);
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if (!ao2_link(sdp_handlers, handler_list)) {
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return -1;
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}
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ast_debug(1, "Registered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
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return 0;
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}
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static int remove_handler(void *obj, void *arg, void *data, int flags)
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{
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struct sdp_handler_list *handler_list = obj;
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struct ast_sip_session_sdp_handler *handler = data;
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struct ast_sip_session_sdp_handler *iter;
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const char *stream_type = arg;
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AST_LIST_TRAVERSE_SAFE_BEGIN(&handler_list->list, iter, next) {
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if (!strcmp(iter->id, handler->id)) {
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AST_LIST_REMOVE_CURRENT(next);
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ast_debug(1, "Unregistered SDP stream handler '%s' for stream type '%s'\n", handler->id, stream_type);
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}
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}
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AST_LIST_TRAVERSE_SAFE_END;
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if (AST_LIST_EMPTY(&handler_list->list)) {
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ast_debug(3, "No more handlers exist for stream type '%s'\n", stream_type);
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return CMP_MATCH;
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} else {
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return CMP_STOP;
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}
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}
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void ast_sip_session_unregister_sdp_handler(struct ast_sip_session_sdp_handler *handler, const char *stream_type)
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{
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ao2_callback_data(sdp_handlers, OBJ_KEY | OBJ_UNLINK | OBJ_NODATA, remove_handler, (void *)stream_type, handler);
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}
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static int media_stats_local_ssrc_cmp(
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const struct ast_rtp_instance_stats *vec_elem, const struct ast_rtp_instance_stats *srch)
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{
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if (vec_elem->local_ssrc == srch->local_ssrc) {
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return 1;
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}
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return 0;
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}
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static struct ast_sip_session_media_state *internal_sip_session_media_state_alloc(
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size_t sessions, size_t read_callbacks)
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{
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struct ast_sip_session_media_state *media_state;
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media_state = ast_calloc(1, sizeof(*media_state));
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if (!media_state) {
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return NULL;
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}
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if (AST_VECTOR_INIT(&media_state->sessions, sessions) < 0) {
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ast_free(media_state);
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return NULL;
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}
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if (AST_VECTOR_INIT(&media_state->read_callbacks, read_callbacks) < 0) {
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AST_VECTOR_FREE(&media_state->sessions);
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ast_free(media_state);
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return NULL;
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}
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return media_state;
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}
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struct ast_sip_session_media_state *ast_sip_session_media_state_alloc(void)
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{
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return internal_sip_session_media_state_alloc(
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DEFAULT_NUM_SESSION_MEDIA, DEFAULT_NUM_SESSION_MEDIA);
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}
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void ast_sip_session_media_stats_save(struct ast_sip_session *sip_session, struct ast_sip_session_media_state *media_state)
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{
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int i;
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int ret;
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if (!media_state || !sip_session) {
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return;
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}
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for (i = 0; i < AST_VECTOR_SIZE(&media_state->sessions); i++) {
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struct ast_rtp_instance_stats *stats_tmp = NULL;
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struct ast_sip_session_media *media = AST_VECTOR_GET(&media_state->sessions, i);
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if (!media || !media->rtp) {
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continue;
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}
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stats_tmp = ast_calloc(1, sizeof(struct ast_rtp_instance_stats));
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if (!stats_tmp) {
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return;
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}
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ret = ast_rtp_instance_get_stats(media->rtp, stats_tmp, AST_RTP_INSTANCE_STAT_ALL);
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if (ret) {
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ast_free(stats_tmp);
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continue;
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}
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/* remove all the duplicated stats if exist */
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AST_VECTOR_REMOVE_CMP_UNORDERED(&sip_session->media_stats, stats_tmp, media_stats_local_ssrc_cmp, ast_free);
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AST_VECTOR_APPEND(&sip_session->media_stats, stats_tmp);
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}
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}
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void ast_sip_session_media_state_reset(struct ast_sip_session_media_state *media_state)
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{
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int index;
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if (!media_state) {
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return;
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}
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AST_VECTOR_RESET(&media_state->sessions, ao2_cleanup);
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AST_VECTOR_RESET(&media_state->read_callbacks, AST_VECTOR_ELEM_CLEANUP_NOOP);
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for (index = 0; index < AST_MEDIA_TYPE_END; ++index) {
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media_state->default_session[index] = NULL;
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}
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ast_stream_topology_free(media_state->topology);
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media_state->topology = NULL;
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}
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struct ast_sip_session_media_state *ast_sip_session_media_state_clone(const struct ast_sip_session_media_state *media_state)
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{
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struct ast_sip_session_media_state *cloned;
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int index;
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if (!media_state) {
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return NULL;
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}
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cloned = internal_sip_session_media_state_alloc(
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AST_VECTOR_SIZE(&media_state->sessions),
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AST_VECTOR_SIZE(&media_state->read_callbacks));
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if (!cloned) {
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return NULL;
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}
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if (media_state->topology) {
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cloned->topology = ast_stream_topology_clone(media_state->topology);
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if (!cloned->topology) {
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ast_sip_session_media_state_free(cloned);
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return NULL;
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}
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}
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for (index = 0; index < AST_VECTOR_SIZE(&media_state->sessions); ++index) {
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struct ast_sip_session_media *session_media = AST_VECTOR_GET(&media_state->sessions, index);
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enum ast_media_type type = ast_stream_get_type(ast_stream_topology_get_stream(cloned->topology, index));
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ao2_bump(session_media);
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if (AST_VECTOR_REPLACE(&cloned->sessions, index, session_media)) {
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ao2_cleanup(session_media);
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}
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if (ast_stream_get_state(ast_stream_topology_get_stream(cloned->topology, index)) != AST_STREAM_STATE_REMOVED &&
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!cloned->default_session[type]) {
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cloned->default_session[type] = session_media;
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}
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}
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for (index = 0; index < AST_VECTOR_SIZE(&media_state->read_callbacks); ++index) {
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struct ast_sip_session_media_read_callback_state *read_callback = AST_VECTOR_GET_ADDR(&media_state->read_callbacks, index);
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AST_VECTOR_REPLACE(&cloned->read_callbacks, index, *read_callback);
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}
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return cloned;
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}
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void ast_sip_session_media_state_free(struct ast_sip_session_media_state *media_state)
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{
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if (!media_state) {
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return;
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}
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|
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/* This will reset the internal state so we only have to free persistent things */
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ast_sip_session_media_state_reset(media_state);
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AST_VECTOR_FREE(&media_state->sessions);
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AST_VECTOR_FREE(&media_state->read_callbacks);
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ast_free(media_state);
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}
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|
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int ast_sip_session_is_pending_stream_default(const struct ast_sip_session *session, const struct ast_stream *stream)
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{
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int index;
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if (!session->pending_media_state->topology) {
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ast_log(LOG_WARNING, "Pending topology was NULL for channel '%s'\n",
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session->channel ? ast_channel_name(session->channel) : "unknown");
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return 0;
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}
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if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
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return 0;
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}
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for (index = 0; index < ast_stream_topology_get_count(session->pending_media_state->topology); ++index) {
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if (ast_stream_get_type(ast_stream_topology_get_stream(session->pending_media_state->topology, index)) !=
|
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ast_stream_get_type(stream)) {
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continue;
|
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}
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|
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return ast_stream_topology_get_stream(session->pending_media_state->topology, index) == stream ? 1 : 0;
|
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}
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|
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return 0;
|
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}
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|
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int ast_sip_session_media_add_read_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
|
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int fd, ast_sip_session_media_read_cb callback)
|
|
{
|
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struct ast_sip_session_media_read_callback_state callback_state = {
|
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.fd = fd,
|
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.read_callback = callback,
|
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.session = session_media,
|
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};
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|
|
/* The contents of the vector are whole structs and not pointers */
|
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return AST_VECTOR_APPEND(&session->pending_media_state->read_callbacks, callback_state);
|
|
}
|
|
|
|
int ast_sip_session_media_set_write_callback(struct ast_sip_session *session, struct ast_sip_session_media *session_media,
|
|
ast_sip_session_media_write_cb callback)
|
|
{
|
|
if (session_media->write_callback) {
|
|
if (session_media->write_callback == callback) {
|
|
return 0;
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
session_media->write_callback = callback;
|
|
|
|
return 0;
|
|
}
|
|
|
|
struct ast_sip_session_media *ast_sip_session_media_get_transport(struct ast_sip_session *session, struct ast_sip_session_media *session_media)
|
|
{
|
|
int index;
|
|
|
|
if (!session->endpoint->media.bundle || ast_strlen_zero(session_media->mid)) {
|
|
return session_media;
|
|
}
|
|
|
|
for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
|
|
struct ast_sip_session_media *bundle_group_session_media;
|
|
|
|
bundle_group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
|
|
|
|
/* The first session which is in the bundle group is considered the authoritative session for transport */
|
|
if (bundle_group_session_media->bundle_group == session_media->bundle_group) {
|
|
return bundle_group_session_media;
|
|
}
|
|
}
|
|
|
|
return session_media;
|
|
}
|
|
|
|
/*!
|
|
* \brief Set an SDP stream handler for a corresponding session media.
|
|
*
|
|
* \note Always use this function to set the SDP handler for a session media.
|
|
*
|
|
* This function will properly free resources on the SDP handler currently being
|
|
* used by the session media, then set the session media to use the new SDP
|
|
* handler.
|
|
*/
|
|
static void session_media_set_handler(struct ast_sip_session_media *session_media,
|
|
struct ast_sip_session_sdp_handler *handler)
|
|
{
|
|
ast_assert(session_media->handler != handler);
|
|
|
|
if (session_media->handler) {
|
|
session_media->handler->stream_destroy(session_media);
|
|
}
|
|
session_media->handler = handler;
|
|
}
|
|
|
|
static int stream_destroy(void *obj, void *arg, int flags)
|
|
{
|
|
struct sdp_handler_list *handler_list = obj;
|
|
struct ast_sip_session_media *session_media = arg;
|
|
struct ast_sip_session_sdp_handler *handler;
|
|
|
|
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
|
|
handler->stream_destroy(session_media);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void session_media_dtor(void *obj)
|
|
{
|
|
struct ast_sip_session_media *session_media = obj;
|
|
|
|
/* It is possible for multiple handlers to have allocated memory on the
|
|
* session media (usually through a stream changing types). Therefore, we
|
|
* traverse all the SDP handlers and let them all call stream_destroy on
|
|
* the session_media
|
|
*/
|
|
ao2_callback(sdp_handlers, 0, stream_destroy, session_media);
|
|
|
|
if (session_media->srtp) {
|
|
ast_sdp_srtp_destroy(session_media->srtp);
|
|
}
|
|
|
|
ast_free(session_media->mid);
|
|
ast_free(session_media->remote_mslabel);
|
|
}
|
|
|
|
struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_session *session,
|
|
struct ast_sip_session_media_state *media_state, enum ast_media_type type, int position)
|
|
{
|
|
struct ast_sip_session_media *session_media = NULL;
|
|
|
|
/* It is possible for this media state to already contain a session for the stream. If this
|
|
* is the case we simply return it.
|
|
*/
|
|
if (position < AST_VECTOR_SIZE(&media_state->sessions)) {
|
|
return AST_VECTOR_GET(&media_state->sessions, position);
|
|
}
|
|
|
|
/* Determine if we can reuse the session media from the active media state if present */
|
|
if (position < AST_VECTOR_SIZE(&session->active_media_state->sessions)) {
|
|
session_media = AST_VECTOR_GET(&session->active_media_state->sessions, position);
|
|
/* A stream can never exist without an accompanying media session */
|
|
if (session_media->type == type) {
|
|
ao2_ref(session_media, +1);
|
|
/*
|
|
* If this session_media was previously removed, its bundle group was probably reset
|
|
* to -1 so if bundling is enabled on the endpoint, we need to reset it to 0, set
|
|
* the bundled flag and reset its mid.
|
|
*/
|
|
if (session->endpoint->media.bundle && session_media->bundle_group == -1) {
|
|
session_media->bundled = session->endpoint->media.webrtc;
|
|
session_media->bundle_group = 0;
|
|
ast_free(session_media->mid);
|
|
if (ast_asprintf(&session_media->mid, "%s-%d", ast_codec_media_type2str(type), position) < 0) {
|
|
ao2_ref(session_media, -1);
|
|
return NULL;
|
|
}
|
|
}
|
|
} else {
|
|
session_media = NULL;
|
|
}
|
|
}
|
|
|
|
if (!session_media) {
|
|
/* No existing media session we can use so create a new one */
|
|
session_media = ao2_alloc_options(sizeof(*session_media), session_media_dtor, AO2_ALLOC_OPT_LOCK_NOLOCK);
|
|
if (!session_media) {
|
|
return NULL;
|
|
}
|
|
|
|
session_media->encryption = session->endpoint->media.rtp.encryption;
|
|
session_media->remote_ice = session->endpoint->media.rtp.ice_support;
|
|
session_media->remote_rtcp_mux = session->endpoint->media.rtcp_mux;
|
|
session_media->keepalive_sched_id = -1;
|
|
session_media->timeout_sched_id = -1;
|
|
session_media->type = type;
|
|
session_media->stream_num = position;
|
|
|
|
if (session->endpoint->media.bundle) {
|
|
/* This is a new stream so create a new mid based on media type and position, which makes it unique.
|
|
* If this is the result of an offer the mid will just end up getting replaced.
|
|
*/
|
|
if (ast_asprintf(&session_media->mid, "%s-%d", ast_codec_media_type2str(type), position) < 0) {
|
|
ao2_ref(session_media, -1);
|
|
return NULL;
|
|
}
|
|
session_media->bundle_group = 0;
|
|
|
|
/* Some WebRTC clients can't handle an offer to bundle media streams. Instead they expect them to
|
|
* already be bundled. Every client handles this scenario though so if WebRTC is enabled just go
|
|
* ahead and treat the streams as having already been bundled.
|
|
*/
|
|
session_media->bundled = session->endpoint->media.webrtc;
|
|
} else {
|
|
session_media->bundle_group = -1;
|
|
}
|
|
}
|
|
|
|
if (AST_VECTOR_REPLACE(&media_state->sessions, position, session_media)) {
|
|
ao2_ref(session_media, -1);
|
|
|
|
return NULL;
|
|
}
|
|
|
|
/* If this stream will be active in some way and it is the first of this type then consider this the default media session to match */
|
|
if (!media_state->default_session[type] && ast_stream_get_state(ast_stream_topology_get_stream(media_state->topology, position)) != AST_STREAM_STATE_REMOVED) {
|
|
media_state->default_session[type] = session_media;
|
|
}
|
|
|
|
return session_media;
|
|
}
|
|
|
|
static int is_stream_limitation_reached(enum ast_media_type type, const struct ast_sip_endpoint *endpoint, int *type_streams)
|
|
{
|
|
switch (type) {
|
|
case AST_MEDIA_TYPE_AUDIO:
|
|
return !(type_streams[type] < endpoint->media.max_audio_streams);
|
|
case AST_MEDIA_TYPE_VIDEO:
|
|
return !(type_streams[type] < endpoint->media.max_video_streams);
|
|
case AST_MEDIA_TYPE_IMAGE:
|
|
/* We don't have an option for image (T.38) streams so cap it to one. */
|
|
return (type_streams[type] > 0);
|
|
case AST_MEDIA_TYPE_UNKNOWN:
|
|
case AST_MEDIA_TYPE_TEXT:
|
|
default:
|
|
/* We don't want any unknown or "other" streams on our endpoint,
|
|
* so always just say we've reached the limit
|
|
*/
|
|
return 1;
|
|
}
|
|
}
|
|
|
|
static int get_mid_bundle_group(const pjmedia_sdp_session *sdp, const char *mid)
|
|
{
|
|
int bundle_group = 0;
|
|
int index;
|
|
|
|
for (index = 0; index < sdp->attr_count; ++index) {
|
|
pjmedia_sdp_attr *attr = sdp->attr[index];
|
|
char value[pj_strlen(&attr->value) + 1], *mids = value, *attr_mid;
|
|
|
|
if (pj_strcmp2(&attr->name, "group") || pj_strncmp2(&attr->value, "BUNDLE", 6)) {
|
|
continue;
|
|
}
|
|
|
|
ast_copy_pj_str(value, &attr->value, sizeof(value));
|
|
|
|
/* Skip the BUNDLE at the front */
|
|
mids += 7;
|
|
|
|
while ((attr_mid = strsep(&mids, " "))) {
|
|
if (!strcmp(attr_mid, mid)) {
|
|
/* The ordering of attributes determines our internal identification of the bundle group based on number,
|
|
* with -1 being not in a bundle group. Since this is only exposed internally for response purposes it's
|
|
* actually even fine if things move around.
|
|
*/
|
|
return bundle_group;
|
|
}
|
|
}
|
|
|
|
bundle_group++;
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
static int set_mid_and_bundle_group(struct ast_sip_session *session,
|
|
struct ast_sip_session_media *session_media,
|
|
const pjmedia_sdp_session *sdp,
|
|
const struct pjmedia_sdp_media *stream)
|
|
{
|
|
pjmedia_sdp_attr *attr;
|
|
|
|
if (!session->endpoint->media.bundle) {
|
|
return 0;
|
|
}
|
|
|
|
/* By default on an incoming negotiation we assume no mid and bundle group is present */
|
|
ast_free(session_media->mid);
|
|
session_media->mid = NULL;
|
|
session_media->bundle_group = -1;
|
|
session_media->bundled = 0;
|
|
|
|
/* Grab the media identifier for the stream */
|
|
attr = pjmedia_sdp_media_find_attr2(stream, "mid", NULL);
|
|
if (!attr) {
|
|
return 0;
|
|
}
|
|
|
|
session_media->mid = ast_calloc(1, attr->value.slen + 1);
|
|
if (!session_media->mid) {
|
|
return 0;
|
|
}
|
|
ast_copy_pj_str(session_media->mid, &attr->value, attr->value.slen + 1);
|
|
|
|
/* Determine what bundle group this is part of */
|
|
session_media->bundle_group = get_mid_bundle_group(sdp, session_media->mid);
|
|
|
|
/* If this is actually part of a bundle group then the other side requested or accepted the bundle request */
|
|
session_media->bundled = session_media->bundle_group != -1;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void set_remote_mslabel_and_stream_group(struct ast_sip_session *session,
|
|
struct ast_sip_session_media *session_media,
|
|
const pjmedia_sdp_session *sdp,
|
|
const struct pjmedia_sdp_media *stream,
|
|
struct ast_stream *asterisk_stream)
|
|
{
|
|
int index;
|
|
|
|
ast_free(session_media->remote_mslabel);
|
|
session_media->remote_mslabel = NULL;
|
|
|
|
for (index = 0; index < stream->attr_count; ++index) {
|
|
pjmedia_sdp_attr *attr = stream->attr[index];
|
|
char attr_value[pj_strlen(&attr->value) + 1];
|
|
char *ssrc_attribute_name, *ssrc_attribute_value = NULL;
|
|
char *msid, *tmp = attr_value;
|
|
static const pj_str_t STR_msid = { "msid", 4 };
|
|
static const pj_str_t STR_ssrc = { "ssrc", 4 };
|
|
|
|
if (!pj_strcmp(&attr->name, &STR_msid)) {
|
|
ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
|
|
msid = strsep(&tmp, " ");
|
|
session_media->remote_mslabel = ast_strdup(msid);
|
|
break;
|
|
} else if (!pj_strcmp(&attr->name, &STR_ssrc)) {
|
|
ast_copy_pj_str(attr_value, &attr->value, sizeof(attr_value));
|
|
|
|
if ((ssrc_attribute_name = strchr(attr_value, ' '))) {
|
|
/* This has an actual attribute */
|
|
*ssrc_attribute_name++ = '\0';
|
|
ssrc_attribute_value = strchr(ssrc_attribute_name, ':');
|
|
if (ssrc_attribute_value) {
|
|
/* Values are actually optional according to the spec */
|
|
*ssrc_attribute_value++ = '\0';
|
|
}
|
|
|
|
if (!strcasecmp(ssrc_attribute_name, "mslabel") && !ast_strlen_zero(ssrc_attribute_value)) {
|
|
session_media->remote_mslabel = ast_strdup(ssrc_attribute_value);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (ast_strlen_zero(session_media->remote_mslabel)) {
|
|
return;
|
|
}
|
|
|
|
/* Iterate through the existing streams looking for a match and if so then group this with it */
|
|
for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
|
|
struct ast_sip_session_media *group_session_media;
|
|
|
|
group_session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
|
|
|
|
if (ast_strlen_zero(group_session_media->remote_mslabel) ||
|
|
strcmp(group_session_media->remote_mslabel, session_media->remote_mslabel)) {
|
|
continue;
|
|
}
|
|
|
|
ast_stream_set_group(asterisk_stream, index);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void remove_stream_from_bundle(struct ast_sip_session_media *session_media,
|
|
struct ast_stream *stream)
|
|
{
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
|
|
ast_free(session_media->mid);
|
|
session_media->mid = NULL;
|
|
session_media->bundle_group = -1;
|
|
session_media->bundled = 0;
|
|
}
|
|
|
|
static int handle_incoming_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
|
|
{
|
|
int i;
|
|
int handled = 0;
|
|
int type_streams[AST_MEDIA_TYPE_END] = {0};
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (session->inv_session && session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Failed to handle incoming SDP. Session has been already disconnected\n");
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Already disconnected\n");
|
|
}
|
|
|
|
/* It is possible for SDP deferral to have already created a pending topology */
|
|
if (!session->pending_media_state->topology) {
|
|
session->pending_media_state->topology = ast_stream_topology_alloc();
|
|
if (!session->pending_media_state->topology) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "No topology\n");
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < sdp->media_count; ++i) {
|
|
/* See if there are registered handlers for this media stream type */
|
|
char media[20];
|
|
struct ast_sip_session_sdp_handler *handler;
|
|
RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
|
|
struct ast_sip_session_media *session_media = NULL;
|
|
int res;
|
|
enum ast_media_type type;
|
|
struct ast_stream *stream = NULL;
|
|
pjmedia_sdp_media *remote_stream = sdp->media[i];
|
|
|
|
/* We need a null-terminated version of the media string */
|
|
ast_copy_pj_str(media, &sdp->media[i]->desc.media, sizeof(media));
|
|
type = ast_media_type_from_str(media);
|
|
|
|
/* See if we have an already existing stream, which can occur from SDP deferral checking */
|
|
if (i < ast_stream_topology_get_count(session->pending_media_state->topology)) {
|
|
stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
|
|
}
|
|
if (!stream) {
|
|
struct ast_stream *existing_stream = NULL;
|
|
|
|
if (session->active_media_state->topology &&
|
|
(i < ast_stream_topology_get_count(session->active_media_state->topology))) {
|
|
existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, i);
|
|
}
|
|
|
|
stream = ast_stream_alloc(existing_stream ? ast_stream_get_name(existing_stream) : ast_codec_media_type2str(type), type);
|
|
if (!stream) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create stream\n");
|
|
}
|
|
if (ast_stream_topology_set_stream(session->pending_media_state->topology, i, stream)) {
|
|
ast_stream_free(stream);
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't set stream\n");
|
|
}
|
|
if (existing_stream) {
|
|
const char *stream_label = ast_stream_get_metadata(existing_stream, "SDP:LABEL");
|
|
|
|
if (!ast_strlen_zero(stream_label)) {
|
|
ast_stream_set_metadata(stream, "SDP:LABEL", stream_label);
|
|
}
|
|
}
|
|
|
|
/* For backwards compatibility with the core the default audio stream is always sendrecv */
|
|
if (!ast_sip_session_is_pending_stream_default(session, stream) || strcmp(media, "audio")) {
|
|
if (pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
|
|
/* Stream state reflects our state of a stream, so in the case of
|
|
* sendonly and recvonly we store the opposite since that is what ours
|
|
* is.
|
|
*/
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
|
|
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "recvonly", NULL)) {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
|
|
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_INACTIVE);
|
|
} else {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
|
|
}
|
|
} else {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
|
|
}
|
|
}
|
|
|
|
session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_media_type_from_str(media), i);
|
|
if (!session_media) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't add session media\n");
|
|
}
|
|
|
|
/* If this stream is already declined mark it as such, or mark it as such if we've reached the limit */
|
|
if (!remote_stream->desc.port || is_stream_limitation_reached(type, session->endpoint, type_streams)) {
|
|
ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
|
|
ast_codec_media_type2str(type), i);
|
|
remove_stream_from_bundle(session_media, stream);
|
|
continue;
|
|
}
|
|
|
|
set_mid_and_bundle_group(session, session_media, sdp, remote_stream);
|
|
set_remote_mslabel_and_stream_group(session, session_media, sdp, remote_stream, stream);
|
|
|
|
if (session_media->handler) {
|
|
handler = session_media->handler;
|
|
ast_debug(1, "Negotiating incoming SDP media stream '%s' using %s SDP handler\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
session_media->handler->id);
|
|
res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
|
|
if (res < 0) {
|
|
/* Catastrophic failure. Abort! */
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't negotiate incoming sdp stream\n");
|
|
} else if (res == 0) {
|
|
ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
|
|
ast_codec_media_type2str(type), i);
|
|
remove_stream_from_bundle(session_media, stream);
|
|
continue;
|
|
} else if (res > 0) {
|
|
ast_debug(1, "Media stream '%s' handled by %s\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
session_media->handler->id);
|
|
/* Handled by this handler. Move to the next stream */
|
|
handled = 1;
|
|
++type_streams[type];
|
|
continue;
|
|
}
|
|
}
|
|
|
|
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
|
|
if (!handler_list) {
|
|
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
|
|
continue;
|
|
}
|
|
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
|
|
if (handler == session_media->handler) {
|
|
continue;
|
|
}
|
|
ast_debug(1, "Negotiating incoming SDP media stream '%s' using %s SDP handler\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
res = handler->negotiate_incoming_sdp_stream(session, session_media, sdp, i, stream);
|
|
if (res < 0) {
|
|
/* Catastrophic failure. Abort! */
|
|
return -1;
|
|
} else if (res == 0) {
|
|
ast_debug(1, "Declining incoming SDP media stream '%s' at position '%d'\n",
|
|
ast_codec_media_type2str(type), i);
|
|
remove_stream_from_bundle(session_media, stream);
|
|
continue;
|
|
} else if (res > 0) {
|
|
ast_debug(1, "Media stream '%s' handled by %s\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
/* Handled by this handler. Move to the next stream */
|
|
session_media_set_handler(session_media, handler);
|
|
handled = 1;
|
|
++type_streams[type];
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
if (!handled) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Not handled\n");
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(0, "Handled. Active: %s Pending: %s\n",
|
|
ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP))
|
|
);
|
|
}
|
|
|
|
static int handle_negotiated_sdp_session_media(struct ast_sip_session_media *session_media,
|
|
struct ast_sip_session *session, const pjmedia_sdp_session *local,
|
|
const pjmedia_sdp_session *remote, int index, struct ast_stream *asterisk_stream)
|
|
{
|
|
/* See if there are registered handlers for this media stream type */
|
|
struct pjmedia_sdp_media *local_stream = local->media[index];
|
|
char media[20];
|
|
struct ast_sip_session_sdp_handler *handler;
|
|
RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
|
|
int res;
|
|
SCOPE_ENTER(1, "%s\n", session ? ast_sip_session_get_name(session) : "unknown");
|
|
|
|
/* We need a null-terminated version of the media string */
|
|
ast_copy_pj_str(media, &local->media[index]->desc.media, sizeof(media));
|
|
|
|
/* For backwards compatibility we only reflect the stream state correctly on
|
|
* the non-default streams and any non-audio streams. This is because the stream
|
|
* state of the default audio stream is also used for signaling that someone has
|
|
* placed us on hold. This situation is not handled currently and can result in
|
|
* the remote side being sorted of placed on hold too.
|
|
*/
|
|
if (!ast_sip_session_is_pending_stream_default(session, asterisk_stream) || strcmp(media, "audio")) {
|
|
/* Determine the state of the stream based on our local SDP */
|
|
if (pjmedia_sdp_media_find_attr2(local_stream, "sendonly", NULL)) {
|
|
ast_stream_set_state(asterisk_stream, AST_STREAM_STATE_SENDONLY);
|
|
} else if (pjmedia_sdp_media_find_attr2(local_stream, "recvonly", NULL)) {
|
|
ast_stream_set_state(asterisk_stream, AST_STREAM_STATE_RECVONLY);
|
|
} else if (pjmedia_sdp_media_find_attr2(local_stream, "inactive", NULL)) {
|
|
ast_stream_set_state(asterisk_stream, AST_STREAM_STATE_INACTIVE);
|
|
} else {
|
|
ast_stream_set_state(asterisk_stream, AST_STREAM_STATE_SENDRECV);
|
|
}
|
|
} else {
|
|
ast_stream_set_state(asterisk_stream, AST_STREAM_STATE_SENDRECV);
|
|
}
|
|
|
|
set_mid_and_bundle_group(session, session_media, remote, remote->media[index]);
|
|
set_remote_mslabel_and_stream_group(session, session_media, remote, remote->media[index], asterisk_stream);
|
|
|
|
handler = session_media->handler;
|
|
if (handler) {
|
|
ast_debug(1, "Applying negotiated SDP media stream '%s' using %s SDP handler\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
res = handler->apply_negotiated_sdp_stream(session, session_media, local, remote, index, asterisk_stream);
|
|
if (res >= 0) {
|
|
ast_debug(1, "Applied negotiated SDP media stream '%s' using %s SDP handler\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: Applied negotiated SDP media stream '%s' using %s SDP handler\n",
|
|
ast_sip_session_get_name(session), ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(-1, "%s: Failed to apply negotiated SDP media stream '%s' using %s SDP handler\n",
|
|
ast_sip_session_get_name(session), ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
}
|
|
|
|
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
|
|
if (!handler_list) {
|
|
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
|
|
SCOPE_EXIT_RTN_VALUE(-1, "%s: No registered SDP handlers for media type '%s'\n",
|
|
ast_sip_session_get_name(session), media);
|
|
}
|
|
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
|
|
if (handler == session_media->handler) {
|
|
continue;
|
|
}
|
|
ast_debug(1, "Applying negotiated SDP media stream '%s' using %s SDP handler\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
res = handler->apply_negotiated_sdp_stream(session, session_media, local, remote, index, asterisk_stream);
|
|
if (res < 0) {
|
|
/* Catastrophic failure. Abort! */
|
|
SCOPE_EXIT_RTN_VALUE(-1, "%s: Handler '%s' returned %d\n",
|
|
ast_sip_session_get_name(session), handler->id, res);
|
|
}
|
|
if (res > 0) {
|
|
ast_debug(1, "Applied negotiated SDP media stream '%s' using %s SDP handler\n",
|
|
ast_codec_media_type2str(session_media->type),
|
|
handler->id);
|
|
/* Handled by this handler. Move to the next stream */
|
|
session_media_set_handler(session_media, handler);
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: Handler '%s' handled this sdp stream\n",
|
|
ast_sip_session_get_name(session), handler->id);
|
|
}
|
|
}
|
|
|
|
res = 0;
|
|
if (session_media->handler && session_media->handler->stream_stop) {
|
|
res = 1;
|
|
ast_debug(1, "Stopping SDP media stream '%s' as it is not currently negotiated\n",
|
|
ast_codec_media_type2str(session_media->type));
|
|
session_media->handler->stream_stop(session_media);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0, "%s: Media type '%s' %s\n",
|
|
ast_sip_session_get_name(session), ast_codec_media_type2str(session_media->type),
|
|
res ? "not negotiated. Stopped" : "handled");
|
|
}
|
|
|
|
static int handle_negotiated_sdp(struct ast_sip_session *session, const pjmedia_sdp_session *local, const pjmedia_sdp_session *remote)
|
|
{
|
|
int i;
|
|
struct ast_stream_topology *topology;
|
|
unsigned int changed = 0; /* 0 = unchanged, 1 = new source, 2 = new topology */
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (!session->pending_media_state->topology) {
|
|
if (session->active_media_state->topology) {
|
|
/*
|
|
* This happens when we have negotiated media after receiving a 183,
|
|
* and we're now receiving a 200 with a new SDP. In this case, there
|
|
* is active_media_state, but the pending_media_state has been reset.
|
|
*/
|
|
struct ast_sip_session_media_state *active_media_state_clone;
|
|
|
|
active_media_state_clone =
|
|
ast_sip_session_media_state_clone(session->active_media_state);
|
|
if (!active_media_state_clone) {
|
|
ast_log(LOG_WARNING, "Unable to clone active media state for channel '%s'\n",
|
|
session->channel ? ast_channel_name(session->channel) : "unknown");
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Unable to clone active media state\n");
|
|
}
|
|
|
|
ast_sip_session_media_state_free(session->pending_media_state);
|
|
session->pending_media_state = active_media_state_clone;
|
|
} else {
|
|
ast_log(LOG_WARNING, "No pending or active media state for channel '%s'\n",
|
|
session->channel ? ast_channel_name(session->channel) : "unknown");
|
|
SCOPE_EXIT_RTN_VALUE(-1, "No media state\n");
|
|
}
|
|
}
|
|
|
|
/* If we're handling negotiated streams, then we should already have set
|
|
* up session media instances (and Asterisk streams) that correspond to
|
|
* the local SDP, and there should be the same number of session medias
|
|
* and streams as there are local SDP streams
|
|
*/
|
|
if (ast_stream_topology_get_count(session->pending_media_state->topology) != local->media_count
|
|
|| AST_VECTOR_SIZE(&session->pending_media_state->sessions) != local->media_count) {
|
|
ast_log(LOG_WARNING, "Local SDP for channel '%s' contains %d media streams while we expected it to contain %u\n",
|
|
session->channel ? ast_channel_name(session->channel) : "unknown",
|
|
ast_stream_topology_get_count(session->pending_media_state->topology), local->media_count);
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Media stream count mismatch\n");
|
|
}
|
|
|
|
for (i = 0; i < local->media_count; ++i) {
|
|
struct ast_sip_session_media *session_media;
|
|
struct ast_stream *stream;
|
|
|
|
if (!remote->media[i]) {
|
|
continue;
|
|
}
|
|
|
|
session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, i);
|
|
stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
|
|
|
|
/* Make sure that this stream is in the correct state. If we need to change
|
|
* the state to REMOVED, then our work here is done, so go ahead and move on
|
|
* to the next stream.
|
|
*/
|
|
if (!remote->media[i]->desc.port) {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
|
|
continue;
|
|
}
|
|
|
|
/* If the stream state is REMOVED, nothing needs to be done, so move on to the
|
|
* next stream. This can occur if an internal thing has requested it to be
|
|
* removed, or if we remove it as a result of the stream limit being reached.
|
|
*/
|
|
if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
|
|
/*
|
|
* Defer removing the handler until we are ready to activate
|
|
* the new topology. The channel's thread may still be using
|
|
* the stream and we could crash before we are ready.
|
|
*/
|
|
continue;
|
|
}
|
|
|
|
if (handle_negotiated_sdp_session_media(session_media, session, local, remote, i, stream)) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Unable to handle negotiated session media\n");
|
|
}
|
|
|
|
changed |= session_media->changed;
|
|
session_media->changed = 0;
|
|
}
|
|
|
|
/* Apply the pending media state to the channel and make it active */
|
|
ast_channel_lock(session->channel);
|
|
|
|
/* Now update the stream handler for any declined/removed streams */
|
|
for (i = 0; i < local->media_count; ++i) {
|
|
struct ast_sip_session_media *session_media;
|
|
struct ast_stream *stream;
|
|
|
|
if (!remote->media[i]) {
|
|
continue;
|
|
}
|
|
|
|
session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, i);
|
|
stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
|
|
|
|
if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED
|
|
&& session_media->handler) {
|
|
/*
|
|
* This stream is no longer being used and the channel's thread
|
|
* is held off because we have the channel lock so release any
|
|
* resources the handler may have on it.
|
|
*/
|
|
session_media_set_handler(session_media, NULL);
|
|
}
|
|
}
|
|
|
|
/* Update the topology on the channel to match the accepted one */
|
|
topology = ast_stream_topology_clone(session->pending_media_state->topology);
|
|
if (topology) {
|
|
ast_channel_set_stream_topology(session->channel, topology);
|
|
/* If this is a remotely done renegotiation that has changed the stream topology notify what is
|
|
* currently handling this channel. Note that fax uses its own process, so if we are transitioning
|
|
* between audio and fax or vice versa we don't notify.
|
|
*/
|
|
if (pjmedia_sdp_neg_was_answer_remote(session->inv_session->neg) == PJ_FALSE &&
|
|
session->active_media_state && session->active_media_state->topology &&
|
|
!ast_stream_topology_equal(session->active_media_state->topology, topology) &&
|
|
!session->active_media_state->default_session[AST_MEDIA_TYPE_IMAGE] &&
|
|
!session->pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE]) {
|
|
changed = 2;
|
|
}
|
|
}
|
|
|
|
/* Remove all current file descriptors from the channel */
|
|
for (i = 0; i < AST_VECTOR_SIZE(&session->active_media_state->read_callbacks); ++i) {
|
|
ast_channel_internal_fd_clear(session->channel, i + AST_EXTENDED_FDS);
|
|
}
|
|
|
|
/* Add all the file descriptors from the pending media state */
|
|
for (i = 0; i < AST_VECTOR_SIZE(&session->pending_media_state->read_callbacks); ++i) {
|
|
struct ast_sip_session_media_read_callback_state *callback_state;
|
|
|
|
callback_state = AST_VECTOR_GET_ADDR(&session->pending_media_state->read_callbacks, i);
|
|
ast_channel_internal_fd_set(session->channel, i + AST_EXTENDED_FDS, callback_state->fd);
|
|
}
|
|
|
|
/* Active and pending flip flop as needed */
|
|
ast_sip_session_media_stats_save(session, session->active_media_state);
|
|
SWAP(session->active_media_state, session->pending_media_state);
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
|
|
ast_channel_unlock(session->channel);
|
|
|
|
if (changed == 1) {
|
|
struct ast_frame f = { AST_FRAME_CONTROL, .subclass.integer = AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED };
|
|
|
|
ast_queue_frame(session->channel, &f);
|
|
} else if (changed == 2) {
|
|
ast_channel_stream_topology_changed_externally(session->channel);
|
|
} else {
|
|
ast_queue_frame(session->channel, &ast_null_frame);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
#define DATASTORE_BUCKETS 53
|
|
#define MEDIA_BUCKETS 7
|
|
|
|
static void session_datastore_destroy(void *obj)
|
|
{
|
|
struct ast_datastore *datastore = obj;
|
|
|
|
/* Using the destroy function (if present) destroy the data */
|
|
if (datastore->info->destroy != NULL && datastore->data != NULL) {
|
|
datastore->info->destroy(datastore->data);
|
|
datastore->data = NULL;
|
|
}
|
|
|
|
ast_free((void *) datastore->uid);
|
|
datastore->uid = NULL;
|
|
}
|
|
|
|
struct ast_datastore *ast_sip_session_alloc_datastore(const struct ast_datastore_info *info, const char *uid)
|
|
{
|
|
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
|
|
char uuid_buf[AST_UUID_STR_LEN];
|
|
const char *uid_ptr = uid;
|
|
|
|
if (!info) {
|
|
return NULL;
|
|
}
|
|
|
|
datastore = ao2_alloc(sizeof(*datastore), session_datastore_destroy);
|
|
if (!datastore) {
|
|
return NULL;
|
|
}
|
|
|
|
datastore->info = info;
|
|
if (ast_strlen_zero(uid)) {
|
|
/* They didn't provide an ID so we'll provide one ourself */
|
|
uid_ptr = ast_uuid_generate_str(uuid_buf, sizeof(uuid_buf));
|
|
}
|
|
|
|
datastore->uid = ast_strdup(uid_ptr);
|
|
if (!datastore->uid) {
|
|
return NULL;
|
|
}
|
|
|
|
ao2_ref(datastore, +1);
|
|
return datastore;
|
|
}
|
|
|
|
int ast_sip_session_add_datastore(struct ast_sip_session *session, struct ast_datastore *datastore)
|
|
{
|
|
ast_assert(datastore != NULL);
|
|
ast_assert(datastore->info != NULL);
|
|
ast_assert(ast_strlen_zero(datastore->uid) == 0);
|
|
|
|
if (!ao2_link(session->datastores, datastore)) {
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
struct ast_datastore *ast_sip_session_get_datastore(struct ast_sip_session *session, const char *name)
|
|
{
|
|
return ao2_find(session->datastores, name, OBJ_KEY);
|
|
}
|
|
|
|
void ast_sip_session_remove_datastore(struct ast_sip_session *session, const char *name)
|
|
{
|
|
ao2_callback(session->datastores, OBJ_KEY | OBJ_UNLINK | OBJ_NODATA, NULL, (void *) name);
|
|
}
|
|
|
|
enum delayed_method {
|
|
DELAYED_METHOD_INVITE,
|
|
DELAYED_METHOD_UPDATE,
|
|
DELAYED_METHOD_BYE,
|
|
};
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Convert delayed method enum value to a string.
|
|
* \since 13.3.0
|
|
*
|
|
* \param method Delayed method enum value to convert to a string.
|
|
*
|
|
* \return String value of delayed method.
|
|
*/
|
|
static const char *delayed_method2str(enum delayed_method method)
|
|
{
|
|
const char *str = "<unknown>";
|
|
|
|
switch (method) {
|
|
case DELAYED_METHOD_INVITE:
|
|
str = "INVITE";
|
|
break;
|
|
case DELAYED_METHOD_UPDATE:
|
|
str = "UPDATE";
|
|
break;
|
|
case DELAYED_METHOD_BYE:
|
|
str = "BYE";
|
|
break;
|
|
}
|
|
|
|
return str;
|
|
}
|
|
|
|
/*!
|
|
* \brief Structure used for sending delayed requests
|
|
*
|
|
* Requests are typically delayed because the current transaction
|
|
* state of an INVITE. Once the pending INVITE transaction terminates,
|
|
* the delayed request will be sent
|
|
*/
|
|
struct ast_sip_session_delayed_request {
|
|
/*! Method of the request */
|
|
enum delayed_method method;
|
|
/*! Callback to call when the delayed request is created. */
|
|
ast_sip_session_request_creation_cb on_request_creation;
|
|
/*! Callback to call when the delayed request SDP is created */
|
|
ast_sip_session_sdp_creation_cb on_sdp_creation;
|
|
/*! Callback to call when the delayed request receives a response */
|
|
ast_sip_session_response_cb on_response;
|
|
/*! Whether to generate new SDP */
|
|
int generate_new_sdp;
|
|
/*! Requested media state for the SDP */
|
|
struct ast_sip_session_media_state *media_state;
|
|
AST_LIST_ENTRY(ast_sip_session_delayed_request) next;
|
|
};
|
|
|
|
static struct ast_sip_session_delayed_request *delayed_request_alloc(
|
|
enum delayed_method method,
|
|
ast_sip_session_request_creation_cb on_request_creation,
|
|
ast_sip_session_sdp_creation_cb on_sdp_creation,
|
|
ast_sip_session_response_cb on_response,
|
|
int generate_new_sdp,
|
|
struct ast_sip_session_media_state *media_state)
|
|
{
|
|
struct ast_sip_session_delayed_request *delay = ast_calloc(1, sizeof(*delay));
|
|
|
|
if (!delay) {
|
|
return NULL;
|
|
}
|
|
delay->method = method;
|
|
delay->on_request_creation = on_request_creation;
|
|
delay->on_sdp_creation = on_sdp_creation;
|
|
delay->on_response = on_response;
|
|
delay->generate_new_sdp = generate_new_sdp;
|
|
delay->media_state = media_state;
|
|
return delay;
|
|
}
|
|
|
|
static void delayed_request_free(struct ast_sip_session_delayed_request *delay)
|
|
{
|
|
ast_sip_session_media_state_free(delay->media_state);
|
|
ast_free(delay);
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Send a delayed request
|
|
*
|
|
* \retval -1 failure
|
|
* \retval 0 success
|
|
* \retval 1 refresh request not sent as no change would occur
|
|
*/
|
|
static int send_delayed_request(struct ast_sip_session *session, struct ast_sip_session_delayed_request *delay)
|
|
{
|
|
int res;
|
|
|
|
ast_debug(3, "Endpoint '%s(%s)' sending delayed %s request.\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
delayed_method2str(delay->method));
|
|
|
|
switch (delay->method) {
|
|
case DELAYED_METHOD_INVITE:
|
|
res = sip_session_refresh(session, delay->on_request_creation,
|
|
delay->on_sdp_creation, delay->on_response,
|
|
AST_SIP_SESSION_REFRESH_METHOD_INVITE, delay->generate_new_sdp, delay->media_state, 1);
|
|
/* Ownership of media state transitions to ast_sip_session_refresh */
|
|
delay->media_state = NULL;
|
|
return res;
|
|
case DELAYED_METHOD_UPDATE:
|
|
res = sip_session_refresh(session, delay->on_request_creation,
|
|
delay->on_sdp_creation, delay->on_response,
|
|
AST_SIP_SESSION_REFRESH_METHOD_UPDATE, delay->generate_new_sdp, delay->media_state, 1);
|
|
/* Ownership of media state transitions to ast_sip_session_refresh */
|
|
delay->media_state = NULL;
|
|
return res;
|
|
case DELAYED_METHOD_BYE:
|
|
ast_sip_session_terminate(session, 0);
|
|
return 0;
|
|
}
|
|
ast_log(LOG_WARNING, "Don't know how to send delayed %s(%d) request.\n",
|
|
delayed_method2str(delay->method), delay->method);
|
|
return -1;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief The current INVITE transaction is in the PROCEEDING state.
|
|
* \since 13.3.0
|
|
*
|
|
* \param vsession Session object.
|
|
*
|
|
* \retval 0 on success.
|
|
* \retval -1 on error.
|
|
*/
|
|
static int invite_proceeding(void *vsession)
|
|
{
|
|
struct ast_sip_session *session = vsession;
|
|
struct ast_sip_session_delayed_request *delay;
|
|
int found = 0;
|
|
int res = 0;
|
|
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&session->delayed_requests, delay, next) {
|
|
switch (delay->method) {
|
|
case DELAYED_METHOD_INVITE:
|
|
break;
|
|
case DELAYED_METHOD_UPDATE:
|
|
AST_LIST_REMOVE_CURRENT(next);
|
|
res = send_delayed_request(session, delay);
|
|
delayed_request_free(delay);
|
|
if (!res) {
|
|
found = 1;
|
|
}
|
|
break;
|
|
case DELAYED_METHOD_BYE:
|
|
/* A BYE is pending so don't bother anymore. */
|
|
found = 1;
|
|
break;
|
|
}
|
|
if (found) {
|
|
break;
|
|
}
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
|
|
|
ao2_ref(session, -1);
|
|
return res;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief The current INVITE transaction is in the TERMINATED state.
|
|
* \since 13.3.0
|
|
*
|
|
* \param vsession Session object.
|
|
*
|
|
* \retval 0 on success.
|
|
* \retval -1 on error.
|
|
*/
|
|
static int invite_terminated(void *vsession)
|
|
{
|
|
struct ast_sip_session *session = vsession;
|
|
struct ast_sip_session_delayed_request *delay;
|
|
int found = 0;
|
|
int res = 0;
|
|
int timer_running;
|
|
|
|
/* re-INVITE collision timer running? */
|
|
timer_running = pj_timer_entry_running(&session->rescheduled_reinvite);
|
|
|
|
AST_LIST_TRAVERSE_SAFE_BEGIN(&session->delayed_requests, delay, next) {
|
|
switch (delay->method) {
|
|
case DELAYED_METHOD_INVITE:
|
|
if (!timer_running) {
|
|
found = 1;
|
|
}
|
|
break;
|
|
case DELAYED_METHOD_UPDATE:
|
|
case DELAYED_METHOD_BYE:
|
|
found = 1;
|
|
break;
|
|
}
|
|
if (found) {
|
|
AST_LIST_REMOVE_CURRENT(next);
|
|
res = send_delayed_request(session, delay);
|
|
delayed_request_free(delay);
|
|
if (!res) {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
AST_LIST_TRAVERSE_SAFE_END;
|
|
|
|
ao2_ref(session, -1);
|
|
return res;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief INVITE collision timeout.
|
|
* \since 13.3.0
|
|
*
|
|
* \param vsession Session object.
|
|
*
|
|
* \retval 0 on success.
|
|
* \retval -1 on error.
|
|
*/
|
|
static int invite_collision_timeout(void *vsession)
|
|
{
|
|
struct ast_sip_session *session = vsession;
|
|
int res;
|
|
|
|
if (session->inv_session->invite_tsx) {
|
|
/*
|
|
* INVITE transaction still active. Let it send
|
|
* the collision re-INVITE when it terminates.
|
|
*/
|
|
ao2_ref(session, -1);
|
|
res = 0;
|
|
} else {
|
|
res = invite_terminated(session);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief The current UPDATE transaction is in the COMPLETED state.
|
|
* \since 13.3.0
|
|
*
|
|
* \param vsession Session object.
|
|
*
|
|
* \retval 0 on success.
|
|
* \retval -1 on error.
|
|
*/
|
|
static int update_completed(void *vsession)
|
|
{
|
|
struct ast_sip_session *session = vsession;
|
|
int res;
|
|
|
|
if (session->inv_session->invite_tsx) {
|
|
res = invite_proceeding(session);
|
|
} else {
|
|
res = invite_terminated(session);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static void check_delayed_requests(struct ast_sip_session *session,
|
|
int (*cb)(void *vsession))
|
|
{
|
|
ao2_ref(session, +1);
|
|
if (ast_sip_push_task(session->serializer, cb, session)) {
|
|
ao2_ref(session, -1);
|
|
}
|
|
}
|
|
|
|
static int delay_request(struct ast_sip_session *session,
|
|
ast_sip_session_request_creation_cb on_request,
|
|
ast_sip_session_sdp_creation_cb on_sdp_creation,
|
|
ast_sip_session_response_cb on_response,
|
|
int generate_new_sdp,
|
|
enum delayed_method method,
|
|
struct ast_sip_session_media_state *media_state)
|
|
{
|
|
struct ast_sip_session_delayed_request *delay = delayed_request_alloc(method,
|
|
on_request, on_sdp_creation, on_response, generate_new_sdp, media_state);
|
|
|
|
if (!delay) {
|
|
ast_sip_session_media_state_free(media_state);
|
|
return -1;
|
|
}
|
|
|
|
if (method == DELAYED_METHOD_BYE) {
|
|
/* Send BYE as early as possible */
|
|
AST_LIST_INSERT_HEAD(&session->delayed_requests, delay, next);
|
|
} else {
|
|
AST_LIST_INSERT_TAIL(&session->delayed_requests, delay, next);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static pjmedia_sdp_session *generate_session_refresh_sdp(struct ast_sip_session *session)
|
|
{
|
|
pjsip_inv_session *inv_session = session->inv_session;
|
|
const pjmedia_sdp_session *previous_sdp = NULL;
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (inv_session->neg) {
|
|
if (pjmedia_sdp_neg_was_answer_remote(inv_session->neg)) {
|
|
pjmedia_sdp_neg_get_active_remote(inv_session->neg, &previous_sdp);
|
|
} else {
|
|
pjmedia_sdp_neg_get_active_local(inv_session->neg, &previous_sdp);
|
|
}
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(create_local_sdp(inv_session, session, previous_sdp));
|
|
}
|
|
|
|
static void set_from_header(struct ast_sip_session *session)
|
|
{
|
|
struct ast_party_id effective_id;
|
|
struct ast_party_id connected_id;
|
|
pj_pool_t *dlg_pool;
|
|
pjsip_fromto_hdr *dlg_info;
|
|
pjsip_contact_hdr *dlg_contact;
|
|
pjsip_name_addr *dlg_info_name_addr;
|
|
pjsip_sip_uri *dlg_info_uri;
|
|
pjsip_sip_uri *dlg_contact_uri;
|
|
int restricted;
|
|
const char *pjsip_from_domain;
|
|
|
|
if (!session->channel || session->saved_from_hdr) {
|
|
return;
|
|
}
|
|
|
|
/* We need to save off connected_id for RPID/PAI generation */
|
|
ast_party_id_init(&connected_id);
|
|
ast_channel_lock(session->channel);
|
|
effective_id = ast_channel_connected_effective_id(session->channel);
|
|
ast_party_id_copy(&connected_id, &effective_id);
|
|
ast_channel_unlock(session->channel);
|
|
|
|
restricted =
|
|
((ast_party_id_presentation(&connected_id) & AST_PRES_RESTRICTION) != AST_PRES_ALLOWED);
|
|
|
|
/* Now set up dlg->local.info so pjsip can correctly generate From */
|
|
|
|
dlg_pool = session->inv_session->dlg->pool;
|
|
dlg_info = session->inv_session->dlg->local.info;
|
|
dlg_contact = session->inv_session->dlg->local.contact;
|
|
dlg_info_name_addr = (pjsip_name_addr *) dlg_info->uri;
|
|
dlg_info_uri = pjsip_uri_get_uri(dlg_info_name_addr);
|
|
dlg_contact_uri = (pjsip_sip_uri*)pjsip_uri_get_uri(dlg_contact->uri);
|
|
|
|
if (session->endpoint->id.trust_outbound || !restricted) {
|
|
ast_sip_modify_id_header(dlg_pool, dlg_info, &connected_id);
|
|
if (ast_sip_get_use_callerid_contact() && ast_strlen_zero(session->endpoint->contact_user)) {
|
|
pj_strdup2(dlg_pool, &dlg_contact_uri->user, S_COR(connected_id.number.valid, connected_id.number.str, ""));
|
|
}
|
|
}
|
|
|
|
ast_party_id_free(&connected_id);
|
|
|
|
if (!ast_strlen_zero(session->endpoint->fromuser)) {
|
|
dlg_info_name_addr->display.ptr = NULL;
|
|
dlg_info_name_addr->display.slen = 0;
|
|
pj_strdup2(dlg_pool, &dlg_info_uri->user, session->endpoint->fromuser);
|
|
}
|
|
|
|
if (!ast_strlen_zero(session->endpoint->fromdomain)) {
|
|
pj_strdup2(dlg_pool, &dlg_info_uri->host, session->endpoint->fromdomain);
|
|
}
|
|
|
|
/*
|
|
* Channel variable for compatibility with chan_sip SIPFROMDOMAIN
|
|
*/
|
|
ast_channel_lock(session->channel);
|
|
pjsip_from_domain = pbx_builtin_getvar_helper(session->channel, "SIPFROMDOMAIN");
|
|
if (!ast_strlen_zero(pjsip_from_domain)) {
|
|
ast_debug(3, "From header domain reset by channel variable SIPFROMDOMAIN (%s)\n", pjsip_from_domain);
|
|
pj_strdup2(dlg_pool, &dlg_info_uri->host, pjsip_from_domain);
|
|
}
|
|
ast_channel_unlock(session->channel);
|
|
|
|
/* We need to save off the non-anonymized From for RPID/PAI generation (for domain) */
|
|
session->saved_from_hdr = pjsip_hdr_clone(dlg_pool, dlg_info);
|
|
ast_sip_add_usereqphone(session->endpoint, dlg_pool, session->saved_from_hdr->uri);
|
|
|
|
/* In chan_sip, fromuser and fromdomain trump restricted so we only
|
|
* anonymize if they're not set.
|
|
*/
|
|
if (restricted) {
|
|
/* fromuser doesn't provide a display name so we always set it */
|
|
pj_strdup2(dlg_pool, &dlg_info_name_addr->display, "Anonymous");
|
|
|
|
if (ast_strlen_zero(session->endpoint->fromuser)) {
|
|
pj_strdup2(dlg_pool, &dlg_info_uri->user, "anonymous");
|
|
}
|
|
|
|
if (ast_sip_get_use_callerid_contact() && ast_strlen_zero(session->endpoint->contact_user)) {
|
|
pj_strdup2(dlg_pool, &dlg_contact_uri->user, "anonymous");
|
|
}
|
|
|
|
if (ast_strlen_zero(session->endpoint->fromdomain)) {
|
|
pj_strdup2(dlg_pool, &dlg_info_uri->host, "anonymous.invalid");
|
|
}
|
|
} else {
|
|
ast_sip_add_usereqphone(session->endpoint, dlg_pool, dlg_info->uri);
|
|
}
|
|
}
|
|
|
|
static int sip_session_refresh(struct ast_sip_session *session,
|
|
ast_sip_session_request_creation_cb on_request_creation,
|
|
ast_sip_session_sdp_creation_cb on_sdp_creation,
|
|
ast_sip_session_response_cb on_response,
|
|
enum ast_sip_session_refresh_method method, int generate_new_sdp,
|
|
struct ast_sip_session_media_state *media_state,
|
|
int queued)
|
|
{
|
|
pjsip_inv_session *inv_session = session->inv_session;
|
|
pjmedia_sdp_session *new_sdp = NULL;
|
|
pjsip_tx_data *tdata;
|
|
|
|
if (media_state && (!media_state->topology || !generate_new_sdp)) {
|
|
ast_sip_session_media_state_free(media_state);
|
|
return -1;
|
|
}
|
|
|
|
if (inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
/* Don't try to do anything with a hung-up call */
|
|
ast_debug(3, "Not sending reinvite to %s because of disconnected state...\n",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
ast_sip_session_media_state_free(media_state);
|
|
return 0;
|
|
}
|
|
|
|
/* If the dialog has not yet been established we have to defer until it has */
|
|
if (inv_session->dlg->state != PJSIP_DIALOG_STATE_ESTABLISHED) {
|
|
ast_debug(3, "Delay sending request to %s because dialog has not been established...\n",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
return delay_request(session, on_request_creation, on_sdp_creation, on_response,
|
|
generate_new_sdp,
|
|
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE
|
|
? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE,
|
|
media_state);
|
|
}
|
|
|
|
if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
|
|
if (inv_session->invite_tsx) {
|
|
/* We can't send a reinvite yet, so delay it */
|
|
ast_debug(3, "Delay sending reinvite to %s because of outstanding transaction...\n",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
return delay_request(session, on_request_creation, on_sdp_creation,
|
|
on_response, generate_new_sdp, DELAYED_METHOD_INVITE, media_state);
|
|
} else if (inv_session->state != PJSIP_INV_STATE_CONFIRMED) {
|
|
/* Initial INVITE transaction failed to progress us to a confirmed state
|
|
* which means re-invites are not possible
|
|
*/
|
|
ast_debug(3, "Not sending reinvite to %s because not in confirmed state...\n",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
ast_sip_session_media_state_free(media_state);
|
|
return 0;
|
|
}
|
|
}
|
|
|
|
if (generate_new_sdp) {
|
|
/* SDP can only be generated if current negotiation has already completed */
|
|
if (inv_session->neg
|
|
&& pjmedia_sdp_neg_get_state(inv_session->neg)
|
|
!= PJMEDIA_SDP_NEG_STATE_DONE) {
|
|
ast_debug(3, "Delay session refresh with new SDP to %s because SDP negotiation is not yet done...\n",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
return delay_request(session, on_request_creation, on_sdp_creation,
|
|
on_response, generate_new_sdp,
|
|
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE
|
|
? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE, media_state);
|
|
}
|
|
|
|
/* If an explicitly requested media state has been provided use it instead of any pending one */
|
|
if (media_state) {
|
|
int index;
|
|
int type_streams[AST_MEDIA_TYPE_END] = {0};
|
|
struct ast_stream *stream;
|
|
|
|
/* Media state conveys a desired media state, so if there are outstanding
|
|
* delayed requests we need to ensure we go into the queue and not jump
|
|
* ahead. If we sent this media state now then updates could go out of
|
|
* order.
|
|
*/
|
|
if (!queued && !AST_LIST_EMPTY(&session->delayed_requests)) {
|
|
ast_debug(3, "Delay sending reinvite to %s because of outstanding requests...\n",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
return delay_request(session, on_request_creation, on_sdp_creation,
|
|
on_response, generate_new_sdp,
|
|
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE
|
|
? DELAYED_METHOD_INVITE : DELAYED_METHOD_UPDATE, media_state);
|
|
}
|
|
|
|
/* Prune the media state so the number of streams fit within the configured limits - we do it here
|
|
* so that the index of the resulting streams in the SDP match. If we simply left the streams out
|
|
* of the SDP when producing it we'd be in trouble. We also enforce formats here for media types that
|
|
* are configurable on the endpoint.
|
|
*/
|
|
for (index = 0; index < ast_stream_topology_get_count(media_state->topology); ++index) {
|
|
struct ast_stream *existing_stream = NULL;
|
|
|
|
stream = ast_stream_topology_get_stream(media_state->topology, index);
|
|
|
|
if (session->active_media_state->topology &&
|
|
index < ast_stream_topology_get_count(session->active_media_state->topology)) {
|
|
existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, index);
|
|
}
|
|
|
|
if (is_stream_limitation_reached(ast_stream_get_type(stream), session->endpoint, type_streams)) {
|
|
if (index < AST_VECTOR_SIZE(&media_state->sessions)) {
|
|
struct ast_sip_session_media *session_media = AST_VECTOR_GET(&media_state->sessions, index);
|
|
|
|
ao2_cleanup(session_media);
|
|
AST_VECTOR_REMOVE(&media_state->sessions, index, 1);
|
|
}
|
|
|
|
ast_stream_topology_del_stream(media_state->topology, index);
|
|
|
|
/* A stream has potentially moved into our spot so we need to jump back so we process it */
|
|
index -= 1;
|
|
continue;
|
|
}
|
|
|
|
/* No need to do anything with stream if it's media state is removed */
|
|
if (ast_stream_get_state(stream) == AST_STREAM_STATE_REMOVED) {
|
|
/* If there is no existing stream we can just not have this stream in the topology at all. */
|
|
if (!existing_stream) {
|
|
ast_stream_topology_del_stream(media_state->topology, index);
|
|
index -= 1;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
/* Enforce the configured allowed codecs on audio and video streams */
|
|
if ((ast_stream_get_type(stream) == AST_MEDIA_TYPE_AUDIO || ast_stream_get_type(stream) == AST_MEDIA_TYPE_VIDEO) &&
|
|
!ast_stream_get_metadata(stream, "pjsip_session_refresh")) {
|
|
struct ast_format_cap *joint_cap;
|
|
|
|
joint_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
|
|
if (!joint_cap) {
|
|
ast_sip_session_media_state_free(media_state);
|
|
return 0;
|
|
}
|
|
ast_format_cap_get_compatible(ast_stream_get_formats(stream), session->endpoint->media.codecs, joint_cap);
|
|
if (!ast_format_cap_count(joint_cap)) {
|
|
ao2_ref(joint_cap, -1);
|
|
|
|
if (!existing_stream) {
|
|
/* If there is no existing stream we can just not have this stream in the topology
|
|
* at all.
|
|
*/
|
|
ast_stream_topology_del_stream(media_state->topology, index);
|
|
index -= 1;
|
|
continue;
|
|
} else if (ast_stream_get_state(stream) != ast_stream_get_state(existing_stream) ||
|
|
strcmp(ast_stream_get_name(stream), ast_stream_get_name(existing_stream))) {
|
|
/* If the underlying stream is a different type or different name then we have to
|
|
* mark it as removed, as it is replacing an existing stream. We do this so order
|
|
* is preserved.
|
|
*/
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_REMOVED);
|
|
continue;
|
|
} else {
|
|
/* However if the stream is otherwise remaining the same we can keep the formats
|
|
* that exist on it already which allows media to continue to flow. We don't modify
|
|
* the format capabilities but do need to cast it so that ao2_bump can raise the
|
|
* reference count.
|
|
*/
|
|
joint_cap = ao2_bump((struct ast_format_cap *)ast_stream_get_formats(existing_stream));
|
|
}
|
|
}
|
|
ast_stream_set_formats(stream, joint_cap);
|
|
ao2_cleanup(joint_cap);
|
|
}
|
|
|
|
++type_streams[ast_stream_get_type(stream)];
|
|
}
|
|
|
|
if (session->active_media_state->topology) {
|
|
/* SDP is a fun thing. Take for example the fact that streams are never removed. They just become
|
|
* declined. To better handle this in the case where something requests a topology change for fewer
|
|
* streams than are currently present we fill in the topology to match the current number of streams
|
|
* that are active.
|
|
*/
|
|
for (index = ast_stream_topology_get_count(media_state->topology);
|
|
index < ast_stream_topology_get_count(session->active_media_state->topology); ++index) {
|
|
struct ast_stream *cloned;
|
|
|
|
stream = ast_stream_topology_get_stream(session->active_media_state->topology, index);
|
|
ast_assert(stream != NULL);
|
|
|
|
cloned = ast_stream_clone(stream, NULL);
|
|
if (!cloned) {
|
|
ast_sip_session_media_state_free(media_state);
|
|
return -1;
|
|
}
|
|
|
|
ast_stream_set_state(cloned, AST_STREAM_STATE_REMOVED);
|
|
if (ast_stream_topology_append_stream(media_state->topology, cloned) < 0) {
|
|
ast_stream_free(cloned);
|
|
ast_sip_session_media_state_free(media_state);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
/* If the resulting media state matches the existing active state don't bother doing a session refresh */
|
|
if (ast_stream_topology_equal(session->active_media_state->topology, media_state->topology)) {
|
|
ast_sip_session_media_state_free(media_state);
|
|
/* For external consumers we return 0 to say success, but internally for
|
|
* send_delayed_request we return a separate value to indicate that this
|
|
* session refresh would be redundant so we didn't send it
|
|
*/
|
|
return queued ? 1 : 0;
|
|
}
|
|
}
|
|
|
|
ast_sip_session_media_state_free(session->pending_media_state);
|
|
session->pending_media_state = media_state;
|
|
}
|
|
|
|
new_sdp = generate_session_refresh_sdp(session);
|
|
if (!new_sdp) {
|
|
ast_log(LOG_ERROR, "Failed to generate session refresh SDP. Not sending session refresh\n");
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
return -1;
|
|
}
|
|
if (on_sdp_creation) {
|
|
if (on_sdp_creation(session, new_sdp)) {
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (method == AST_SIP_SESSION_REFRESH_METHOD_INVITE) {
|
|
if (pjsip_inv_reinvite(inv_session, NULL, new_sdp, &tdata)) {
|
|
ast_log(LOG_WARNING, "Failed to create reinvite properly.\n");
|
|
if (generate_new_sdp) {
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
}
|
|
return -1;
|
|
}
|
|
} else if (pjsip_inv_update(inv_session, NULL, new_sdp, &tdata)) {
|
|
ast_log(LOG_WARNING, "Failed to create UPDATE properly.\n");
|
|
if (generate_new_sdp) {
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
}
|
|
return -1;
|
|
}
|
|
if (on_request_creation) {
|
|
if (on_request_creation(session, tdata)) {
|
|
if (generate_new_sdp) {
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
}
|
|
return -1;
|
|
}
|
|
}
|
|
ast_debug(3, "Sending session refresh SDP via %s to %s\n",
|
|
method == AST_SIP_SESSION_REFRESH_METHOD_INVITE ? "re-INVITE" : "UPDATE",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
ast_sip_session_send_request_with_cb(session, tdata, on_response);
|
|
return 0;
|
|
}
|
|
|
|
int ast_sip_session_refresh(struct ast_sip_session *session,
|
|
ast_sip_session_request_creation_cb on_request_creation,
|
|
ast_sip_session_sdp_creation_cb on_sdp_creation,
|
|
ast_sip_session_response_cb on_response,
|
|
enum ast_sip_session_refresh_method method, int generate_new_sdp,
|
|
struct ast_sip_session_media_state *media_state)
|
|
{
|
|
return sip_session_refresh(session, on_request_creation, on_sdp_creation,
|
|
on_response, method, generate_new_sdp, media_state, 0);
|
|
}
|
|
|
|
int ast_sip_session_regenerate_answer(struct ast_sip_session *session,
|
|
ast_sip_session_sdp_creation_cb on_sdp_creation)
|
|
{
|
|
pjsip_inv_session *inv_session = session->inv_session;
|
|
pjmedia_sdp_session *new_answer = NULL;
|
|
const pjmedia_sdp_session *previous_offer = NULL;
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
/* The SDP answer can only be regenerated if it is still pending to be sent */
|
|
if (!inv_session->neg || (pjmedia_sdp_neg_get_state(inv_session->neg) != PJMEDIA_SDP_NEG_STATE_REMOTE_OFFER &&
|
|
pjmedia_sdp_neg_get_state(inv_session->neg) != PJMEDIA_SDP_NEG_STATE_WAIT_NEGO)) {
|
|
ast_log(LOG_WARNING, "Requested to regenerate local SDP answer for channel '%s' but negotiation in state '%s'\n",
|
|
ast_channel_name(session->channel), pjmedia_sdp_neg_state_str(pjmedia_sdp_neg_get_state(inv_session->neg)));
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Bad negotiation state\n");
|
|
}
|
|
|
|
pjmedia_sdp_neg_get_neg_remote(inv_session->neg, &previous_offer);
|
|
if (pjmedia_sdp_neg_get_state(inv_session->neg) == PJMEDIA_SDP_NEG_STATE_WAIT_NEGO) {
|
|
/* Transition the SDP negotiator back to when it received the remote offer */
|
|
pjmedia_sdp_neg_negotiate(inv_session->pool, inv_session->neg, 0);
|
|
pjmedia_sdp_neg_set_remote_offer(inv_session->pool, inv_session->neg, previous_offer);
|
|
}
|
|
|
|
new_answer = create_local_sdp(inv_session, session, previous_offer);
|
|
if (!new_answer) {
|
|
ast_log(LOG_WARNING, "Could not create a new local SDP answer for channel '%s'\n",
|
|
ast_channel_name(session->channel));
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create new SDP\n");
|
|
}
|
|
|
|
if (on_sdp_creation) {
|
|
if (on_sdp_creation(session, new_answer)) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Callback failed\n");
|
|
}
|
|
}
|
|
|
|
pjsip_inv_set_sdp_answer(inv_session, new_answer);
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
void ast_sip_session_send_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
|
|
{
|
|
handle_outgoing_response(session, tdata);
|
|
pjsip_inv_send_msg(session->inv_session, tdata);
|
|
return;
|
|
}
|
|
|
|
static pj_bool_t session_on_rx_request(pjsip_rx_data *rdata);
|
|
static pj_bool_t session_on_rx_response(pjsip_rx_data *rdata);
|
|
static void session_on_tsx_state(pjsip_transaction *tsx, pjsip_event *e);
|
|
|
|
static pjsip_module session_module = {
|
|
.name = {"Session Module", 14},
|
|
.priority = PJSIP_MOD_PRIORITY_APPLICATION,
|
|
.on_rx_request = session_on_rx_request,
|
|
.on_rx_response = session_on_rx_response,
|
|
.on_tsx_state = session_on_tsx_state,
|
|
};
|
|
|
|
/*! \brief Determine whether the SDP provided requires deferral of negotiating or not
|
|
*
|
|
* \retval 1 re-invite should be deferred and resumed later
|
|
* \retval 0 re-invite should not be deferred
|
|
*/
|
|
static int sdp_requires_deferral(struct ast_sip_session *session, const pjmedia_sdp_session *sdp)
|
|
{
|
|
int i;
|
|
|
|
if (!session->pending_media_state->topology) {
|
|
session->pending_media_state->topology = ast_stream_topology_alloc();
|
|
if (!session->pending_media_state->topology) {
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < sdp->media_count; ++i) {
|
|
/* See if there are registered handlers for this media stream type */
|
|
char media[20];
|
|
struct ast_sip_session_sdp_handler *handler;
|
|
RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
|
|
struct ast_stream *existing_stream = NULL;
|
|
struct ast_stream *stream;
|
|
enum ast_media_type type;
|
|
struct ast_sip_session_media *session_media = NULL;
|
|
enum ast_sip_session_sdp_stream_defer res;
|
|
pjmedia_sdp_media *remote_stream = sdp->media[i];
|
|
|
|
/* We need a null-terminated version of the media string */
|
|
ast_copy_pj_str(media, &sdp->media[i]->desc.media, sizeof(media));
|
|
|
|
if (session->active_media_state->topology &&
|
|
(i < ast_stream_topology_get_count(session->active_media_state->topology))) {
|
|
existing_stream = ast_stream_topology_get_stream(session->active_media_state->topology, i);
|
|
}
|
|
|
|
type = ast_media_type_from_str(media);
|
|
stream = ast_stream_alloc(existing_stream ? ast_stream_get_name(existing_stream) : ast_codec_media_type2str(type), type);
|
|
if (!stream) {
|
|
return -1;
|
|
}
|
|
|
|
/* As this is only called on an incoming SDP offer before processing it is not possible
|
|
* for streams and their media sessions to exist.
|
|
*/
|
|
if (ast_stream_topology_set_stream(session->pending_media_state->topology, i, stream)) {
|
|
ast_stream_free(stream);
|
|
return -1;
|
|
}
|
|
|
|
if (existing_stream) {
|
|
const char *stream_label = ast_stream_get_metadata(existing_stream, "SDP:LABEL");
|
|
|
|
if (!ast_strlen_zero(stream_label)) {
|
|
ast_stream_set_metadata(stream, "SDP:LABEL", stream_label);
|
|
}
|
|
}
|
|
|
|
session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_media_type_from_str(media), i);
|
|
if (!session_media) {
|
|
return -1;
|
|
}
|
|
|
|
/* For backwards compatibility with the core the default audio stream is always sendrecv */
|
|
if (!ast_sip_session_is_pending_stream_default(session, stream) || strcmp(media, "audio")) {
|
|
if (pjmedia_sdp_media_find_attr2(remote_stream, "sendonly", NULL)) {
|
|
/* Stream state reflects our state of a stream, so in the case of
|
|
* sendonly and recvonly we store the opposite since that is what ours
|
|
* is.
|
|
*/
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_RECVONLY);
|
|
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "recvonly", NULL)) {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_SENDONLY);
|
|
} else if (pjmedia_sdp_media_find_attr2(remote_stream, "inactive", NULL)) {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_INACTIVE);
|
|
} else {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
|
|
}
|
|
} else {
|
|
ast_stream_set_state(stream, AST_STREAM_STATE_SENDRECV);
|
|
}
|
|
|
|
if (session_media->handler) {
|
|
handler = session_media->handler;
|
|
if (handler->defer_incoming_sdp_stream) {
|
|
res = handler->defer_incoming_sdp_stream(session, session_media, sdp,
|
|
sdp->media[i]);
|
|
switch (res) {
|
|
case AST_SIP_SESSION_SDP_DEFER_NOT_HANDLED:
|
|
break;
|
|
case AST_SIP_SESSION_SDP_DEFER_ERROR:
|
|
return 0;
|
|
case AST_SIP_SESSION_SDP_DEFER_NOT_NEEDED:
|
|
break;
|
|
case AST_SIP_SESSION_SDP_DEFER_NEEDED:
|
|
return 1;
|
|
}
|
|
}
|
|
/* Handled by this handler. Move to the next stream */
|
|
continue;
|
|
}
|
|
|
|
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
|
|
if (!handler_list) {
|
|
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
|
|
continue;
|
|
}
|
|
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
|
|
if (handler == session_media->handler) {
|
|
continue;
|
|
}
|
|
if (!handler->defer_incoming_sdp_stream) {
|
|
continue;
|
|
}
|
|
res = handler->defer_incoming_sdp_stream(session, session_media, sdp,
|
|
sdp->media[i]);
|
|
switch (res) {
|
|
case AST_SIP_SESSION_SDP_DEFER_NOT_HANDLED:
|
|
continue;
|
|
case AST_SIP_SESSION_SDP_DEFER_ERROR:
|
|
session_media_set_handler(session_media, handler);
|
|
return 0;
|
|
case AST_SIP_SESSION_SDP_DEFER_NOT_NEEDED:
|
|
/* Handled by this handler. */
|
|
session_media_set_handler(session_media, handler);
|
|
break;
|
|
case AST_SIP_SESSION_SDP_DEFER_NEEDED:
|
|
/* Handled by this handler. */
|
|
session_media_set_handler(session_media, handler);
|
|
return 1;
|
|
}
|
|
/* Move to the next stream */
|
|
break;
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static pj_bool_t session_reinvite_on_rx_request(pjsip_rx_data *rdata)
|
|
{
|
|
pjsip_dialog *dlg;
|
|
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
|
|
pjsip_rdata_sdp_info *sdp_info;
|
|
int deferred;
|
|
|
|
if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD ||
|
|
!(dlg = pjsip_ua_find_dialog(&rdata->msg_info.cid->id, &rdata->msg_info.to->tag, &rdata->msg_info.from->tag, PJ_FALSE)) ||
|
|
!(session = ast_sip_dialog_get_session(dlg)) ||
|
|
!session->channel) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
if (session->deferred_reinvite) {
|
|
pj_str_t key, deferred_key;
|
|
pjsip_tx_data *tdata;
|
|
|
|
/* We use memory from the new request on purpose so the deferred reinvite pool does not grow uncontrollably */
|
|
pjsip_tsx_create_key(rdata->tp_info.pool, &key, PJSIP_ROLE_UAS, &rdata->msg_info.cseq->method, rdata);
|
|
pjsip_tsx_create_key(rdata->tp_info.pool, &deferred_key, PJSIP_ROLE_UAS, &session->deferred_reinvite->msg_info.cseq->method,
|
|
session->deferred_reinvite);
|
|
|
|
/* If this is a retransmission ignore it */
|
|
if (!pj_strcmp(&key, &deferred_key)) {
|
|
return PJ_TRUE;
|
|
}
|
|
|
|
/* Otherwise this is a new re-invite, so reject it */
|
|
if (pjsip_dlg_create_response(dlg, rdata, 491, NULL, &tdata) == PJ_SUCCESS) {
|
|
if (pjsip_endpt_send_response2(ast_sip_get_pjsip_endpoint(), rdata, tdata, NULL, NULL) != PJ_SUCCESS) {
|
|
pjsip_tx_data_dec_ref(tdata);
|
|
}
|
|
}
|
|
|
|
return PJ_TRUE;
|
|
}
|
|
|
|
if (!(sdp_info = pjsip_rdata_get_sdp_info(rdata)) ||
|
|
(sdp_info->sdp_err != PJ_SUCCESS)) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
if (!sdp_info->sdp) {
|
|
const pjmedia_sdp_session *local;
|
|
int i;
|
|
|
|
ast_queue_unhold(session->channel);
|
|
|
|
pjmedia_sdp_neg_get_active_local(session->inv_session->neg, &local);
|
|
if (!local) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
/*
|
|
* Some devices indicate hold with deferred SDP reinvites (i.e. no SDP in the reinvite).
|
|
* When hold is initially indicated, we
|
|
* - Receive an INVITE with no SDP
|
|
* - Send a 200 OK with SDP, indicating sendrecv in the media streams
|
|
* - Receive an ACK with SDP, indicating sendonly in the media streams
|
|
*
|
|
* At this point, the pjmedia negotiator saves the state of the media direction so that
|
|
* if we are to send any offers, we'll offer recvonly in the media streams. This is
|
|
* problematic if the device is attempting to unhold, though. If the device unholds
|
|
* by sending a reinvite with no SDP, then we will respond with a 200 OK with recvonly.
|
|
* According to RFC 3264, if an offerer offers recvonly, then the answerer MUST respond
|
|
* with sendonly or inactive. The result of this is that the stream is not off hold.
|
|
*
|
|
* Therefore, in this case, when we receive a reinvite while the stream is on hold, we
|
|
* need to be sure to offer sendrecv. This way, the answerer can respond with sendrecv
|
|
* in order to get the stream off hold. If this is actually a different purpose reinvite
|
|
* (like a session timer refresh), then the answerer can respond to our sendrecv with
|
|
* sendonly, keeping the stream on hold.
|
|
*/
|
|
for (i = 0; i < local->media_count; ++i) {
|
|
pjmedia_sdp_media *m = local->media[i];
|
|
pjmedia_sdp_attr *recvonly;
|
|
pjmedia_sdp_attr *inactive;
|
|
|
|
recvonly = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "recvonly", NULL);
|
|
inactive = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "inactive", NULL);
|
|
if (recvonly || inactive) {
|
|
pjmedia_sdp_attr *to_remove = recvonly ?: inactive;
|
|
pjmedia_sdp_attr *sendrecv;
|
|
|
|
pjmedia_sdp_attr_remove(&m->attr_count, m->attr, to_remove);
|
|
|
|
sendrecv = pjmedia_sdp_attr_create(session->inv_session->pool, "sendrecv", NULL);
|
|
pjmedia_sdp_media_add_attr(m, sendrecv);
|
|
}
|
|
}
|
|
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
deferred = sdp_requires_deferral(session, sdp_info->sdp);
|
|
if (deferred == -1) {
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
return PJ_FALSE;
|
|
} else if (!deferred) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
pjsip_rx_data_clone(rdata, 0, &session->deferred_reinvite);
|
|
|
|
return PJ_TRUE;
|
|
}
|
|
|
|
void ast_sip_session_resume_reinvite(struct ast_sip_session *session)
|
|
{
|
|
if (!session->deferred_reinvite) {
|
|
return;
|
|
}
|
|
|
|
if (session->channel) {
|
|
pjsip_endpt_process_rx_data(ast_sip_get_pjsip_endpoint(),
|
|
session->deferred_reinvite, NULL, NULL);
|
|
}
|
|
pjsip_rx_data_free_cloned(session->deferred_reinvite);
|
|
session->deferred_reinvite = NULL;
|
|
}
|
|
|
|
static pjsip_module session_reinvite_module = {
|
|
.name = { "Session Re-Invite Module", 24 },
|
|
.priority = PJSIP_MOD_PRIORITY_UA_PROXY_LAYER - 1,
|
|
.on_rx_request = session_reinvite_on_rx_request,
|
|
};
|
|
|
|
|
|
void ast_sip_session_send_request_with_cb(struct ast_sip_session *session, pjsip_tx_data *tdata,
|
|
ast_sip_session_response_cb on_response)
|
|
{
|
|
pjsip_inv_session *inv_session = session->inv_session;
|
|
|
|
/* For every request except BYE we disallow sending of the message when
|
|
* the session has been disconnected. A BYE request is special though
|
|
* because it can be sent again after the session is disconnected except
|
|
* with credentials.
|
|
*/
|
|
if (inv_session->state == PJSIP_INV_STATE_DISCONNECTED &&
|
|
tdata->msg->line.req.method.id != PJSIP_BYE_METHOD) {
|
|
return;
|
|
}
|
|
|
|
ast_sip_mod_data_set(tdata->pool, tdata->mod_data, session_module.id,
|
|
MOD_DATA_ON_RESPONSE, on_response);
|
|
|
|
handle_outgoing_request(session, tdata);
|
|
pjsip_inv_send_msg(session->inv_session, tdata);
|
|
|
|
return;
|
|
}
|
|
|
|
void ast_sip_session_send_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
|
|
{
|
|
ast_sip_session_send_request_with_cb(session, tdata, NULL);
|
|
}
|
|
|
|
int ast_sip_session_create_invite(struct ast_sip_session *session, pjsip_tx_data **tdata)
|
|
{
|
|
pjmedia_sdp_session *offer;
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
if (!(offer = create_local_sdp(session->inv_session, session, NULL))) {
|
|
pjsip_inv_terminate(session->inv_session, 500, PJ_FALSE);
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Couldn't create offer\n");
|
|
}
|
|
|
|
pjsip_inv_set_local_sdp(session->inv_session, offer);
|
|
pjmedia_sdp_neg_set_prefer_remote_codec_order(session->inv_session->neg, PJ_FALSE);
|
|
#ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
|
|
if (!session->endpoint->preferred_codec_only) {
|
|
pjmedia_sdp_neg_set_answer_multiple_codecs(session->inv_session->neg, PJ_TRUE);
|
|
}
|
|
#endif
|
|
|
|
/*
|
|
* We MUST call set_from_header() before pjsip_inv_invite. If we don't, the
|
|
* From in the initial INVITE will be wrong but the rest of the messages will be OK.
|
|
*/
|
|
set_from_header(session);
|
|
|
|
if (pjsip_inv_invite(session->inv_session, tdata) != PJ_SUCCESS) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "pjsip_inv_invite failed\n");
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
static int datastore_hash(const void *obj, int flags)
|
|
{
|
|
const struct ast_datastore *datastore = obj;
|
|
const char *uid = flags & OBJ_KEY ? obj : datastore->uid;
|
|
|
|
ast_assert(uid != NULL);
|
|
|
|
return ast_str_hash(uid);
|
|
}
|
|
|
|
static int datastore_cmp(void *obj, void *arg, int flags)
|
|
{
|
|
const struct ast_datastore *datastore1 = obj;
|
|
const struct ast_datastore *datastore2 = arg;
|
|
const char *uid2 = flags & OBJ_KEY ? arg : datastore2->uid;
|
|
|
|
ast_assert(datastore1->uid != NULL);
|
|
ast_assert(uid2 != NULL);
|
|
|
|
return strcmp(datastore1->uid, uid2) ? 0 : CMP_MATCH | CMP_STOP;
|
|
}
|
|
|
|
static void session_destructor(void *obj)
|
|
{
|
|
struct ast_sip_session *session = obj;
|
|
struct ast_sip_session_delayed_request *delay;
|
|
|
|
/* We dup the endpoint ID in case the endpoint gets freed out from under us */
|
|
const char *endpoint_name = session->endpoint ?
|
|
ast_strdupa(ast_sorcery_object_get_id(session->endpoint)) : "<none>";
|
|
|
|
ast_debug(3, "Destroying SIP session with endpoint %s\n", endpoint_name);
|
|
|
|
ast_test_suite_event_notify("SESSION_DESTROYING",
|
|
"Endpoint: %s\r\n"
|
|
"AOR: %s\r\n"
|
|
"Contact: %s"
|
|
, endpoint_name
|
|
, session->aor ? ast_sorcery_object_get_id(session->aor) : "<none>"
|
|
, session->contact ? ast_sorcery_object_get_id(session->contact) : "<none>"
|
|
);
|
|
|
|
/* fire session destroy handler */
|
|
handle_session_destroy(session);
|
|
|
|
/* remove all registered supplements */
|
|
ast_sip_session_remove_supplements(session);
|
|
AST_LIST_HEAD_DESTROY(&session->supplements);
|
|
|
|
/* remove all saved media stats */
|
|
AST_VECTOR_RESET(&session->media_stats, ast_free);
|
|
AST_VECTOR_FREE(&session->media_stats);
|
|
|
|
ast_taskprocessor_unreference(session->serializer);
|
|
ao2_cleanup(session->datastores);
|
|
ast_sip_session_media_state_free(session->active_media_state);
|
|
ast_sip_session_media_state_free(session->pending_media_state);
|
|
|
|
while ((delay = AST_LIST_REMOVE_HEAD(&session->delayed_requests, next))) {
|
|
delayed_request_free(delay);
|
|
}
|
|
ast_party_id_free(&session->id);
|
|
ao2_cleanup(session->endpoint);
|
|
ao2_cleanup(session->aor);
|
|
ao2_cleanup(session->contact);
|
|
ao2_cleanup(session->direct_media_cap);
|
|
|
|
ast_dsp_free(session->dsp);
|
|
|
|
if (session->inv_session) {
|
|
pjsip_dlg_dec_session(session->inv_session->dlg, &session_module);
|
|
}
|
|
|
|
ast_test_suite_event_notify("SESSION_DESTROYED", "Endpoint: %s", endpoint_name);
|
|
}
|
|
|
|
/*! \brief Destructor for SIP channel */
|
|
static void sip_channel_destroy(void *obj)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = obj;
|
|
|
|
ao2_cleanup(channel->pvt);
|
|
ao2_cleanup(channel->session);
|
|
}
|
|
|
|
struct ast_sip_channel_pvt *ast_sip_channel_pvt_alloc(void *pvt, struct ast_sip_session *session)
|
|
{
|
|
struct ast_sip_channel_pvt *channel = ao2_alloc(sizeof(*channel), sip_channel_destroy);
|
|
|
|
if (!channel) {
|
|
return NULL;
|
|
}
|
|
|
|
ao2_ref(pvt, +1);
|
|
channel->pvt = pvt;
|
|
ao2_ref(session, +1);
|
|
channel->session = session;
|
|
|
|
return channel;
|
|
}
|
|
|
|
struct ast_sip_session *ast_sip_session_alloc(struct ast_sip_endpoint *endpoint,
|
|
struct ast_sip_contact *contact, pjsip_inv_session *inv_session, pjsip_rx_data *rdata)
|
|
{
|
|
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
|
|
struct ast_sip_session *ret_session;
|
|
int dsp_features = 0;
|
|
|
|
session = ao2_alloc(sizeof(*session), session_destructor);
|
|
if (!session) {
|
|
return NULL;
|
|
}
|
|
|
|
AST_LIST_HEAD_INIT(&session->supplements);
|
|
AST_LIST_HEAD_INIT_NOLOCK(&session->delayed_requests);
|
|
ast_party_id_init(&session->id);
|
|
|
|
session->direct_media_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
|
|
if (!session->direct_media_cap) {
|
|
return NULL;
|
|
}
|
|
session->datastores = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
|
|
DATASTORE_BUCKETS, datastore_hash, NULL, datastore_cmp);
|
|
if (!session->datastores) {
|
|
return NULL;
|
|
}
|
|
session->active_media_state = ast_sip_session_media_state_alloc();
|
|
if (!session->active_media_state) {
|
|
return NULL;
|
|
}
|
|
session->pending_media_state = ast_sip_session_media_state_alloc();
|
|
if (!session->pending_media_state) {
|
|
return NULL;
|
|
}
|
|
if (AST_VECTOR_INIT(&session->media_stats, 1) < 0) {
|
|
return NULL;
|
|
}
|
|
|
|
if (endpoint->dtmf == AST_SIP_DTMF_INBAND || endpoint->dtmf == AST_SIP_DTMF_AUTO) {
|
|
dsp_features |= DSP_FEATURE_DIGIT_DETECT;
|
|
}
|
|
if (endpoint->faxdetect) {
|
|
dsp_features |= DSP_FEATURE_FAX_DETECT;
|
|
}
|
|
if (dsp_features) {
|
|
session->dsp = ast_dsp_new();
|
|
if (!session->dsp) {
|
|
return NULL;
|
|
}
|
|
|
|
ast_dsp_set_features(session->dsp, dsp_features);
|
|
}
|
|
|
|
session->endpoint = ao2_bump(endpoint);
|
|
|
|
if (rdata) {
|
|
/*
|
|
* We must continue using the serializer that the original
|
|
* INVITE came in on for the dialog. There may be
|
|
* retransmissions already enqueued in the original
|
|
* serializer that can result in reentrancy and message
|
|
* sequencing problems.
|
|
*/
|
|
session->serializer = ast_sip_get_distributor_serializer(rdata);
|
|
} else {
|
|
char tps_name[AST_TASKPROCESSOR_MAX_NAME + 1];
|
|
|
|
/* Create name with seq number appended. */
|
|
ast_taskprocessor_build_name(tps_name, sizeof(tps_name), "pjsip/outsess/%s",
|
|
ast_sorcery_object_get_id(endpoint));
|
|
|
|
session->serializer = ast_sip_create_serializer(tps_name);
|
|
}
|
|
if (!session->serializer) {
|
|
return NULL;
|
|
}
|
|
ast_sip_dialog_set_serializer(inv_session->dlg, session->serializer);
|
|
ast_sip_dialog_set_endpoint(inv_session->dlg, endpoint);
|
|
pjsip_dlg_inc_session(inv_session->dlg, &session_module);
|
|
inv_session->mod_data[session_module.id] = ao2_bump(session);
|
|
session->contact = ao2_bump(contact);
|
|
session->inv_session = inv_session;
|
|
|
|
session->dtmf = endpoint->dtmf;
|
|
session->moh_passthrough = endpoint->moh_passthrough;
|
|
|
|
if (ast_sip_session_add_supplements(session)) {
|
|
/* Release the ref held by session->inv_session */
|
|
ao2_ref(session, -1);
|
|
return NULL;
|
|
}
|
|
|
|
/* Fire seesion begin handlers */
|
|
handle_session_begin(session);
|
|
|
|
/* Avoid unnecessary ref manipulation to return a session */
|
|
ret_session = session;
|
|
session = NULL;
|
|
return ret_session;
|
|
}
|
|
|
|
/*! \brief struct controlling the suspension of the session's serializer. */
|
|
struct ast_sip_session_suspender {
|
|
ast_cond_t cond_suspended;
|
|
ast_cond_t cond_complete;
|
|
int suspended;
|
|
int complete;
|
|
};
|
|
|
|
static void sip_session_suspender_dtor(void *vdoomed)
|
|
{
|
|
struct ast_sip_session_suspender *doomed = vdoomed;
|
|
|
|
ast_cond_destroy(&doomed->cond_suspended);
|
|
ast_cond_destroy(&doomed->cond_complete);
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Block the session serializer thread task.
|
|
*
|
|
* \param data Pushed serializer task data for suspension.
|
|
*
|
|
* \retval 0
|
|
*/
|
|
static int sip_session_suspend_task(void *data)
|
|
{
|
|
struct ast_sip_session_suspender *suspender = data;
|
|
|
|
ao2_lock(suspender);
|
|
|
|
/* Signal that the serializer task is now suspended. */
|
|
suspender->suspended = 1;
|
|
ast_cond_signal(&suspender->cond_suspended);
|
|
|
|
/* Wait for the serializer suspension to be completed. */
|
|
while (!suspender->complete) {
|
|
ast_cond_wait(&suspender->cond_complete, ao2_object_get_lockaddr(suspender));
|
|
}
|
|
|
|
ao2_unlock(suspender);
|
|
ao2_ref(suspender, -1);
|
|
|
|
return 0;
|
|
}
|
|
|
|
void ast_sip_session_suspend(struct ast_sip_session *session)
|
|
{
|
|
struct ast_sip_session_suspender *suspender;
|
|
int res;
|
|
|
|
ast_assert(session->suspended == NULL);
|
|
|
|
if (ast_taskprocessor_is_task(session->serializer)) {
|
|
/* I am the session's serializer thread so I cannot suspend. */
|
|
return;
|
|
}
|
|
|
|
if (ast_taskprocessor_is_suspended(session->serializer)) {
|
|
/* The serializer already suspended. */
|
|
return;
|
|
}
|
|
|
|
suspender = ao2_alloc(sizeof(*suspender), sip_session_suspender_dtor);
|
|
if (!suspender) {
|
|
/* We will just have to hope that the system does not deadlock */
|
|
return;
|
|
}
|
|
ast_cond_init(&suspender->cond_suspended, NULL);
|
|
ast_cond_init(&suspender->cond_complete, NULL);
|
|
|
|
ao2_ref(suspender, +1);
|
|
res = ast_sip_push_task(session->serializer, sip_session_suspend_task, suspender);
|
|
if (res) {
|
|
/* We will just have to hope that the system does not deadlock */
|
|
ao2_ref(suspender, -2);
|
|
return;
|
|
}
|
|
|
|
session->suspended = suspender;
|
|
|
|
/* Wait for the serializer to get suspended. */
|
|
ao2_lock(suspender);
|
|
while (!suspender->suspended) {
|
|
ast_cond_wait(&suspender->cond_suspended, ao2_object_get_lockaddr(suspender));
|
|
}
|
|
ao2_unlock(suspender);
|
|
|
|
ast_taskprocessor_suspend(session->serializer);
|
|
}
|
|
|
|
void ast_sip_session_unsuspend(struct ast_sip_session *session)
|
|
{
|
|
struct ast_sip_session_suspender *suspender = session->suspended;
|
|
|
|
if (!suspender) {
|
|
/* Nothing to do */
|
|
return;
|
|
}
|
|
session->suspended = NULL;
|
|
|
|
/* Signal that the serializer task suspension is now complete. */
|
|
ao2_lock(suspender);
|
|
suspender->complete = 1;
|
|
ast_cond_signal(&suspender->cond_complete);
|
|
ao2_unlock(suspender);
|
|
|
|
ao2_ref(suspender, -1);
|
|
|
|
ast_taskprocessor_unsuspend(session->serializer);
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Handle initial INVITE challenge response message.
|
|
* \since 13.5.0
|
|
*
|
|
* \param rdata PJSIP receive response message data.
|
|
*
|
|
* \retval PJ_FALSE Did not handle message.
|
|
* \retval PJ_TRUE Handled message.
|
|
*/
|
|
static pj_bool_t outbound_invite_auth(pjsip_rx_data *rdata)
|
|
{
|
|
pjsip_transaction *tsx;
|
|
pjsip_dialog *dlg;
|
|
pjsip_inv_session *inv;
|
|
pjsip_tx_data *tdata;
|
|
struct ast_sip_session *session;
|
|
|
|
if (rdata->msg_info.msg->line.status.code != 401
|
|
&& rdata->msg_info.msg->line.status.code != 407) {
|
|
/* Doesn't pertain to us. Move on */
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
tsx = pjsip_rdata_get_tsx(rdata);
|
|
dlg = pjsip_rdata_get_dlg(rdata);
|
|
if (!dlg || !tsx) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
if (tsx->method.id != PJSIP_INVITE_METHOD) {
|
|
/* Not an INVITE that needs authentication */
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
inv = pjsip_dlg_get_inv_session(dlg);
|
|
if (PJSIP_INV_STATE_CONFIRMED <= inv->state) {
|
|
/*
|
|
* We cannot handle reINVITE authentication at this
|
|
* time because the reINVITE transaction is still in
|
|
* progress.
|
|
*/
|
|
ast_debug(1, "A reINVITE is being challenged.\n");
|
|
return PJ_FALSE;
|
|
}
|
|
ast_debug(1, "Initial INVITE is being challenged.\n");
|
|
|
|
session = inv->mod_data[session_module.id];
|
|
|
|
if (ast_sip_create_request_with_auth(&session->endpoint->outbound_auths, rdata,
|
|
tsx->last_tx, &tdata)) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
/*
|
|
* Restart the outgoing initial INVITE transaction to deal
|
|
* with authentication.
|
|
*/
|
|
pjsip_inv_uac_restart(inv, PJ_FALSE);
|
|
|
|
ast_sip_session_send_request(session, tdata);
|
|
return PJ_TRUE;
|
|
}
|
|
|
|
static pjsip_module outbound_invite_auth_module = {
|
|
.name = {"Outbound INVITE Auth", 20},
|
|
.priority = PJSIP_MOD_PRIORITY_DIALOG_USAGE,
|
|
.on_rx_response = outbound_invite_auth,
|
|
};
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Setup outbound initial INVITE authentication.
|
|
* \since 13.5.0
|
|
*
|
|
* \param dlg PJSIP dialog to attach outbound authentication.
|
|
*
|
|
* \retval 0 on success.
|
|
* \retval -1 on error.
|
|
*/
|
|
static int setup_outbound_invite_auth(pjsip_dialog *dlg)
|
|
{
|
|
pj_status_t status;
|
|
|
|
++dlg->sess_count;
|
|
status = pjsip_dlg_add_usage(dlg, &outbound_invite_auth_module, NULL);
|
|
--dlg->sess_count;
|
|
|
|
return status != PJ_SUCCESS ? -1 : 0;
|
|
}
|
|
|
|
struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint *endpoint,
|
|
struct ast_sip_contact *contact, const char *location, const char *request_user,
|
|
struct ast_stream_topology *req_topology)
|
|
{
|
|
const char *uri = NULL;
|
|
RAII_VAR(struct ast_sip_aor *, found_aor, NULL, ao2_cleanup);
|
|
RAII_VAR(struct ast_sip_contact *, found_contact, NULL, ao2_cleanup);
|
|
pjsip_timer_setting timer;
|
|
pjsip_dialog *dlg;
|
|
struct pjsip_inv_session *inv_session;
|
|
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
|
|
struct ast_sip_session *ret_session;
|
|
SCOPE_ENTER(1, "%s %s Topology: %s\n", ast_sorcery_object_get_id(endpoint), request_user,
|
|
ast_str_tmp(256, ast_stream_topology_to_str(req_topology, &STR_TMP)));
|
|
|
|
/* If no location has been provided use the AOR list from the endpoint itself */
|
|
if (location || !contact) {
|
|
location = S_OR(location, endpoint->aors);
|
|
|
|
ast_sip_location_retrieve_contact_and_aor_from_list_filtered(location, AST_SIP_CONTACT_FILTER_REACHABLE,
|
|
&found_aor, &found_contact);
|
|
if (!found_contact || ast_strlen_zero(found_contact->uri)) {
|
|
uri = location;
|
|
} else {
|
|
uri = found_contact->uri;
|
|
}
|
|
} else {
|
|
uri = contact->uri;
|
|
}
|
|
|
|
/* If we still have no URI to dial fail to create the session */
|
|
if (ast_strlen_zero(uri)) {
|
|
ast_log(LOG_ERROR, "Endpoint '%s': No URI available. Is endpoint registered?\n",
|
|
ast_sorcery_object_get_id(endpoint));
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "No URI\n");
|
|
}
|
|
|
|
if (!(dlg = ast_sip_create_dialog_uac(endpoint, uri, request_user))) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create dialog\n");
|
|
}
|
|
|
|
if (setup_outbound_invite_auth(dlg)) {
|
|
pjsip_dlg_terminate(dlg);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't setup auth\n");
|
|
}
|
|
|
|
if (pjsip_inv_create_uac(dlg, NULL, endpoint->extensions.flags, &inv_session) != PJ_SUCCESS) {
|
|
pjsip_dlg_terminate(dlg);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create uac\n");
|
|
}
|
|
#if defined(HAVE_PJSIP_REPLACE_MEDIA_STREAM) || defined(PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE)
|
|
inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
|
|
#endif
|
|
|
|
pjsip_timer_setting_default(&timer);
|
|
timer.min_se = endpoint->extensions.timer.min_se;
|
|
timer.sess_expires = endpoint->extensions.timer.sess_expires;
|
|
pjsip_timer_init_session(inv_session, &timer);
|
|
|
|
session = ast_sip_session_alloc(endpoint, found_contact ? found_contact : contact,
|
|
inv_session, NULL);
|
|
if (!session) {
|
|
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
|
|
return NULL;
|
|
}
|
|
session->aor = ao2_bump(found_aor);
|
|
session->call_direction = AST_SIP_SESSION_OUTGOING_CALL;
|
|
|
|
ast_party_id_copy(&session->id, &endpoint->id.self);
|
|
|
|
if (ast_stream_topology_get_count(req_topology) > 0) {
|
|
/* get joint caps between req_topology and endpoint topology */
|
|
int i;
|
|
|
|
for (i = 0; i < ast_stream_topology_get_count(req_topology); ++i) {
|
|
struct ast_stream *req_stream;
|
|
struct ast_stream *clone_stream;
|
|
|
|
req_stream = ast_stream_topology_get_stream(req_topology, i);
|
|
|
|
if (ast_stream_get_state(req_stream) == AST_STREAM_STATE_REMOVED) {
|
|
continue;
|
|
}
|
|
|
|
clone_stream = ast_sip_session_create_joint_call_stream(session, req_stream);
|
|
if (!clone_stream || ast_stream_get_format_count(clone_stream) == 0) {
|
|
ast_stream_free(clone_stream);
|
|
continue;
|
|
}
|
|
|
|
if (!session->pending_media_state->topology) {
|
|
session->pending_media_state->topology = ast_stream_topology_alloc();
|
|
if (!session->pending_media_state->topology) {
|
|
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
|
|
ao2_ref(session, -1);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't create topology\n");
|
|
}
|
|
}
|
|
|
|
if (ast_stream_topology_append_stream(session->pending_media_state->topology, clone_stream) < 0) {
|
|
ast_stream_free(clone_stream);
|
|
continue;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (!session->pending_media_state->topology) {
|
|
/* Use the configured topology on the endpoint as the pending one */
|
|
session->pending_media_state->topology = ast_stream_topology_clone(endpoint->media.topology);
|
|
if (!session->pending_media_state->topology) {
|
|
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
|
|
ao2_ref(session, -1);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't clone topology\n");
|
|
}
|
|
}
|
|
|
|
if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
|
|
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
|
|
/* Since we are not notifying ourselves that the INVITE session is being terminated
|
|
* we need to manually drop its reference to session
|
|
*/
|
|
ao2_ref(session, -1);
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't add usage\n");
|
|
}
|
|
|
|
/* Avoid unnecessary ref manipulation to return a session */
|
|
ret_session = session;
|
|
session = NULL;
|
|
SCOPE_EXIT_RTN_VALUE(ret_session);
|
|
}
|
|
|
|
static int session_end(void *vsession);
|
|
static int session_end_completion(void *vsession);
|
|
|
|
void ast_sip_session_terminate(struct ast_sip_session *session, int response)
|
|
{
|
|
pj_status_t status;
|
|
pjsip_tx_data *packet = NULL;
|
|
SCOPE_ENTER(1, "%s Response %d\n", ast_sip_session_get_name(session), response);
|
|
|
|
if (session->defer_terminate) {
|
|
session->terminate_while_deferred = 1;
|
|
SCOPE_EXIT_RTN("Deferred\n");
|
|
}
|
|
|
|
if (!response) {
|
|
response = 603;
|
|
}
|
|
|
|
/* The media sessions need to exist for the lifetime of the underlying channel
|
|
* to ensure that anything (such as bridge_native_rtp) has access to them as
|
|
* appropriate. Since ast_sip_session_terminate is called by chan_pjsip and other
|
|
* places when the session is to be terminated we terminate any existing
|
|
* media sessions here.
|
|
*/
|
|
ast_sip_session_media_stats_save(session, session->active_media_state);
|
|
SWAP(session->active_media_state, session->pending_media_state);
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
|
|
switch (session->inv_session->state) {
|
|
case PJSIP_INV_STATE_NULL:
|
|
if (!session->inv_session->invite_tsx) {
|
|
/*
|
|
* Normally, it's pjproject's transaction cleanup that ultimately causes the
|
|
* final session reference to be released but if both STATE and invite_tsx are NULL,
|
|
* we never created a transaction in the first place. In this case, we need to
|
|
* do the cleanup ourselves.
|
|
*/
|
|
/* Transfer the inv_session session reference to the session_end_task */
|
|
session->inv_session->mod_data[session_module.id] = NULL;
|
|
pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
|
|
session_end(session);
|
|
/*
|
|
* session_end_completion will cleanup the final session reference unless
|
|
* ast_sip_session_terminate's caller is holding one.
|
|
*/
|
|
session_end_completion(session);
|
|
} else {
|
|
pjsip_inv_terminate(session->inv_session, response, PJ_TRUE);
|
|
}
|
|
break;
|
|
case PJSIP_INV_STATE_CONFIRMED:
|
|
if (session->inv_session->invite_tsx) {
|
|
ast_debug(3, "Delay sending BYE to %s because of outstanding transaction...\n",
|
|
ast_sorcery_object_get_id(session->endpoint));
|
|
/* If this is delayed the only thing that will happen is a BYE request so we don't
|
|
* actually need to store the response code for when it happens.
|
|
*/
|
|
delay_request(session, NULL, NULL, NULL, 0, DELAYED_METHOD_BYE, NULL);
|
|
break;
|
|
}
|
|
/* Fall through */
|
|
default:
|
|
status = pjsip_inv_end_session(session->inv_session, response, NULL, &packet);
|
|
if (status == PJ_SUCCESS && packet) {
|
|
struct ast_sip_session_delayed_request *delay;
|
|
|
|
/* Flush any delayed requests so they cannot overlap this transaction. */
|
|
while ((delay = AST_LIST_REMOVE_HEAD(&session->delayed_requests, next))) {
|
|
delayed_request_free(delay);
|
|
}
|
|
|
|
if (packet->msg->type == PJSIP_RESPONSE_MSG) {
|
|
ast_sip_session_send_response(session, packet);
|
|
} else {
|
|
ast_sip_session_send_request(session, packet);
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static int session_termination_task(void *data)
|
|
{
|
|
struct ast_sip_session *session = data;
|
|
|
|
if (session->defer_terminate) {
|
|
session->defer_terminate = 0;
|
|
if (session->inv_session) {
|
|
ast_sip_session_terminate(session, 0);
|
|
}
|
|
}
|
|
|
|
ao2_ref(session, -1);
|
|
return 0;
|
|
}
|
|
|
|
static void session_termination_cb(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry)
|
|
{
|
|
struct ast_sip_session *session = entry->user_data;
|
|
|
|
if (ast_sip_push_task(session->serializer, session_termination_task, session)) {
|
|
ao2_cleanup(session);
|
|
}
|
|
}
|
|
|
|
int ast_sip_session_defer_termination(struct ast_sip_session *session)
|
|
{
|
|
pj_time_val delay = { .sec = 60, };
|
|
int res;
|
|
|
|
/* The session should not have an active deferred termination request. */
|
|
ast_assert(!session->defer_terminate);
|
|
|
|
session->defer_terminate = 1;
|
|
|
|
session->defer_end = 1;
|
|
session->ended_while_deferred = 0;
|
|
|
|
ao2_ref(session, +1);
|
|
pj_timer_entry_init(&session->scheduled_termination, 0, session, session_termination_cb);
|
|
|
|
res = (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(),
|
|
&session->scheduled_termination, &delay) != PJ_SUCCESS) ? -1 : 0;
|
|
if (res) {
|
|
session->defer_terminate = 0;
|
|
ao2_ref(session, -1);
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Stop the defer termination timer if it is still running.
|
|
* \since 13.5.0
|
|
*
|
|
* \param session Which session to stop the timer.
|
|
*
|
|
* \return Nothing
|
|
*/
|
|
static void sip_session_defer_termination_stop_timer(struct ast_sip_session *session)
|
|
{
|
|
if (pj_timer_heap_cancel_if_active(pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()),
|
|
&session->scheduled_termination, session->scheduled_termination.id)) {
|
|
ao2_ref(session, -1);
|
|
}
|
|
}
|
|
|
|
void ast_sip_session_defer_termination_cancel(struct ast_sip_session *session)
|
|
{
|
|
if (!session->defer_terminate) {
|
|
/* Already canceled or timer fired. */
|
|
return;
|
|
}
|
|
|
|
session->defer_terminate = 0;
|
|
|
|
if (session->terminate_while_deferred) {
|
|
/* Complete the termination started by the upper layer. */
|
|
ast_sip_session_terminate(session, 0);
|
|
}
|
|
|
|
/* Stop the termination timer if it is still running. */
|
|
sip_session_defer_termination_stop_timer(session);
|
|
}
|
|
|
|
void ast_sip_session_end_if_deferred(struct ast_sip_session *session)
|
|
{
|
|
if (!session->defer_end) {
|
|
return;
|
|
}
|
|
|
|
session->defer_end = 0;
|
|
|
|
if (session->ended_while_deferred) {
|
|
/* Complete the session end started by the remote hangup. */
|
|
ast_debug(3, "Ending session (%p) after being deferred\n", session);
|
|
session->ended_while_deferred = 0;
|
|
session_end(session);
|
|
}
|
|
}
|
|
|
|
struct ast_sip_session *ast_sip_dialog_get_session(pjsip_dialog *dlg)
|
|
{
|
|
pjsip_inv_session *inv_session = pjsip_dlg_get_inv_session(dlg);
|
|
struct ast_sip_session *session;
|
|
|
|
if (!inv_session ||
|
|
!(session = inv_session->mod_data[session_module.id])) {
|
|
return NULL;
|
|
}
|
|
|
|
ao2_ref(session, +1);
|
|
|
|
return session;
|
|
}
|
|
|
|
enum sip_get_destination_result {
|
|
/*! The extension was successfully found */
|
|
SIP_GET_DEST_EXTEN_FOUND,
|
|
/*! The extension specified in the RURI was not found */
|
|
SIP_GET_DEST_EXTEN_NOT_FOUND,
|
|
/*! The extension specified in the RURI was a partial match */
|
|
SIP_GET_DEST_EXTEN_PARTIAL,
|
|
/*! The RURI is of an unsupported scheme */
|
|
SIP_GET_DEST_UNSUPPORTED_URI,
|
|
};
|
|
|
|
/*!
|
|
* \brief Determine where in the dialplan a call should go
|
|
*
|
|
* This uses the username in the request URI to try to match
|
|
* an extension in the endpoint's configured context in order
|
|
* to route the call.
|
|
*
|
|
* \param session The inbound SIP session
|
|
* \param rdata The SIP INVITE
|
|
*/
|
|
static enum sip_get_destination_result get_destination(struct ast_sip_session *session, pjsip_rx_data *rdata)
|
|
{
|
|
pjsip_uri *ruri = rdata->msg_info.msg->line.req.uri;
|
|
pjsip_sip_uri *sip_ruri;
|
|
struct ast_features_pickup_config *pickup_cfg;
|
|
const char *pickupexten;
|
|
|
|
if (!PJSIP_URI_SCHEME_IS_SIP(ruri) && !PJSIP_URI_SCHEME_IS_SIPS(ruri)) {
|
|
return SIP_GET_DEST_UNSUPPORTED_URI;
|
|
}
|
|
|
|
sip_ruri = pjsip_uri_get_uri(ruri);
|
|
ast_copy_pj_str(session->exten, &sip_ruri->user, sizeof(session->exten));
|
|
|
|
/*
|
|
* We may want to match in the dialplan without any user
|
|
* options getting in the way.
|
|
*/
|
|
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten);
|
|
|
|
pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
|
|
if (!pickup_cfg) {
|
|
ast_log(LOG_ERROR, "Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n");
|
|
pickupexten = "";
|
|
} else {
|
|
pickupexten = ast_strdupa(pickup_cfg->pickupexten);
|
|
ao2_ref(pickup_cfg, -1);
|
|
}
|
|
|
|
if (!strcmp(session->exten, pickupexten) ||
|
|
ast_exists_extension(NULL, session->endpoint->context, session->exten, 1, NULL)) {
|
|
size_t size = pj_strlen(&sip_ruri->host) + 1;
|
|
char *domain = ast_alloca(size);
|
|
|
|
ast_copy_pj_str(domain, &sip_ruri->host, size);
|
|
pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
|
|
|
|
/*
|
|
* Save off the INVITE Request-URI in case it is
|
|
* needed: CHANNEL(pjsip,request_uri)
|
|
*/
|
|
session->request_uri = pjsip_uri_clone(session->inv_session->pool, ruri);
|
|
|
|
return SIP_GET_DEST_EXTEN_FOUND;
|
|
}
|
|
|
|
/*
|
|
* Check for partial match via overlap dialling (if enabled)
|
|
*/
|
|
if (session->endpoint->allow_overlap && (
|
|
!strncmp(session->exten, pickupexten, strlen(session->exten)) ||
|
|
ast_canmatch_extension(NULL, session->endpoint->context, session->exten, 1, NULL))) {
|
|
/* Overlap partial match */
|
|
return SIP_GET_DEST_EXTEN_PARTIAL;
|
|
}
|
|
|
|
return SIP_GET_DEST_EXTEN_NOT_FOUND;
|
|
}
|
|
|
|
static pjsip_inv_session *pre_session_setup(pjsip_rx_data *rdata, const struct ast_sip_endpoint *endpoint)
|
|
{
|
|
pjsip_tx_data *tdata;
|
|
pjsip_dialog *dlg;
|
|
pjsip_inv_session *inv_session;
|
|
unsigned int options = endpoint->extensions.flags;
|
|
pj_status_t dlg_status;
|
|
|
|
if (pjsip_inv_verify_request(rdata, &options, NULL, NULL, ast_sip_get_pjsip_endpoint(), &tdata) != PJ_SUCCESS) {
|
|
if (tdata) {
|
|
if (pjsip_endpt_send_response2(ast_sip_get_pjsip_endpoint(), rdata, tdata, NULL, NULL) != PJ_SUCCESS) {
|
|
pjsip_tx_data_dec_ref(tdata);
|
|
}
|
|
} else {
|
|
pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
|
|
}
|
|
return NULL;
|
|
}
|
|
dlg = ast_sip_create_dialog_uas(endpoint, rdata, &dlg_status);
|
|
if (!dlg) {
|
|
if (dlg_status != PJ_EEXISTS) {
|
|
pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
|
|
}
|
|
return NULL;
|
|
}
|
|
if (pjsip_inv_create_uas(dlg, rdata, NULL, options, &inv_session) != PJ_SUCCESS) {
|
|
pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, 500, NULL, NULL, NULL);
|
|
pjsip_dlg_terminate(dlg);
|
|
return NULL;
|
|
}
|
|
|
|
#if defined(HAVE_PJSIP_REPLACE_MEDIA_STREAM) || defined(PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE)
|
|
inv_session->sdp_neg_flags = PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE;
|
|
#endif
|
|
if (pjsip_dlg_add_usage(dlg, &session_module, NULL) != PJ_SUCCESS) {
|
|
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) != PJ_SUCCESS) {
|
|
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
|
|
}
|
|
pjsip_inv_send_msg(inv_session, tdata);
|
|
return NULL;
|
|
}
|
|
return inv_session;
|
|
}
|
|
|
|
struct new_invite {
|
|
/*! \brief Session created for the new INVITE */
|
|
struct ast_sip_session *session;
|
|
|
|
/*! \brief INVITE request itself */
|
|
pjsip_rx_data *rdata;
|
|
};
|
|
|
|
static int check_sdp_content_type_supported(pjsip_media_type *content_type)
|
|
{
|
|
pjsip_media_type app_sdp;
|
|
pjsip_media_type_init2(&app_sdp, "application", "sdp");
|
|
|
|
if (!pjsip_media_type_cmp(content_type, &app_sdp, 0)) {
|
|
return 1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int check_content_disposition_in_multipart(pjsip_multipart_part *part)
|
|
{
|
|
pjsip_hdr *hdr = part->hdr.next;
|
|
static const pj_str_t str_handling_required = {"handling=required", 16};
|
|
|
|
while (hdr != &part->hdr) {
|
|
if (hdr->type == PJSIP_H_OTHER) {
|
|
pjsip_generic_string_hdr *generic_hdr = (pjsip_generic_string_hdr*)hdr;
|
|
|
|
if (!pj_stricmp2(&hdr->name, "Content-Disposition") &&
|
|
pj_stristr(&generic_hdr->hvalue, &str_handling_required) &&
|
|
!check_sdp_content_type_supported(&part->body->content_type)) {
|
|
return 1;
|
|
}
|
|
}
|
|
hdr = hdr->next;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* if there is required media we don't understand, return 1
|
|
*/
|
|
static int check_content_disposition(pjsip_rx_data *rdata)
|
|
{
|
|
pjsip_msg_body *body = rdata->msg_info.msg->body;
|
|
pjsip_ctype_hdr *ctype_hdr = rdata->msg_info.ctype;
|
|
|
|
if (body && ctype_hdr &&
|
|
!pj_stricmp2(&ctype_hdr->media.type, "multipart") &&
|
|
(!pj_stricmp2(&ctype_hdr->media.subtype, "mixed") ||
|
|
!pj_stricmp2(&ctype_hdr->media.subtype, "alternative"))) {
|
|
pjsip_multipart_part *part = pjsip_multipart_get_first_part(body);
|
|
while (part != NULL) {
|
|
if (check_content_disposition_in_multipart(part)) {
|
|
return 1;
|
|
}
|
|
part = pjsip_multipart_get_next_part(body, part);
|
|
}
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int new_invite(struct new_invite *invite)
|
|
{
|
|
pjsip_tx_data *tdata = NULL;
|
|
pjsip_timer_setting timer;
|
|
pjsip_rdata_sdp_info *sdp_info;
|
|
pjmedia_sdp_session *local = NULL;
|
|
char buffer[AST_SOCKADDR_BUFLEN];
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(invite->session));
|
|
|
|
/* From this point on, any calls to pjsip_inv_terminate have the last argument as PJ_TRUE
|
|
* so that we will be notified so we can destroy the session properly
|
|
*/
|
|
|
|
if (invite->session->inv_session->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
|
|
invite->session->inv_session->cause,
|
|
pjsip_get_status_text(invite->session->inv_session->cause)->ptr);
|
|
#ifdef HAVE_PJSIP_INV_SESSION_REF
|
|
pjsip_inv_dec_ref(invite->session->inv_session);
|
|
#endif
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Already disconnected\n");
|
|
}
|
|
|
|
switch (get_destination(invite->session, invite->rdata)) {
|
|
case SIP_GET_DEST_EXTEN_FOUND:
|
|
/* Things worked. Keep going */
|
|
break;
|
|
case SIP_GET_DEST_UNSUPPORTED_URI:
|
|
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 416, NULL, NULL, &tdata) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(invite->session, tdata);
|
|
} else {
|
|
pjsip_inv_terminate(invite->session->inv_session, 416, PJ_TRUE);
|
|
}
|
|
goto end;
|
|
case SIP_GET_DEST_EXTEN_PARTIAL:
|
|
ast_debug(1, "Call from '%s' (%s:%s) to extension '%s' - partial match\n",
|
|
ast_sorcery_object_get_id(invite->session->endpoint),
|
|
invite->rdata->tp_info.transport->type_name,
|
|
pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
|
|
invite->session->exten);
|
|
|
|
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 484, NULL, NULL, &tdata) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(invite->session, tdata);
|
|
} else {
|
|
pjsip_inv_terminate(invite->session->inv_session, 484, PJ_TRUE);
|
|
}
|
|
goto end;
|
|
case SIP_GET_DEST_EXTEN_NOT_FOUND:
|
|
default:
|
|
ast_log(LOG_NOTICE, "Call from '%s' (%s:%s) to extension '%s' rejected because extension not found in context '%s'.\n",
|
|
ast_sorcery_object_get_id(invite->session->endpoint),
|
|
invite->rdata->tp_info.transport->type_name,
|
|
pj_sockaddr_print(&invite->rdata->pkt_info.src_addr, buffer, sizeof(buffer), 3),
|
|
invite->session->exten,
|
|
invite->session->endpoint->context);
|
|
|
|
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 404, NULL, NULL, &tdata) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(invite->session, tdata);
|
|
} else {
|
|
pjsip_inv_terminate(invite->session->inv_session, 404, PJ_TRUE);
|
|
}
|
|
goto end;
|
|
};
|
|
|
|
if (check_content_disposition(invite->rdata)) {
|
|
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 415, NULL, NULL, &tdata) == PJ_SUCCESS) {
|
|
ast_sip_session_send_response(invite->session, tdata);
|
|
} else {
|
|
pjsip_inv_terminate(invite->session->inv_session, 415, PJ_TRUE);
|
|
}
|
|
goto end;
|
|
}
|
|
|
|
pjsip_timer_setting_default(&timer);
|
|
timer.min_se = invite->session->endpoint->extensions.timer.min_se;
|
|
timer.sess_expires = invite->session->endpoint->extensions.timer.sess_expires;
|
|
pjsip_timer_init_session(invite->session->inv_session, &timer);
|
|
|
|
/*
|
|
* At this point, we've verified what we can that won't take awhile,
|
|
* so let's go ahead and send a 100 Trying out to stop any
|
|
* retransmissions.
|
|
*/
|
|
if (pjsip_inv_initial_answer(invite->session->inv_session, invite->rdata, 100, NULL, NULL, &tdata) != PJ_SUCCESS) {
|
|
pjsip_inv_terminate(invite->session->inv_session, 500, PJ_TRUE);
|
|
goto end;
|
|
}
|
|
ast_sip_session_send_response(invite->session, tdata);
|
|
|
|
sdp_info = pjsip_rdata_get_sdp_info(invite->rdata);
|
|
if (sdp_info && (sdp_info->sdp_err == PJ_SUCCESS) && sdp_info->sdp) {
|
|
if (handle_incoming_sdp(invite->session, sdp_info->sdp)) {
|
|
tdata = NULL;
|
|
if (pjsip_inv_end_session(invite->session->inv_session, 488, NULL, &tdata) == PJ_SUCCESS
|
|
&& tdata) {
|
|
ast_sip_session_send_response(invite->session, tdata);
|
|
}
|
|
goto end;
|
|
}
|
|
/* We are creating a local SDP which is an answer to their offer */
|
|
local = create_local_sdp(invite->session->inv_session, invite->session, sdp_info->sdp);
|
|
} else {
|
|
/* We are creating a local SDP which is an offer */
|
|
local = create_local_sdp(invite->session->inv_session, invite->session, NULL);
|
|
}
|
|
|
|
/* If we were unable to create a local SDP terminate the session early, it won't go anywhere */
|
|
if (!local) {
|
|
tdata = NULL;
|
|
if (pjsip_inv_end_session(invite->session->inv_session, 500, NULL, &tdata) == PJ_SUCCESS
|
|
&& tdata) {
|
|
ast_sip_session_send_response(invite->session, tdata);
|
|
}
|
|
goto end;
|
|
}
|
|
|
|
pjsip_inv_set_local_sdp(invite->session->inv_session, local);
|
|
pjmedia_sdp_neg_set_prefer_remote_codec_order(invite->session->inv_session->neg, PJ_FALSE);
|
|
#ifdef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
|
|
if (!invite->session->endpoint->preferred_codec_only) {
|
|
pjmedia_sdp_neg_set_answer_multiple_codecs(invite->session->inv_session->neg, PJ_TRUE);
|
|
}
|
|
#endif
|
|
|
|
handle_incoming_request(invite->session, invite->rdata);
|
|
|
|
end:
|
|
#ifdef HAVE_PJSIP_INV_SESSION_REF
|
|
pjsip_inv_dec_ref(invite->session->inv_session);
|
|
#endif
|
|
SCOPE_EXIT_RTN_VALUE(0, "Handled. Active: %s Pending: %s\n",
|
|
ast_str_tmp(256, ast_stream_topology_to_str(invite->session->active_media_state->topology, &STR_TMP)),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(invite->session->pending_media_state->topology, &STR_TMP))
|
|
);
|
|
}
|
|
|
|
static void handle_new_invite_request(pjsip_rx_data *rdata)
|
|
{
|
|
RAII_VAR(struct ast_sip_endpoint *, endpoint,
|
|
ast_pjsip_rdata_get_endpoint(rdata), ao2_cleanup);
|
|
pjsip_tx_data *tdata = NULL;
|
|
pjsip_inv_session *inv_session = NULL;
|
|
struct ast_sip_session *session;
|
|
struct new_invite invite;
|
|
char *req_uri = TRACE_ATLEAST(1) ? ast_alloca(256) : "";
|
|
int res = TRACE_ATLEAST(1) ? pjsip_uri_print(PJSIP_URI_IN_REQ_URI, rdata->msg_info.msg->line.req.uri, req_uri, 256) : 0;
|
|
SCOPE_ENTER(1, "Request: %s\n", res ? req_uri : "");
|
|
|
|
ast_assert(endpoint != NULL);
|
|
|
|
inv_session = pre_session_setup(rdata, endpoint);
|
|
if (!inv_session) {
|
|
/* pre_session_setup() returns a response on failure */
|
|
SCOPE_EXIT_RTN("Failure in pre session setup\n");
|
|
}
|
|
|
|
#ifdef HAVE_PJSIP_INV_SESSION_REF
|
|
if (pjsip_inv_add_ref(inv_session) != PJ_SUCCESS) {
|
|
ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
|
|
if (inv_session->state != PJSIP_INV_STATE_DISCONNECTED) {
|
|
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
|
|
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
|
|
} else {
|
|
pjsip_inv_send_msg(inv_session, tdata);
|
|
}
|
|
}
|
|
SCOPE_EXIT_RTN("Couldn't add invite session reference\n");
|
|
}
|
|
#endif
|
|
|
|
session = ast_sip_session_alloc(endpoint, NULL, inv_session, rdata);
|
|
if (!session) {
|
|
if (pjsip_inv_initial_answer(inv_session, rdata, 500, NULL, NULL, &tdata) == PJ_SUCCESS) {
|
|
pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
|
|
} else {
|
|
pjsip_inv_send_msg(inv_session, tdata);
|
|
}
|
|
#ifdef HAVE_PJSIP_INV_SESSION_REF
|
|
pjsip_inv_dec_ref(inv_session);
|
|
#endif
|
|
SCOPE_EXIT_RTN("Couldn't create session\n");
|
|
}
|
|
session->call_direction = AST_SIP_SESSION_INCOMING_CALL;
|
|
|
|
/*
|
|
* The current thread is supposed be the session serializer to prevent
|
|
* any initial INVITE retransmissions from trying to setup the same
|
|
* call again.
|
|
*/
|
|
ast_assert(ast_taskprocessor_is_task(session->serializer));
|
|
|
|
invite.session = session;
|
|
invite.rdata = rdata;
|
|
new_invite(&invite);
|
|
|
|
SCOPE_EXIT("Request: %s Session: %s\n", req_uri, ast_sip_session_get_name(session));
|
|
ao2_ref(session, -1);
|
|
}
|
|
|
|
static pj_bool_t does_method_match(const pj_str_t *message_method, const char *supplement_method)
|
|
{
|
|
pj_str_t method;
|
|
|
|
if (ast_strlen_zero(supplement_method)) {
|
|
return PJ_TRUE;
|
|
}
|
|
|
|
pj_cstr(&method, supplement_method);
|
|
|
|
return pj_stristr(&method, message_method) ? PJ_TRUE : PJ_FALSE;
|
|
}
|
|
|
|
static pj_bool_t has_supplement(const struct ast_sip_session *session, const pjsip_rx_data *rdata)
|
|
{
|
|
struct ast_sip_session_supplement *supplement;
|
|
struct pjsip_method *method = &rdata->msg_info.msg->line.req.method;
|
|
|
|
if (!session) {
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
|
|
if (does_method_match(&method->name, supplement->method)) {
|
|
return PJ_TRUE;
|
|
}
|
|
}
|
|
return PJ_FALSE;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* Added for debugging purposes
|
|
*/
|
|
static void session_on_tsx_state(pjsip_transaction *tsx, pjsip_event *e)
|
|
{
|
|
|
|
pjsip_dialog *dlg = pjsip_tsx_get_dlg(tsx);
|
|
pjsip_inv_session *inv_session = (dlg ? pjsip_dlg_get_inv_session(dlg) : NULL);
|
|
struct ast_sip_session *session = (inv_session ? inv_session->mod_data[session_module.id] : NULL);
|
|
SCOPE_ENTER(1, "%s TSX State: %s Inv State: %s\n", ast_sip_session_get_name(session),
|
|
pjsip_tsx_state_str(tsx->state), inv_session ? pjsip_inv_state_name(inv_session->state) : "unknown");
|
|
|
|
if (session) {
|
|
ast_trace(2, "Topology: Pending: %s Active: %s\n",
|
|
ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(session->active_media_state->topology, &STR_TMP)));
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* Added for debugging purposes
|
|
*/
|
|
static pj_bool_t session_on_rx_response(pjsip_rx_data *rdata)
|
|
{
|
|
|
|
struct pjsip_status_line status = rdata->msg_info.msg->line.status;
|
|
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
|
|
pjsip_inv_session *inv_session = dlg ? pjsip_dlg_get_inv_session(dlg) : NULL;
|
|
struct ast_sip_session *session = (inv_session ? inv_session->mod_data[session_module.id] : NULL);
|
|
SCOPE_ENTER(1, "%s Method: %.*s Status: %d\n", ast_sip_session_get_name(session),
|
|
(int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
|
|
|
|
SCOPE_EXIT_RTN_VALUE(PJ_FALSE);
|
|
}
|
|
|
|
/*!
|
|
* \brief Called when a new SIP request comes into PJSIP
|
|
*
|
|
* This function is called under two circumstances
|
|
* 1) An out-of-dialog request is received by PJSIP
|
|
* 2) An in-dialog request that the inv_session layer does not
|
|
* handle is received (such as an in-dialog INFO)
|
|
*
|
|
* Except for INVITEs, there is very little we actually do in this function
|
|
* 1) For requests we don't handle, we return PJ_FALSE
|
|
* 2) For new INVITEs, handle them now to prevent retransmissions from
|
|
* trying to setup the same call again.
|
|
* 3) For in-dialog requests we handle, we process them in the
|
|
* .on_state_changed = session_inv_on_state_changed or
|
|
* .on_tsx_state_changed = session_inv_on_tsx_state_changed
|
|
* callbacks instead.
|
|
*/
|
|
static pj_bool_t session_on_rx_request(pjsip_rx_data *rdata)
|
|
{
|
|
pj_status_t handled = PJ_FALSE;
|
|
struct pjsip_request_line req = rdata->msg_info.msg->line.req;
|
|
pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata);
|
|
pjsip_inv_session *inv_session = (dlg ? pjsip_dlg_get_inv_session(dlg) : NULL);
|
|
struct ast_sip_session *session = (inv_session ? inv_session->mod_data[session_module.id] : NULL);
|
|
char *req_uri = TRACE_ATLEAST(1) ? ast_alloca(256) : "";
|
|
int res = TRACE_ATLEAST(1) ? pjsip_uri_print(PJSIP_URI_IN_REQ_URI, rdata->msg_info.msg->line.req.uri, req_uri, 256) : 0;
|
|
SCOPE_ENTER(1, "%s Request: %.*s %s\n", ast_sip_session_get_name(session),
|
|
(int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name), res ? req_uri : "");
|
|
|
|
switch (req.method.id) {
|
|
case PJSIP_INVITE_METHOD:
|
|
if (dlg) {
|
|
ast_log(LOG_WARNING, "on_rx_request called for INVITE in mid-dialog?\n");
|
|
break;
|
|
}
|
|
handled = PJ_TRUE;
|
|
handle_new_invite_request(rdata);
|
|
break;
|
|
default:
|
|
/* Handle other in-dialog methods if their supplements have been registered */
|
|
handled = dlg && (inv_session = pjsip_dlg_get_inv_session(dlg)) &&
|
|
has_supplement(inv_session->mod_data[session_module.id], rdata);
|
|
break;
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(handled, "%s Handled request %.*s %s ? %s\n", ast_sip_session_get_name(session),
|
|
(int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name), req_uri,
|
|
handled == PJ_TRUE ? "yes" : "no");
|
|
}
|
|
|
|
static void resend_reinvite(pj_timer_heap_t *timer, pj_timer_entry *entry)
|
|
{
|
|
struct ast_sip_session *session = entry->user_data;
|
|
|
|
ast_debug(3, "Endpoint '%s(%s)' re-INVITE collision timer expired.\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "");
|
|
|
|
if (AST_LIST_EMPTY(&session->delayed_requests)) {
|
|
/* No delayed request pending, so just return */
|
|
ao2_ref(session, -1);
|
|
return;
|
|
}
|
|
if (ast_sip_push_task(session->serializer, invite_collision_timeout, session)) {
|
|
/*
|
|
* Uh oh. We now have nothing in the foreseeable future
|
|
* to trigger sending the delayed requests.
|
|
*/
|
|
ao2_ref(session, -1);
|
|
}
|
|
}
|
|
|
|
static void reschedule_reinvite(struct ast_sip_session *session, ast_sip_session_response_cb on_response)
|
|
{
|
|
pjsip_inv_session *inv = session->inv_session;
|
|
pj_time_val tv;
|
|
|
|
ast_debug(3, "Endpoint '%s(%s)' re-INVITE collision.\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "");
|
|
if (delay_request(session, NULL, NULL, on_response, 1, DELAYED_METHOD_INVITE, NULL)) {
|
|
return;
|
|
}
|
|
if (pj_timer_entry_running(&session->rescheduled_reinvite)) {
|
|
/* Timer already running. Something weird is going on. */
|
|
ast_debug(1, "Endpoint '%s(%s)' re-INVITE collision while timer running!!!\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "");
|
|
return;
|
|
}
|
|
|
|
tv.sec = 0;
|
|
if (inv->role == PJSIP_ROLE_UAC) {
|
|
tv.msec = 2100 + ast_random() % 2000;
|
|
} else {
|
|
tv.msec = ast_random() % 2000;
|
|
}
|
|
pj_timer_entry_init(&session->rescheduled_reinvite, 0, session, resend_reinvite);
|
|
|
|
ao2_ref(session, +1);
|
|
if (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(),
|
|
&session->rescheduled_reinvite, &tv) != PJ_SUCCESS) {
|
|
ao2_ref(session, -1);
|
|
}
|
|
}
|
|
|
|
static void __print_debug_details(const char *function, pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
|
|
{
|
|
int id = session_module.id;
|
|
struct ast_sip_session *session = NULL;
|
|
|
|
if (!DEBUG_ATLEAST(5)) {
|
|
/* Debug not spamy enough */
|
|
return;
|
|
}
|
|
|
|
ast_log(LOG_DEBUG, "Function %s called on event %s\n",
|
|
function, pjsip_event_str(e->type));
|
|
if (!inv) {
|
|
ast_log(LOG_DEBUG, "Transaction %p does not belong to an inv_session?\n", tsx);
|
|
ast_log(LOG_DEBUG, "The transaction state is %s\n",
|
|
pjsip_tsx_state_str(tsx->state));
|
|
return;
|
|
}
|
|
if (id > -1) {
|
|
session = inv->mod_data[session_module.id];
|
|
}
|
|
if (!session) {
|
|
ast_log(LOG_DEBUG, "inv_session %p has no ast session\n", inv);
|
|
} else {
|
|
ast_log(LOG_DEBUG, "The state change pertains to the endpoint '%s(%s)'\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "");
|
|
}
|
|
if (inv->invite_tsx) {
|
|
ast_log(LOG_DEBUG, "The inv session still has an invite_tsx (%p)\n",
|
|
inv->invite_tsx);
|
|
} else {
|
|
ast_log(LOG_DEBUG, "The inv session does NOT have an invite_tsx\n");
|
|
}
|
|
if (tsx) {
|
|
ast_log(LOG_DEBUG, "The %s %.*s transaction involved in this state change is %p\n",
|
|
pjsip_role_name(tsx->role),
|
|
(int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
|
|
tsx);
|
|
ast_log(LOG_DEBUG, "The current transaction state is %s\n",
|
|
pjsip_tsx_state_str(tsx->state));
|
|
ast_log(LOG_DEBUG, "The transaction state change event is %s\n",
|
|
pjsip_event_str(e->body.tsx_state.type));
|
|
} else {
|
|
ast_log(LOG_DEBUG, "There is no transaction involved in this state change\n");
|
|
}
|
|
ast_log(LOG_DEBUG, "The current inv state is %s\n", pjsip_inv_state_name(inv->state));
|
|
}
|
|
|
|
#define print_debug_details(inv, tsx, e) __print_debug_details(__PRETTY_FUNCTION__, (inv), (tsx), (e))
|
|
|
|
static void handle_incoming_request(struct ast_sip_session *session, pjsip_rx_data *rdata)
|
|
{
|
|
struct ast_sip_session_supplement *supplement;
|
|
struct pjsip_request_line req = rdata->msg_info.msg->line.req;
|
|
SCOPE_ENTER(1, "%s Method: %.*s\n", ast_sip_session_get_name(session),
|
|
(int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));
|
|
|
|
ast_debug(3, "Method is %.*s\n", (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));
|
|
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
|
|
if (supplement->incoming_request && does_method_match(&req.method.name, supplement->method)) {
|
|
if (supplement->incoming_request(session, rdata)) {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static void handle_session_begin(struct ast_sip_session *session)
|
|
{
|
|
struct ast_sip_session_supplement *iter;
|
|
|
|
AST_LIST_TRAVERSE(&session->supplements, iter, next) {
|
|
if (iter->session_begin) {
|
|
iter->session_begin(session);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void handle_session_destroy(struct ast_sip_session *session)
|
|
{
|
|
struct ast_sip_session_supplement *iter;
|
|
|
|
AST_LIST_TRAVERSE(&session->supplements, iter, next) {
|
|
if (iter->session_destroy) {
|
|
iter->session_destroy(session);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void handle_session_end(struct ast_sip_session *session)
|
|
{
|
|
struct ast_sip_session_supplement *iter;
|
|
|
|
/* Session is dead. Notify the supplements. */
|
|
AST_LIST_TRAVERSE(&session->supplements, iter, next) {
|
|
if (iter->session_end) {
|
|
iter->session_end(session);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void handle_incoming_response(struct ast_sip_session *session, pjsip_rx_data *rdata,
|
|
enum ast_sip_session_response_priority response_priority)
|
|
{
|
|
struct ast_sip_session_supplement *supplement;
|
|
struct pjsip_status_line status = rdata->msg_info.msg->line.status;
|
|
SCOPE_ENTER(1, "%s Method: %.*s Status: %d Priority %s\n", ast_sip_session_get_name(session),
|
|
(int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr,
|
|
rdata->msg_info.msg->line.status.code,
|
|
response_priority == AST_SIP_SESSION_AFTER_MEDIA ? "after" : "before");
|
|
|
|
ast_debug(3, "Response is %d %.*s\n", status.code, (int) pj_strlen(&status.reason),
|
|
pj_strbuf(&status.reason));
|
|
|
|
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
|
|
if (!(supplement->response_priority & response_priority)) {
|
|
continue;
|
|
}
|
|
if (supplement->incoming_response && does_method_match(&rdata->msg_info.cseq->method.name, supplement->method)) {
|
|
supplement->incoming_response(session, rdata);
|
|
}
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static int handle_incoming(struct ast_sip_session *session, pjsip_rx_data *rdata,
|
|
enum ast_sip_session_response_priority response_priority)
|
|
{
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
ast_debug(3, "Received %s\n", rdata->msg_info.msg->type == PJSIP_REQUEST_MSG ?
|
|
"request" : "response");
|
|
|
|
if (rdata->msg_info.msg->type == PJSIP_REQUEST_MSG) {
|
|
handle_incoming_request(session, rdata);
|
|
} else {
|
|
handle_incoming_response(session, rdata, response_priority);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN_VALUE(0);
|
|
}
|
|
|
|
static void handle_outgoing_request(struct ast_sip_session *session, pjsip_tx_data *tdata)
|
|
{
|
|
struct ast_sip_session_supplement *supplement;
|
|
struct pjsip_request_line req = tdata->msg->line.req;
|
|
SCOPE_ENTER(1, "%s Method: %.*s\n", ast_sip_session_get_name(session),
|
|
(int) pj_strlen(&req.method.name),
|
|
pj_strbuf(&req.method.name));
|
|
|
|
ast_debug(3, "Method is %.*s\n", (int) pj_strlen(&req.method.name), pj_strbuf(&req.method.name));
|
|
|
|
ast_sip_message_apply_transport(session->endpoint->transport, tdata);
|
|
|
|
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
|
|
if (supplement->outgoing_request && does_method_match(&req.method.name, supplement->method)) {
|
|
supplement->outgoing_request(session, tdata);
|
|
}
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static void handle_outgoing_response(struct ast_sip_session *session, pjsip_tx_data *tdata)
|
|
{
|
|
struct ast_sip_session_supplement *supplement;
|
|
struct pjsip_status_line status = tdata->msg->line.status;
|
|
pjsip_cseq_hdr *cseq = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CSEQ, NULL);
|
|
SCOPE_ENTER(1, "%s Method is %.*s, Response is %d %.*s\n", ast_sip_session_get_name(session),
|
|
(int) pj_strlen(&cseq->method.name),
|
|
pj_strbuf(&cseq->method.name), status.code, (int) pj_strlen(&status.reason),
|
|
pj_strbuf(&status.reason));
|
|
|
|
if (!cseq) {
|
|
ast_log(LOG_ERROR, "Cannot send response due to missing sequence header");
|
|
SCOPE_EXIT_RTN("Missing cseq\n");
|
|
}
|
|
|
|
ast_debug(3, "Method is %.*s, Response is %d %.*s\n", (int) pj_strlen(&cseq->method.name),
|
|
pj_strbuf(&cseq->method.name), status.code, (int) pj_strlen(&status.reason),
|
|
pj_strbuf(&status.reason));
|
|
|
|
ast_sip_message_apply_transport(session->endpoint->transport, tdata);
|
|
|
|
AST_LIST_TRAVERSE(&session->supplements, supplement, next) {
|
|
if (supplement->outgoing_response && does_method_match(&cseq->method.name, supplement->method)) {
|
|
supplement->outgoing_response(session, tdata);
|
|
}
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static int session_end(void *vsession)
|
|
{
|
|
struct ast_sip_session *session = vsession;
|
|
|
|
/* Stop the scheduled termination */
|
|
sip_session_defer_termination_stop_timer(session);
|
|
|
|
/* Session is dead. Notify the supplements. */
|
|
handle_session_end(session);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Complete ending session activities.
|
|
* \since 13.5.0
|
|
*
|
|
* \param vsession Which session to complete stopping.
|
|
*
|
|
* \retval 0 on success.
|
|
* \retval -1 on error.
|
|
*/
|
|
static int session_end_completion(void *vsession)
|
|
{
|
|
struct ast_sip_session *session = vsession;
|
|
|
|
ast_sip_dialog_set_serializer(session->inv_session->dlg, NULL);
|
|
ast_sip_dialog_set_endpoint(session->inv_session->dlg, NULL);
|
|
|
|
/* Now we can release the ref that was held by session->inv_session */
|
|
ao2_cleanup(session);
|
|
return 0;
|
|
}
|
|
|
|
static int check_request_status(pjsip_inv_session *inv, pjsip_event *e)
|
|
{
|
|
struct ast_sip_session *session = inv->mod_data[session_module.id];
|
|
pjsip_transaction *tsx = e->body.tsx_state.tsx;
|
|
|
|
if (tsx->status_code != 503 && tsx->status_code != 408) {
|
|
return 0;
|
|
}
|
|
|
|
if (!ast_sip_failover_request(tsx->last_tx)) {
|
|
return 0;
|
|
}
|
|
|
|
pjsip_inv_uac_restart(inv, PJ_FALSE);
|
|
/*
|
|
* Bump the ref since it will be on a new transaction and
|
|
* we don't want it to go away along with the old transaction.
|
|
*/
|
|
pjsip_tx_data_add_ref(tsx->last_tx);
|
|
ast_sip_session_send_request(session, tsx->last_tx);
|
|
return 1;
|
|
}
|
|
|
|
static void handle_incoming_before_media(pjsip_inv_session *inv,
|
|
struct ast_sip_session *session, pjsip_rx_data *rdata)
|
|
{
|
|
pjsip_msg *msg;
|
|
SCOPE_ENTER(1, "%s\n", ast_sip_session_get_name(session));
|
|
|
|
handle_incoming(session, rdata, AST_SIP_SESSION_BEFORE_MEDIA);
|
|
msg = rdata->msg_info.msg;
|
|
if (msg->type == PJSIP_REQUEST_MSG
|
|
&& msg->line.req.method.id == PJSIP_ACK_METHOD
|
|
&& pjmedia_sdp_neg_get_state(inv->neg) != PJMEDIA_SDP_NEG_STATE_DONE) {
|
|
pjsip_tx_data *tdata;
|
|
|
|
/*
|
|
* SDP negotiation failed on an incoming call that delayed
|
|
* negotiation and then gave us an invalid SDP answer. We
|
|
* need to send a BYE to end the call because of the invalid
|
|
* SDP answer.
|
|
*/
|
|
ast_debug(1,
|
|
"Endpoint '%s(%s)': Ending session due to incomplete SDP negotiation. %s\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_rx_data_get_info(rdata));
|
|
if (pjsip_inv_end_session(inv, 400, NULL, &tdata) == PJ_SUCCESS
|
|
&& tdata) {
|
|
ast_sip_session_send_request(session, tdata);
|
|
}
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static void session_inv_on_state_changed(pjsip_inv_session *inv, pjsip_event *e)
|
|
{
|
|
pjsip_event_id_e type;
|
|
struct ast_sip_session *session = inv->mod_data[session_module.id];
|
|
SCOPE_ENTER(1, "%s Event: %s Inv State: %s\n", ast_sip_session_get_name(session),
|
|
pjsip_event_str(e->type), pjsip_inv_state_name(inv->state));
|
|
|
|
if (ast_shutdown_final()) {
|
|
SCOPE_EXIT_RTN("Shutting down\n");
|
|
}
|
|
|
|
if (e) {
|
|
print_debug_details(inv, NULL, e);
|
|
type = e->type;
|
|
} else {
|
|
type = PJSIP_EVENT_UNKNOWN;
|
|
}
|
|
|
|
session = inv->mod_data[session_module.id];
|
|
if (!session) {
|
|
SCOPE_EXIT_RTN("No session\n");
|
|
}
|
|
|
|
switch(type) {
|
|
case PJSIP_EVENT_TX_MSG:
|
|
break;
|
|
case PJSIP_EVENT_RX_MSG:
|
|
handle_incoming_before_media(inv, session, e->body.rx_msg.rdata);
|
|
break;
|
|
case PJSIP_EVENT_TSX_STATE:
|
|
ast_debug(3, "Source of transaction state change is %s\n", pjsip_event_str(e->body.tsx_state.type));
|
|
/* Transaction state changes are prompted by some other underlying event. */
|
|
switch(e->body.tsx_state.type) {
|
|
case PJSIP_EVENT_TX_MSG:
|
|
break;
|
|
case PJSIP_EVENT_RX_MSG:
|
|
if (!check_request_status(inv, e)) {
|
|
handle_incoming_before_media(inv, session, e->body.tsx_state.src.rdata);
|
|
}
|
|
break;
|
|
case PJSIP_EVENT_TRANSPORT_ERROR:
|
|
case PJSIP_EVENT_TIMER:
|
|
/*
|
|
* Check the request status on transport error or timeout. A transport
|
|
* error can occur when a TCP socket closes and that can be the result
|
|
* of a 503. Also we may need to failover on a timeout (408).
|
|
*/
|
|
check_request_status(inv, e);
|
|
break;
|
|
case PJSIP_EVENT_USER:
|
|
case PJSIP_EVENT_UNKNOWN:
|
|
case PJSIP_EVENT_TSX_STATE:
|
|
/* Inception? */
|
|
break;
|
|
}
|
|
break;
|
|
case PJSIP_EVENT_TRANSPORT_ERROR:
|
|
case PJSIP_EVENT_TIMER:
|
|
case PJSIP_EVENT_UNKNOWN:
|
|
case PJSIP_EVENT_USER:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
if (session->defer_end) {
|
|
ast_debug(3, "Deferring session (%p) end\n", session);
|
|
session->ended_while_deferred = 1;
|
|
SCOPE_EXIT_RTN("Deferring\n");
|
|
}
|
|
|
|
if (ast_sip_push_task(session->serializer, session_end, session)) {
|
|
/* Do it anyway even though this is not the right thread. */
|
|
session_end(session);
|
|
}
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static void session_inv_on_new_session(pjsip_inv_session *inv, pjsip_event *e)
|
|
{
|
|
/* XXX STUB */
|
|
}
|
|
|
|
static int session_end_if_disconnected(int id, pjsip_inv_session *inv)
|
|
{
|
|
struct ast_sip_session *session;
|
|
|
|
if (inv->state != PJSIP_INV_STATE_DISCONNECTED) {
|
|
return 0;
|
|
}
|
|
|
|
/*
|
|
* We are locking because ast_sip_dialog_get_session() needs
|
|
* the dialog locked to get the session by other threads.
|
|
*/
|
|
pjsip_dlg_inc_lock(inv->dlg);
|
|
session = inv->mod_data[id];
|
|
inv->mod_data[id] = NULL;
|
|
pjsip_dlg_dec_lock(inv->dlg);
|
|
|
|
/*
|
|
* Pass the session ref held by session->inv_session to
|
|
* session_end_completion().
|
|
*/
|
|
if (session
|
|
&& ast_sip_push_task(session->serializer, session_end_completion, session)) {
|
|
/* Do it anyway even though this is not the right thread. */
|
|
session_end_completion(session);
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
static void session_inv_on_tsx_state_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e)
|
|
{
|
|
ast_sip_session_response_cb cb;
|
|
int id = session_module.id;
|
|
pjsip_tx_data *tdata;
|
|
struct ast_sip_session *session = inv->mod_data[session_module.id];
|
|
SCOPE_ENTER(1, "%s TSX State: %s Inv State: %s\n", ast_sip_session_get_name(session),
|
|
pjsip_tsx_state_str(tsx->state), pjsip_inv_state_name(inv->state));
|
|
|
|
if (ast_shutdown_final()) {
|
|
SCOPE_EXIT_RTN("Shutting down\n");
|
|
}
|
|
|
|
session = inv->mod_data[id];
|
|
|
|
print_debug_details(inv, tsx, e);
|
|
if (!session) {
|
|
/* The session has ended. Ignore the transaction change. */
|
|
SCOPE_EXIT_RTN("Session ended\n");
|
|
}
|
|
|
|
/*
|
|
* If the session is disconnected really nothing else to do unless currently transacting
|
|
* a BYE. If a BYE then hold off destruction until the transaction timeout occurs. This
|
|
* has to be done for BYEs because sometimes the dialog can be in a disconnected
|
|
* state but the BYE request transaction has not yet completed.
|
|
*/
|
|
if (tsx->method.id != PJSIP_BYE_METHOD && session_end_if_disconnected(id, inv)) {
|
|
SCOPE_EXIT_RTN("Disconnected\n");
|
|
}
|
|
|
|
switch (e->body.tsx_state.type) {
|
|
case PJSIP_EVENT_TX_MSG:
|
|
/* When we create an outgoing request, we do not have access to the transaction that
|
|
* is created. Instead, We have to place transaction-specific data in the tdata. Here,
|
|
* we transfer the data into the transaction. This way, when we receive a response, we
|
|
* can dig this data out again
|
|
*/
|
|
tsx->mod_data[id] = e->body.tsx_state.src.tdata->mod_data[id];
|
|
break;
|
|
case PJSIP_EVENT_RX_MSG:
|
|
cb = ast_sip_mod_data_get(tsx->mod_data, id, MOD_DATA_ON_RESPONSE);
|
|
/* As the PJSIP invite session implementation responds with a 200 OK before we have a
|
|
* chance to be invoked session supplements for BYE requests actually end up executing
|
|
* in the invite session state callback as well. To prevent session supplements from
|
|
* running on the BYE request again we explicitly squash invocation of them here.
|
|
*/
|
|
if ((e->body.tsx_state.src.rdata->msg_info.msg->type != PJSIP_REQUEST_MSG) ||
|
|
(tsx->method.id != PJSIP_BYE_METHOD)) {
|
|
handle_incoming(session, e->body.tsx_state.src.rdata,
|
|
AST_SIP_SESSION_AFTER_MEDIA);
|
|
}
|
|
if (tsx->method.id == PJSIP_INVITE_METHOD) {
|
|
if (tsx->role == PJSIP_ROLE_UAC) {
|
|
if (tsx->state == PJSIP_TSX_STATE_COMPLETED) {
|
|
/* This means we got a non 2XX final response to our outgoing INVITE */
|
|
if (tsx->status_code == PJSIP_SC_REQUEST_PENDING) {
|
|
reschedule_reinvite(session, cb);
|
|
SCOPE_EXIT_RTN("Non 2XX final response\n");
|
|
}
|
|
if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
|
|
ast_debug(1, "reINVITE received final response code %d\n",
|
|
tsx->status_code);
|
|
if ((tsx->status_code == 401 || tsx->status_code == 407)
|
|
&& !ast_sip_create_request_with_auth(
|
|
&session->endpoint->outbound_auths,
|
|
e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
|
|
/* Send authed reINVITE */
|
|
ast_sip_session_send_request_with_cb(session, tdata, cb);
|
|
SCOPE_EXIT_RTN("Sending authed reinvite\n");
|
|
}
|
|
/* Per RFC3261 14.1 a response to a re-INVITE should only terminate
|
|
* the dialog if a 481 or 408 occurs. All other responses should leave
|
|
* the dialog untouched.
|
|
*/
|
|
if (tsx->status_code == 481 || tsx->status_code == 408) {
|
|
if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS
|
|
&& tdata) {
|
|
ast_sip_session_send_request(session, tdata);
|
|
}
|
|
}
|
|
}
|
|
} else if (tsx->state == PJSIP_TSX_STATE_TERMINATED) {
|
|
if (!inv->cancelling
|
|
&& inv->role == PJSIP_ROLE_UAC
|
|
&& inv->state == PJSIP_INV_STATE_CONFIRMED
|
|
&& pjmedia_sdp_neg_was_answer_remote(inv->neg)
|
|
&& pjmedia_sdp_neg_get_state(inv->neg) == PJMEDIA_SDP_NEG_STATE_DONE
|
|
&& (session->channel && ast_channel_hangupcause(session->channel) == AST_CAUSE_BEARERCAPABILITY_NOTAVAIL)
|
|
) {
|
|
/*
|
|
* We didn't send a CANCEL but the UAS sent us the 200 OK with an invalid or unacceptable codec SDP.
|
|
* In this case the SDP negotiation is incomplete and PJPROJECT has already sent the ACK.
|
|
* So, we send the BYE with 503 status code here. And the actual hangup cause code is already set
|
|
* to AST_CAUSE_BEARERCAPABILITY_NOTAVAIL by the session_inv_on_media_update(), setting the 503
|
|
* status code doesn't affect to hangup cause code.
|
|
*/
|
|
ast_debug(1, "Endpoint '%s(%s)': Ending session due to 200 OK with incomplete SDP negotiation. %s\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
|
|
pjsip_inv_end_session(session->inv_session, 503, NULL, &tdata);
|
|
SCOPE_EXIT_RTN("Incomplete SDP negotiation\n");
|
|
}
|
|
|
|
if (inv->cancelling && tsx->status_code == PJSIP_SC_OK) {
|
|
int sdp_negotiation_done =
|
|
pjmedia_sdp_neg_get_state(inv->neg) == PJMEDIA_SDP_NEG_STATE_DONE;
|
|
|
|
/*
|
|
* We can get here for the following reasons.
|
|
*
|
|
* 1) The race condition detailed in RFC5407 section 3.1.2.
|
|
* We sent a CANCEL at the same time that the UAS sent us a
|
|
* 200 OK with a valid SDP for the original INVITE. As a
|
|
* result, we have now received a 200 OK for a cancelled
|
|
* call and the SDP negotiation is complete. We need to
|
|
* immediately send a BYE to end the dialog.
|
|
*
|
|
* 2) We sent a CANCEL and hit the race condition but the
|
|
* UAS sent us an invalid SDP with the 200 OK. In this case
|
|
* the SDP negotiation is incomplete and PJPROJECT has
|
|
* already sent the BYE for us because of the invalid SDP.
|
|
*/
|
|
ast_test_suite_event_notify("PJSIP_SESSION_CANCELED",
|
|
"Endpoint: %s\r\n"
|
|
"Channel: %s\r\n"
|
|
"Message: %s\r\n"
|
|
"SDP: %s",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_rx_data_get_info(e->body.tsx_state.src.rdata),
|
|
sdp_negotiation_done ? "complete" : "incomplete");
|
|
if (!sdp_negotiation_done) {
|
|
ast_debug(1, "Endpoint '%s(%s)': Incomplete SDP negotiation cancelled session. %s\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
|
|
} else if (pjsip_inv_end_session(inv, 500, NULL, &tdata) == PJ_SUCCESS
|
|
&& tdata) {
|
|
ast_debug(1, "Endpoint '%s(%s)': Ending session due to RFC5407 race condition. %s\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_rx_data_get_info(e->body.tsx_state.src.rdata));
|
|
ast_sip_session_send_request(session, tdata);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
} else {
|
|
/* All other methods */
|
|
if (tsx->role == PJSIP_ROLE_UAC) {
|
|
if (tsx->state == PJSIP_TSX_STATE_COMPLETED) {
|
|
/* This means we got a final response to our outgoing method */
|
|
ast_debug(1, "%.*s received final response code %d\n",
|
|
(int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name),
|
|
tsx->status_code);
|
|
if ((tsx->status_code == 401 || tsx->status_code == 407)
|
|
&& !ast_sip_create_request_with_auth(
|
|
&session->endpoint->outbound_auths,
|
|
e->body.tsx_state.src.rdata, tsx->last_tx, &tdata)) {
|
|
/* Send authed version of the method */
|
|
ast_sip_session_send_request_with_cb(session, tdata, cb);
|
|
SCOPE_EXIT_RTN("Sending authed %.*s\n",
|
|
(int) pj_strlen(&tsx->method.name), pj_strbuf(&tsx->method.name));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
if (cb) {
|
|
cb(session, e->body.tsx_state.src.rdata);
|
|
}
|
|
break;
|
|
case PJSIP_EVENT_TRANSPORT_ERROR:
|
|
case PJSIP_EVENT_TIMER:
|
|
/*
|
|
* The timer event is run by the pjsip monitor thread and not
|
|
* by the session serializer.
|
|
*/
|
|
if (session_end_if_disconnected(id, inv)) {
|
|
SCOPE_EXIT_RTN("Disconnected\n");
|
|
}
|
|
break;
|
|
case PJSIP_EVENT_USER:
|
|
case PJSIP_EVENT_UNKNOWN:
|
|
case PJSIP_EVENT_TSX_STATE:
|
|
/* Inception? */
|
|
break;
|
|
}
|
|
|
|
if (AST_LIST_EMPTY(&session->delayed_requests)) {
|
|
/* No delayed request pending, so just return */
|
|
SCOPE_EXIT_RTN("Nothing delayed\n");
|
|
}
|
|
|
|
if (tsx->method.id == PJSIP_INVITE_METHOD) {
|
|
if (tsx->state == PJSIP_TSX_STATE_PROCEEDING) {
|
|
ast_debug(3, "Endpoint '%s(%s)' INVITE delay check. tsx-state:%s\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_tsx_state_str(tsx->state));
|
|
check_delayed_requests(session, invite_proceeding);
|
|
} else if (tsx->state == PJSIP_TSX_STATE_TERMINATED) {
|
|
/*
|
|
* Terminated INVITE transactions always should result in
|
|
* queuing delayed requests, no matter what event caused
|
|
* the transaction to terminate.
|
|
*/
|
|
ast_debug(3, "Endpoint '%s(%s)' INVITE delay check. tsx-state:%s\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_tsx_state_str(tsx->state));
|
|
check_delayed_requests(session, invite_terminated);
|
|
}
|
|
} else if (tsx->role == PJSIP_ROLE_UAC
|
|
&& tsx->state == PJSIP_TSX_STATE_COMPLETED
|
|
&& !pj_strcmp2(&tsx->method.name, "UPDATE")) {
|
|
ast_debug(3, "Endpoint '%s(%s)' UPDATE delay check. tsx-state:%s\n",
|
|
ast_sorcery_object_get_id(session->endpoint),
|
|
session->channel ? ast_channel_name(session->channel) : "",
|
|
pjsip_tsx_state_str(tsx->state));
|
|
check_delayed_requests(session, update_completed);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static int add_sdp_streams(struct ast_sip_session_media *session_media,
|
|
struct ast_sip_session *session, pjmedia_sdp_session *answer,
|
|
const struct pjmedia_sdp_session *remote,
|
|
struct ast_stream *stream)
|
|
{
|
|
struct ast_sip_session_sdp_handler *handler = session_media->handler;
|
|
RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
|
|
int res = 0;
|
|
SCOPE_ENTER(1, "%s Stream: %s\n", ast_sip_session_get_name(session),
|
|
ast_str_tmp(128, ast_stream_to_str(stream, &STR_TMP)));
|
|
|
|
if (handler) {
|
|
/* if an already assigned handler reports a catastrophic error, fail */
|
|
res = handler->create_outgoing_sdp_stream(session, session_media, answer, remote, stream);
|
|
if (res < 0) {
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Coudn't create sdp stream\n");
|
|
}
|
|
SCOPE_EXIT_RTN_VALUE(0, "Had handler\n");
|
|
}
|
|
|
|
handler_list = ao2_find(sdp_handlers, ast_codec_media_type2str(session_media->type), OBJ_KEY);
|
|
if (!handler_list) {
|
|
SCOPE_EXIT_RTN_VALUE(0, "No handlers\n");
|
|
}
|
|
|
|
/* no handler for this stream type and we have a list to search */
|
|
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
|
|
if (handler == session_media->handler) {
|
|
continue;
|
|
}
|
|
res = handler->create_outgoing_sdp_stream(session, session_media, answer, remote, stream);
|
|
if (res < 0) {
|
|
/* catastrophic error */
|
|
SCOPE_EXIT_RTN_VALUE(-1, "Coudn't create sdp stream\n");
|
|
}
|
|
if (res > 0) {
|
|
/* Handled by this handler. Move to the next stream */
|
|
session_media_set_handler(session_media, handler);
|
|
SCOPE_EXIT_RTN_VALUE(0, "Handled\n");
|
|
}
|
|
}
|
|
|
|
/* streams that weren't handled won't be included in generated outbound SDP */
|
|
SCOPE_EXIT_RTN_VALUE(0, "Not handled\n");
|
|
}
|
|
|
|
/*! \brief Bundle group building structure */
|
|
struct sip_session_media_bundle_group {
|
|
/*! \brief The media identifiers in this bundle group */
|
|
char *mids[PJMEDIA_MAX_SDP_MEDIA];
|
|
/*! \brief SDP attribute string */
|
|
struct ast_str *attr_string;
|
|
};
|
|
|
|
static int add_bundle_groups(struct ast_sip_session *session, pj_pool_t *pool, pjmedia_sdp_session *answer)
|
|
{
|
|
pj_str_t stmp;
|
|
pjmedia_sdp_attr *attr;
|
|
struct sip_session_media_bundle_group bundle_groups[PJMEDIA_MAX_SDP_MEDIA];
|
|
int index, mid_id;
|
|
struct sip_session_media_bundle_group *bundle_group;
|
|
|
|
if (session->endpoint->media.webrtc) {
|
|
attr = pjmedia_sdp_attr_create(pool, "msid-semantic", pj_cstr(&stmp, "WMS *"));
|
|
pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
|
|
}
|
|
|
|
if (!session->endpoint->media.bundle) {
|
|
return 0;
|
|
}
|
|
|
|
memset(bundle_groups, 0, sizeof(bundle_groups));
|
|
|
|
/* Build the bundle group layout so we can then add it to the SDP */
|
|
for (index = 0; index < AST_VECTOR_SIZE(&session->pending_media_state->sessions); ++index) {
|
|
struct ast_sip_session_media *session_media = AST_VECTOR_GET(&session->pending_media_state->sessions, index);
|
|
|
|
/* If this stream is not part of a bundle group we can't add it */
|
|
if (session_media->bundle_group == -1) {
|
|
continue;
|
|
}
|
|
|
|
bundle_group = &bundle_groups[session_media->bundle_group];
|
|
|
|
/* If this is the first mid then we need to allocate the attribute string and place BUNDLE in front */
|
|
if (!bundle_group->mids[0]) {
|
|
bundle_group->mids[0] = session_media->mid;
|
|
bundle_group->attr_string = ast_str_create(64);
|
|
if (!bundle_group->attr_string) {
|
|
continue;
|
|
}
|
|
|
|
ast_str_set(&bundle_group->attr_string, 0, "BUNDLE %s", session_media->mid);
|
|
continue;
|
|
}
|
|
|
|
for (mid_id = 1; mid_id < PJMEDIA_MAX_SDP_MEDIA; ++mid_id) {
|
|
if (!bundle_group->mids[mid_id]) {
|
|
bundle_group->mids[mid_id] = session_media->mid;
|
|
ast_str_append(&bundle_group->attr_string, 0, " %s", session_media->mid);
|
|
break;
|
|
} else if (!strcmp(bundle_group->mids[mid_id], session_media->mid)) {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Add all bundle groups that have mids to the SDP */
|
|
for (index = 0; index < PJMEDIA_MAX_SDP_MEDIA; ++index) {
|
|
bundle_group = &bundle_groups[index];
|
|
|
|
if (!bundle_group->attr_string) {
|
|
continue;
|
|
}
|
|
|
|
attr = pjmedia_sdp_attr_create(pool, "group", pj_cstr(&stmp, ast_str_buffer(bundle_group->attr_string)));
|
|
pjmedia_sdp_attr_add(&answer->attr_count, answer->attr, attr);
|
|
|
|
ast_free(bundle_group->attr_string);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct pjmedia_sdp_session *create_local_sdp(pjsip_inv_session *inv, struct ast_sip_session *session, const pjmedia_sdp_session *offer)
|
|
{
|
|
static const pj_str_t STR_IN = { "IN", 2 };
|
|
static const pj_str_t STR_IP4 = { "IP4", 3 };
|
|
static const pj_str_t STR_IP6 = { "IP6", 3 };
|
|
pjmedia_sdp_session *local;
|
|
int i;
|
|
int stream;
|
|
SCOPE_ENTER(1, "%s Topology: %s\n", ast_sip_session_get_name(session),
|
|
ast_str_tmp(256, ast_stream_topology_to_str(session->pending_media_state->topology, &STR_TMP)));
|
|
|
|
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
|
|
ast_log(LOG_ERROR, "Failed to create session SDP. Session has been already disconnected\n");
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Disconnected\n");
|
|
}
|
|
|
|
if (!inv->pool_prov || !(local = PJ_POOL_ZALLOC_T(inv->pool_prov, pjmedia_sdp_session))) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Pool alloc failure\n");
|
|
}
|
|
|
|
if (!offer) {
|
|
local->origin.version = local->origin.id = (pj_uint32_t)(ast_random());
|
|
} else {
|
|
local->origin.version = offer->origin.version + 1;
|
|
local->origin.id = offer->origin.id;
|
|
}
|
|
|
|
pj_strdup2(inv->pool_prov, &local->origin.user, session->endpoint->media.sdpowner);
|
|
pj_strdup2(inv->pool_prov, &local->name, session->endpoint->media.sdpsession);
|
|
|
|
if (!session->pending_media_state->topology || !ast_stream_topology_get_count(session->pending_media_state->topology)) {
|
|
/* We've encountered a situation where we have been told to create a local SDP but noone has given us any indication
|
|
* of what kind of stream topology they would like. We try to not alter the current state of the SDP negotiation
|
|
* by using what is currently negotiated. If this is unavailable we fall back to what is configured on the endpoint.
|
|
*/
|
|
ast_stream_topology_free(session->pending_media_state->topology);
|
|
if (session->active_media_state->topology) {
|
|
session->pending_media_state->topology = ast_stream_topology_clone(session->active_media_state->topology);
|
|
} else {
|
|
session->pending_media_state->topology = ast_stream_topology_clone(session->endpoint->media.topology);
|
|
}
|
|
if (!session->pending_media_state->topology) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "No pending topology\n");
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < ast_stream_topology_get_count(session->pending_media_state->topology); ++i) {
|
|
struct ast_sip_session_media *session_media;
|
|
struct ast_stream *stream;
|
|
unsigned int streams = local->media_count;
|
|
|
|
/* This code does not enforce any maximum stream count limitations as that is done on either
|
|
* the handling of an incoming SDP offer or on the handling of a session refresh.
|
|
*/
|
|
|
|
stream = ast_stream_topology_get_stream(session->pending_media_state->topology, i);
|
|
|
|
session_media = ast_sip_session_media_state_add(session, session->pending_media_state, ast_stream_get_type(stream), i);
|
|
if (!session_media) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't add state\n");
|
|
}
|
|
|
|
if (add_sdp_streams(session_media, session, local, offer, stream)) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't add stream\n");
|
|
}
|
|
|
|
/* If a stream was actually added then add any additional details */
|
|
if (streams != local->media_count) {
|
|
pjmedia_sdp_media *media = local->media[streams];
|
|
pj_str_t stmp;
|
|
pjmedia_sdp_attr *attr;
|
|
|
|
/* Add the media identifier if present */
|
|
if (!ast_strlen_zero(session_media->mid)) {
|
|
attr = pjmedia_sdp_attr_create(inv->pool_prov, "mid", pj_cstr(&stmp, session_media->mid));
|
|
pjmedia_sdp_attr_add(&media->attr_count, media->attr, attr);
|
|
}
|
|
}
|
|
|
|
/* Ensure that we never exceed the maximum number of streams PJMEDIA will allow. */
|
|
if (local->media_count == PJMEDIA_MAX_SDP_MEDIA) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* Add any bundle groups that are present on the media state */
|
|
if (add_bundle_groups(session, inv->pool_prov, local)) {
|
|
SCOPE_EXIT_RTN_VALUE(NULL, "Couldn't add bundled\n");
|
|
}
|
|
|
|
/* Use the connection details of an available media if possible for SDP level */
|
|
for (stream = 0; stream < local->media_count; stream++) {
|
|
if (!local->media[stream]->conn) {
|
|
continue;
|
|
}
|
|
|
|
if (local->conn) {
|
|
if (!pj_strcmp(&local->conn->net_type, &local->media[stream]->conn->net_type) &&
|
|
!pj_strcmp(&local->conn->addr_type, &local->media[stream]->conn->addr_type) &&
|
|
!pj_strcmp(&local->conn->addr, &local->media[stream]->conn->addr)) {
|
|
local->media[stream]->conn = NULL;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
/* This stream's connection info will serve as the connection details for SDP level */
|
|
local->conn = local->media[stream]->conn;
|
|
local->media[stream]->conn = NULL;
|
|
|
|
continue;
|
|
}
|
|
|
|
/* If no SDP level connection details are present then create some */
|
|
if (!local->conn) {
|
|
local->conn = pj_pool_zalloc(inv->pool_prov, sizeof(struct pjmedia_sdp_conn));
|
|
local->conn->net_type = STR_IN;
|
|
local->conn->addr_type = session->endpoint->media.rtp.ipv6 ? STR_IP6 : STR_IP4;
|
|
|
|
if (!ast_strlen_zero(session->endpoint->media.address)) {
|
|
pj_strdup2(inv->pool_prov, &local->conn->addr, session->endpoint->media.address);
|
|
} else {
|
|
pj_strdup2(inv->pool_prov, &local->conn->addr, ast_sip_get_host_ip_string(session->endpoint->media.rtp.ipv6 ? pj_AF_INET6() : pj_AF_INET()));
|
|
}
|
|
}
|
|
|
|
pj_strassign(&local->origin.net_type, &local->conn->net_type);
|
|
pj_strassign(&local->origin.addr_type, &local->conn->addr_type);
|
|
pj_strassign(&local->origin.addr, &local->conn->addr);
|
|
|
|
SCOPE_EXIT_RTN_VALUE(local);
|
|
}
|
|
|
|
static void session_inv_on_rx_offer(pjsip_inv_session *inv, const pjmedia_sdp_session *offer)
|
|
{
|
|
pjmedia_sdp_session *answer;
|
|
struct ast_sip_session *session = inv->mod_data[session_module.id];
|
|
SCOPE_ENTER(1, "%s Inv State: %s\n", ast_sip_session_get_name(session),
|
|
pjsip_inv_state_name(inv->state));
|
|
|
|
if (ast_shutdown_final()) {
|
|
SCOPE_EXIT_RTN("Shutting down\n");
|
|
}
|
|
|
|
session = inv->mod_data[session_module.id];
|
|
if (handle_incoming_sdp(session, offer)) {
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
SCOPE_EXIT_RTN("Couldn't handle sdp\n");
|
|
}
|
|
|
|
if ((answer = create_local_sdp(inv, session, offer))) {
|
|
pjsip_inv_set_sdp_answer(inv, answer);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
#if 0
|
|
static void session_inv_on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_session **p_offer)
|
|
{
|
|
/* XXX STUB */
|
|
}
|
|
#endif
|
|
|
|
static void session_inv_on_media_update(pjsip_inv_session *inv, pj_status_t status)
|
|
{
|
|
const pjmedia_sdp_session *local, *remote;
|
|
struct ast_sip_session *session = inv->mod_data[session_module.id];
|
|
SCOPE_ENTER(1, "%s Inv State: %s\n", ast_sip_session_get_name(session),
|
|
pjsip_inv_state_name(inv->state));
|
|
|
|
if (ast_shutdown_final()) {
|
|
SCOPE_EXIT_RTN("Shutting down\n");
|
|
}
|
|
|
|
session = inv->mod_data[session_module.id];
|
|
if (!session || !session->channel) {
|
|
/*
|
|
* If we don't have a session or channel then we really
|
|
* don't care about media updates.
|
|
* Just ignore
|
|
*/
|
|
SCOPE_EXIT_RTN("No session or channel\n");
|
|
}
|
|
|
|
if (session->endpoint) {
|
|
int bail = 0;
|
|
|
|
/*
|
|
* If following_fork is set, then this is probably the result of a
|
|
* forked INVITE and SDP asnwers coming from the different fork UAS
|
|
* destinations. In this case updated_sdp_answer will also be set.
|
|
*
|
|
* If only updated_sdp_answer is set, then this is the non-forking
|
|
* scenario where the same UAS just needs to change something like
|
|
* the media port.
|
|
*/
|
|
|
|
if (inv->following_fork) {
|
|
if (session->endpoint->media.rtp.follow_early_media_fork) {
|
|
ast_debug(3, "Following early media fork with different To tags\n");
|
|
} else {
|
|
ast_debug(3, "Not following early media fork with different To tags\n");
|
|
bail = 1;
|
|
}
|
|
}
|
|
#ifdef HAVE_PJSIP_INV_ACCEPT_MULTIPLE_SDP_ANSWERS
|
|
else if (inv->updated_sdp_answer) {
|
|
if (session->endpoint->media.rtp.accept_multiple_sdp_answers) {
|
|
ast_debug(3, "Accepting updated SDP with same To tag\n");
|
|
} else {
|
|
ast_debug(3, "Ignoring updated SDP answer with same To tag\n");
|
|
bail = 1;
|
|
}
|
|
}
|
|
#endif
|
|
if (bail) {
|
|
SCOPE_EXIT_RTN("Bail\n");
|
|
}
|
|
}
|
|
|
|
if ((status != PJ_SUCCESS) || (pjmedia_sdp_neg_get_active_local(inv->neg, &local) != PJ_SUCCESS) ||
|
|
(pjmedia_sdp_neg_get_active_remote(inv->neg, &remote) != PJ_SUCCESS)) {
|
|
ast_channel_hangupcause_set(session->channel, AST_CAUSE_BEARERCAPABILITY_NOTAVAIL);
|
|
ast_set_hangupsource(session->channel, ast_channel_name(session->channel), 0);
|
|
ast_queue_hangup(session->channel);
|
|
SCOPE_EXIT_RTN("Couldn't get active local\n");return;
|
|
}
|
|
|
|
if (handle_negotiated_sdp(session, local, remote)) {
|
|
ast_sip_session_media_state_reset(session->pending_media_state);
|
|
}
|
|
|
|
SCOPE_EXIT_RTN();
|
|
}
|
|
|
|
static pjsip_redirect_op session_inv_on_redirected(pjsip_inv_session *inv, const pjsip_uri *target, const pjsip_event *e)
|
|
{
|
|
struct ast_sip_session *session;
|
|
const pjsip_sip_uri *uri;
|
|
|
|
if (ast_shutdown_final()) {
|
|
return PJSIP_REDIRECT_STOP;
|
|
}
|
|
|
|
session = inv->mod_data[session_module.id];
|
|
if (!session || !session->channel) {
|
|
return PJSIP_REDIRECT_STOP;
|
|
}
|
|
|
|
if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_PJSIP) {
|
|
return PJSIP_REDIRECT_ACCEPT;
|
|
}
|
|
|
|
if (!PJSIP_URI_SCHEME_IS_SIP(target) && !PJSIP_URI_SCHEME_IS_SIPS(target)) {
|
|
return PJSIP_REDIRECT_STOP;
|
|
}
|
|
|
|
handle_incoming(session, e->body.rx_msg.rdata, AST_SIP_SESSION_BEFORE_REDIRECTING);
|
|
|
|
uri = pjsip_uri_get_uri(target);
|
|
|
|
if (session->endpoint->redirect_method == AST_SIP_REDIRECT_USER) {
|
|
char exten[AST_MAX_EXTENSION];
|
|
|
|
ast_copy_pj_str(exten, &uri->user, sizeof(exten));
|
|
|
|
/*
|
|
* We may want to match in the dialplan without any user
|
|
* options getting in the way.
|
|
*/
|
|
AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(exten);
|
|
|
|
ast_channel_call_forward_set(session->channel, exten);
|
|
} else if (session->endpoint->redirect_method == AST_SIP_REDIRECT_URI_CORE) {
|
|
char target_uri[PJSIP_MAX_URL_SIZE];
|
|
/* PJSIP/ + endpoint length + / + max URL size */
|
|
char forward[8 + strlen(ast_sorcery_object_get_id(session->endpoint)) + PJSIP_MAX_URL_SIZE];
|
|
|
|
pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri, target_uri, sizeof(target_uri));
|
|
sprintf(forward, "PJSIP/%s/%s", ast_sorcery_object_get_id(session->endpoint), target_uri);
|
|
ast_channel_call_forward_set(session->channel, forward);
|
|
}
|
|
|
|
return PJSIP_REDIRECT_STOP;
|
|
}
|
|
|
|
static pjsip_inv_callback inv_callback = {
|
|
.on_state_changed = session_inv_on_state_changed,
|
|
.on_new_session = session_inv_on_new_session,
|
|
.on_tsx_state_changed = session_inv_on_tsx_state_changed,
|
|
.on_rx_offer = session_inv_on_rx_offer,
|
|
.on_media_update = session_inv_on_media_update,
|
|
.on_redirected = session_inv_on_redirected,
|
|
};
|
|
|
|
/*! \brief Hook for modifying outgoing messages with SDP to contain the proper address information */
|
|
static void session_outgoing_nat_hook(pjsip_tx_data *tdata, struct ast_sip_transport *transport)
|
|
{
|
|
RAII_VAR(struct ast_sip_transport_state *, transport_state, ast_sip_get_transport_state(ast_sorcery_object_get_id(transport)), ao2_cleanup);
|
|
struct ast_sip_nat_hook *hook = ast_sip_mod_data_get(
|
|
tdata->mod_data, session_module.id, MOD_DATA_NAT_HOOK);
|
|
struct pjmedia_sdp_session *sdp;
|
|
int stream;
|
|
|
|
/* SDP produced by us directly will never be multipart */
|
|
if (!transport_state || hook || !tdata->msg->body ||
|
|
!ast_sip_is_content_type(&tdata->msg->body->content_type, "application", "sdp") ||
|
|
ast_strlen_zero(transport->external_media_address)) {
|
|
return;
|
|
}
|
|
|
|
sdp = tdata->msg->body->data;
|
|
|
|
if (sdp->conn) {
|
|
char host[NI_MAXHOST];
|
|
struct ast_sockaddr our_sdp_addr = { { 0, } };
|
|
|
|
ast_copy_pj_str(host, &sdp->conn->addr, sizeof(host));
|
|
ast_sockaddr_parse(&our_sdp_addr, host, PARSE_PORT_FORBID);
|
|
|
|
/* Reversed check here. We don't check the remote
|
|
* endpoint being in our local net, but whether our
|
|
* outgoing session IP is local. If it is, we'll do
|
|
* rewriting. No localnet configured? Always rewrite. */
|
|
if (ast_sip_transport_is_local(transport_state, &our_sdp_addr) || !transport_state->localnet) {
|
|
ast_debug(5, "Setting external media address to %s\n", ast_sockaddr_stringify_addr_remote(&transport_state->external_media_address));
|
|
pj_strdup2(tdata->pool, &sdp->conn->addr, ast_sockaddr_stringify_addr_remote(&transport_state->external_media_address));
|
|
pj_strassign(&sdp->origin.addr, &sdp->conn->addr);
|
|
}
|
|
}
|
|
|
|
for (stream = 0; stream < sdp->media_count; ++stream) {
|
|
/* See if there are registered handlers for this media stream type */
|
|
char media[20];
|
|
struct ast_sip_session_sdp_handler *handler;
|
|
RAII_VAR(struct sdp_handler_list *, handler_list, NULL, ao2_cleanup);
|
|
|
|
/* We need a null-terminated version of the media string */
|
|
ast_copy_pj_str(media, &sdp->media[stream]->desc.media, sizeof(media));
|
|
|
|
handler_list = ao2_find(sdp_handlers, media, OBJ_KEY);
|
|
if (!handler_list) {
|
|
ast_debug(1, "No registered SDP handlers for media type '%s'\n", media);
|
|
continue;
|
|
}
|
|
AST_LIST_TRAVERSE(&handler_list->list, handler, next) {
|
|
if (handler->change_outgoing_sdp_stream_media_address) {
|
|
handler->change_outgoing_sdp_stream_media_address(tdata, sdp->media[stream], transport);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* We purposely do this so that the hook will not be invoked multiple times, ie: if a retransmit occurs */
|
|
ast_sip_mod_data_set(tdata->pool, tdata->mod_data, session_module.id, MOD_DATA_NAT_HOOK, nat_hook);
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
pjsip_endpoint *endpt;
|
|
|
|
if (!ast_sip_get_sorcery() || !ast_sip_get_pjsip_endpoint()) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
if (!(nat_hook = ast_sorcery_alloc(ast_sip_get_sorcery(), "nat_hook", NULL))) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
nat_hook->outgoing_external_message = session_outgoing_nat_hook;
|
|
ast_sorcery_create(ast_sip_get_sorcery(), nat_hook);
|
|
sdp_handlers = ao2_container_alloc_hash(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
|
|
SDP_HANDLER_BUCKETS, sdp_handler_list_hash, NULL, sdp_handler_list_cmp);
|
|
if (!sdp_handlers) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
endpt = ast_sip_get_pjsip_endpoint();
|
|
pjsip_inv_usage_init(endpt, &inv_callback);
|
|
pjsip_100rel_init_module(endpt);
|
|
pjsip_timer_init_module(endpt);
|
|
if (ast_sip_register_service(&session_module)) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
ast_sip_register_service(&session_reinvite_module);
|
|
ast_sip_register_service(&outbound_invite_auth_module);
|
|
|
|
ast_module_shutdown_ref(ast_module_info->self);
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
ast_sip_unregister_service(&outbound_invite_auth_module);
|
|
ast_sip_unregister_service(&session_reinvite_module);
|
|
ast_sip_unregister_service(&session_module);
|
|
ast_sorcery_delete(ast_sip_get_sorcery(), nat_hook);
|
|
ao2_cleanup(nat_hook);
|
|
ao2_cleanup(sdp_handlers);
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "PJSIP Session resource",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.load_pri = AST_MODPRI_APP_DEPEND,
|
|
.requires = "res_pjsip",
|
|
);
|